mio Voice without mio box!

[liangtam]

The main guide has been condensed. Here are the updated steps.
There are 2 method where you can use to achieve mio Voice.

See below post for information

Note: Do not want to support this thread liao

[liangtam]

Method 1: See avatarlam's thread

Method 2: The older method. Enter voip.singtel.com as the voip server to your softphone OR ATA. The more costly method. But is inline with the thread title, you may now keep or give away the 2Wires.

Take ur pick :)

PS: Please DO NOT ASK SingTel for support with regards to the use of a VoIP phone adapter or any other issue! This is also one of the reason why SingTel offers the 2Wire instead.

[danxiaogui86]

Nice......

[shanex88]

nice info there! btw for the softphone, do i need to disable my mio voice at the mio box to use it? coz i try the softphone keep register error...

[liangtam]

Try disabling it.
I have not figured out how their server register request with number of connection yet.

Some VoIP provider restricts to one connection/login at any time.

[Lastexile]

thanks liangtam, just open alot of options for mio users. We are no longer bounded to the 2wire modem!

[felime]

just tried it. very clear outgoing calls but when talking to someone using this software, my friend say when talking to me can hear very bad sound quality like as if talking to some monster.

[Lastexile]

one question, i'm curious with the ip is inserted in the domain field and not the ip address field instead?

[liangtam]

just tried it. very clear outgoing calls but when talking to someone using this software, my friend say when talking to me can hear very bad sound quality like as if talking to some monster.
Thats bad. Try removing the 2Wire to test, cant ascertain if the HGV can interfere not.

[liangtam]

one question, i'm curious with the ip is inserted in the domain field and not the ip address field instead?
No worries, a domain is still resolved to a IP at the end also.
Just like accessing http://www.google.com is akin to accessing http://64.233.189.104, its the same.
Since there is no domain for the mio server, we'll just enter the direct server address. 10.161.129.36

It is also called the proxy server in some instance.

[thiamhui]

I extract the following information from the 2Wire modem.

VOIP Service Type: SIP
Server IP Address: 10.161.129.36
Server Port: 5060
Domain: moip.dect

I'm testing on some software which can do VoIP on my PDA. But after use, I found that I can't make any calls using the phone connected to the mio box. Got to reset the mio box to get back the connection.

Any other ways to work around this issue?

[liangtam]

I extract the following information from the 2Wire modem.

VOIP Service Type: SIP
Server IP Address: 10.161.129.36
Server Port: 5060
Domain: moip.dect

I'm testing on some software which can do VoIP on my PDA. But after use, I found that I can't make any calls using the phone connected to the mio box. Got to reset the mio box to get back the connection.

Any other ways to work around this issue?
Precisely mentioned on other VoIP thread worldwide. Some server only accept connection from the device that connects in later. The device connected earlier will not get to make or receive a call, unless you can get the mio box to re-establish a conn without restarting.

So how was the experience with your PDA? Getting a VoIP box would be good, no need go thru mio box, less problem for some :)

[felime]

Thats bad. Try removing the 2Wire to test, cant ascertain if the HGV can interfere not.

tried already and the monstreous sound still there. it's only good for outgoing calls cos can hear crystal clear sound.. the only setback is the other party will hear you talking like a monster.. so i'll stick to my mio voice connected thru mio box since there's no problem with it.

[Lastexile]

tried already and the monstreous sound still there. it's only good for outgoing calls cos can hear crystal clear sound.. the only setback is the other party will hear you talking like a monster.. so i'll stick to my mio voice connected thru mio box since there's no problem with it.

did you tried it with a voip adapter or ...?

[felime]

did you tried it with a voip adapter or ...?

i tried the softphone recommended by TS

[richneo]

tried yesterdae with xlite and voip adaptor yesterdae no problems at all

[tkdguru]

wad i wanna noe is that if those adapters are complete replacement of that 2wire hgv2 r not, adsl voip all in 1. if also got more of such devices any recommendations

[liangtam]

It is based on SIP standards, so quite a few h/w will work, just that SingTel will not support the other device.
Still find using a SIP box to make call better than using spkr and mic with softphone.

[teemie]

tried yesterdae with xlite and voip adaptor yesterdae no problems at all

hey which voip adaptor r u using? i've tried using the PAP2T and it gives me very loud (unwanted) noise on the PAP2T side. on the mobile side (which i'm calling), there's no sound at all, even if i speak on either side.

i noticed someone said something about the silence supression thing and huawei SIP server?

anyone tried the PAP2T before?

[outer92]

2) Using a VoIP Phone adaptor

Models available on the market which you may consider would be the D-Link DVG 2001s, Linksys/Sipura PAP2. Of course, if you have more money to splurge on, you can even get a tiny WiFi phone.

Similarly, with the credentials available above, we can just configure in these info into the device.

D-Link DVG-2001S

http://i9.tinypic.com/4tuazk2.jpg

http://i10.tinypic.com/6hdfec9.jpg

Heres the same qn again: Any 2Wire for moi??

useing this setup Configuration, still cant connect to mio. fail

[liangtam]

Gotch see my PM yet?

[Lastexile]

cannot huh? my friend brought the dlink voip adapter already

[liangtam]

Pls do not forget to turn off the ALG and reserve port 5060 TCP/UDP for the ATA box, if you rly wan to use the mio box for routing still.

Pls reply my PM if u'r still keen.

[Lastexile]

still cant get it work after disabling alg and port forward 5060

[Lastexile]

useing this setup Configuration, still cant connect to mio. fail


any update?

[Lastexile]

http://i16.tinypic.com/4u941uc.jpg

outer92 can you try the settings here? not very sure about the standards of mio. worth a test. maybe gonna try it at my friend's place later too

[bizcon]

can I use it in other SNBB connection??.....portability??

[liangtam]

can I use it in other SNBB connection??.....portability??
Dun think the VoIP server is accessible for non mio Voice user.

[teemie]

Dun think the VoIP server is accessible for non mio Voice user.
i've confirmed that you'll need the mio box to access the mio voice VoIP server, or at least a modem that can access multiple links/channels at the same time.

[bizcon]

multiple links/channel??

Can ST570 do the job?

[liangtam]

Pls see the first post. Can't figure out what had SingTel done yet with the mio to make it work, compared to 3rd party h/w.

[bizcon]

i tried that with a softphone......now my MIO box itself cannot make any call..... despite serveral reboot. but my softphone can.....any sense to any1 here??

by the way, I am wondering if we can do some port forwarding to the sip server instead. Thus we can get connected by dialing our router IP instead. The SIP server address is internally within singnet but externally from the router. anyway?

[richneo]

exit ur softphone and restart ur modem so it can register with the sip server again

[Lastexile]

Hi i have extracted the static route for 2wire

Route List
Subnet IP Subnet Mask Gateway Interface
127.0.0.1 255.255.255.255 127.0.0.1 lo0
220.255.58.240 255.255.255.255 220.255.58.240 ppp0
192.168.1.254 255.255.255.255 192.168.1.254 bridge0
10.214.120.74 255.255.255.255 10.214.120.74 bridge5
121.6.36.1 255.255.255.255 220.255.58.240 ppp0
220.255.58.240 255.255.255.255 220.255.58.239 bridge0
192.168.1.0 255.255.255.0 192.168.1.254 bridge0
10.214.64.0 255.255.192.0 10.214.120.74 bridge5
10.161.0.0 255.255.0.0 10.214.127.254 bridge5
127.0.0.0 255.0.0.0 127.0.0.1 lo0
0.0.0.0 0.0.0.0 121.6.36.1 ppp0
10.161.0.0 255.255.0.0 -- ipnet4

issit possible to add static route to dlink routers?

[Lastexile]

at first i thought singnet did mac fliter but it seems to be the routing instead.

[snarl]

if you signed up for Mio Voice, would you be able to use something like the Dlink SIP device to connect overseas, on using the overseas broadband? Basically, I'm wondering if you can get a service comparable to Starhubs Digital Voice travel, i.e. singapore call rates overseas, with singapore number.

Thanks for any advice!

[liangtam]

if you signed up for Mio Voice, would you be able to use something like the Dlink SIP device to connect overseas, on using the overseas broadband? Basically, I'm wondering if you can get a service comparable to Starhubs Digital Voice travel, i.e. singapore call rates overseas, with singapore number.

Thanks for any advice!
The SingTel mio voice SIP server is on a private network only accessible with some login parameters, thus you cannot use it on non-SingNet connection. And must be a 2Wire unless anyone can figure out what they've done.

You can subscribe to pfingo, XLvoice etc. Maybe mio mobile might be accessible, but then they don't provide or support external devices - so........

[snarl]

Thanks for the info. So can rule out Mio Voice then.

Will do a search on other VOIP providers then - if you have any information on VOIP providers with local numbers, let me know!

[NoM2005]

Thanks for the info. So can rule out Mio Voice then.

Will do a search on other VOIP providers then - if you have any information on VOIP providers with local numbers, let me know!
Local provider... Saw this sometime back and posted here before, but looks like using other adaptor - not Linksys Adaptor... $5.00 per month

http://www.smarttel.com.sg/services_corporate_smartvoice.html


however they also have another using Linksys SAP3102 Adaptor... $10.00 per month

http://www.smarttel.com.sg/services_corporate_smartaccess.html

:)

[quakytan]

anyone wants to trail pfingo for 6 months? I can send invites to you............When you sign up you get a voip number starting with 3 and $3 of idd and $5 of sms free (I think). The trial is for 6 months and U can make free local calls to land and mobile nos free. They have not say how much they will charge after that.
I'm using it with my E61 and software on my notebook.

----------------Errrrrrrrr
Just found out that you can go to the website to register for a free account without invites.......www.pfingo.com................

[liangtam]

anyone wants to trail pfingo for 6 months? I can send invites to you............When you sign up you get a voip number starting with 3 and $3 of idd and $5 of sms free (I think). The trial is for 6 months and U can make free local calls to land and mobile nos free. They have not say how much they will charge after that.
I'm using it with my E61 and software on my notebook.

----------------Errrrrrrrr
Just found out that you can go to the website to register for a free account without invites.......www.pfingo.com................
With invites, you get a $5 credit, the person who registered under invites will get double IDD value for free trial, not bad la.

[tech010101]

SingTel never make money from VOIP, like that make to make profit?

[Chervelle]

any idea which r the phones to be supported by mio or pfingo in the near future?
not many phones r supported now

[liangtam]

any idea which r the phones to be supported by mio or pfingo in the near future?
not many phones r supported now
most likely still the mio box.
For pfingo can use any ATA SIP box as u like.

[hangyong]

liangtam or others using MIO Voice

if you plug the ATA into port 4 of the MIO box, will the ATA work with the MIO VoIP server?

[liangtam]

When the unit in gateway or bridged mode?

[hangyong]

When the unit in gateway or bridged mode?

would like to know for both.

[liangtam]

port 4 ish the one with bridging to allow for mioTV is it?

The mio Voice VoIP service is only reachable when in gateway mode tho, thus you can have a ATA working, otherwise can't do so when in normal bridge as haven figure out(its a failure)

Anyway, the 2wire ish working fine now, so no need for a external ATA device?

[hangyong]

port 4 ish the one with bridging to allow for mioTV is it?

The mio Voice VoIP service is only reachable when in gateway mode tho, thus you can have a ATA working, otherwise can't do so when in normal bridge as haven figure out(its a failure)

Anyway, the 2wire ish working fine now, so no need for a external ATA device?

well.. it used to be...

what happens with my current 2700HGV is that I set it to bridge mode... but no matter how I plug the cables, it can detect if its MIOTV or bridged to router, automatically.

so just wondering if I put an ATA in, it will work for MIO Voice.

[hangyong]

so has anyone managed to get an ATA to work with Mio Voice?

[richneo]

so has anyone managed to get an ATA to work with Mio Voice?
waiting for u to test with ur siemens voip leh

[hangyong]

waiting for u to test with ur siemens voip leh

I dun have mio voice account

[liangtam]

So far mio Voice server contactable only in gateway mode. Until someone figured out, that ish.
Do note that I do not have a 2Wire HGV either.

[hangyong]

So far mio Voice server contactable only in gateway mode. Until someone figured out, that ish.
Do note that I do not have a 2Wire HGV either.

ok, from your analysis to date, how do you think the blocking is done? routing? or merely blocked?

as far as I know, the phone port is just assigned like another port, much like a network port.

Is its IP address different from MIO TV? Or is it the same IP, but using different ports? Or is it a different IP address all together?

for those who cannot get connected, is the ATA getting a 10.x.x.x IP address?

anyone able to extract out the routing table of the 2wire?

[liangtam]

You can see the static route posted by lastexile.
10.161.x IP is with reference to ipnet4...

IF you use it in gateway mode with your Siemens DECT, the guide on 1st post should apply ba.

[hangyong]

hmm.... if you bound ipnet4 to one of the physical ports, it should solve the problem, no?

[liangtam]

That I gotch no idea wor.
But the goal is still to have 2Wire not in the setup of eq. :)

[hangyong]

That I gotch no idea wor.
But the goal is still to have 2Wire not in the setup of eq. :)

it is not possible to take the 2wire out of the picture unless you have a fully configurable router like cisco.

my 2wire is now in bridged mode, and connected to the D-Link DIR-655, which serves the rest of the network.

The MioTV, however, is connected to the 2Wire.

The interesting thing is, regardless of which port I plug the DIR or the MIO into on the 2Wire, its able to detect them and work accordingly, so seems like they have managed to make it dynamic.

So I am wondering, if I plug an ATA into the 2Wire, will it be able to detect automatically, since the phone port is simply another physical/logical port to the 2Wire.

[liangtam]

Don't think so. Afterall mioTV STB is from SingTel, while ur ATA adapter is not.
You sign up few months try lor.

[hangyong]

Don't think so. Afterall mioTV STB is from SingTel, while ur ATA adapter is not.
You sign up few months try lor.

kaoz.. minimum 3 months leh!

[arunhlms]

hi there ... I got the Linksys pap2 ... followed the settings as indicated ... also got the port forwarding enabled on my router ... but can't get it to work ...

any ideas on what could be wrong?

[alanchia67]

liangtam has responded in your original thread, you still need the 2wire to be around.

[lwq2000]

Hey guys, take a look at this :http://home.singtel.com/mio/mio_mobile_setup_se-2.html

Although this is for mio mobile, I tried the address followed by a ":5060" (suggested by an earlier entry in this thread) for my mio voice and suprizingly, it works with a Singnet broadband connection. Can anyone verify that if this address is accessible outside(ie.other provider's net service)?Or it might also work with a mio mobile?

[deckcard]

Hi,
sorry to join the thread this late and my question may have been answered.
Question:
Do I need to sign up anything to have VoIP? I have Mio plan now.

[arunhlms]

Hi,

Did you try using the Linksys PAP2 or some other device? Also did you use the Mio Box provided by Singtel or another ADSL modem?



Hey guys, take a look at this :http://home.singtel.com/mio/mio_mobile_setup_se-2.html

Although this is for mio mobile, I tried the address followed by a ":5060" (suggested by an earlier entry in this thread) for my mio voice and suprizingly, it works with a Singnet broadband connection. Can anyone verify that if this address is accessible outside(ie.other provider's net service)?Or it might also work with a mio mobile?

[liangtam]

Hey guys, take a look at this :http://home.singtel.com/mio/mio_mobile_setup_se-2.html

Although this is for mio mobile, I tried the address followed by a ":5060" (suggested by an earlier entry in this thread) for my mio voice and suprizingly, it works with a Singnet broadband connection. Can anyone verify that if this address is accessible outside(ie.other provider's net service)?Or it might also work with a mio mobile?
You refering to mio mobile VoIP or mio Voice??

[infinity2030]

Hey guys, take a look at this :http://home.singtel.com/mio/mio_mobile_setup_se-2.html

Although this is for mio mobile, I tried the address followed by a ":5060" (suggested by an earlier entry in this thread) for my mio voice and suprizingly, it works with a Singnet broadband connection. Can anyone verify that if this address is accessible outside(ie.other provider's net service)?Or it might also work with a mio mobile?
It works for both mio voice and mobile. However, for mobile's case, it can only be used for outgoing call where the mio voice can use it for bothways.

[lwq2000]

I tried it with my mio Voice using Singnet Broadband connection via mio box(2wire). Thats why I am asking to see if anyone can try out at the wireless@sg hotspots or via a Starhub broadband connection or even overseas internet connection(maybe we can therefore save on IDD). Lol.

[infinity2030]

I tried it with my mio Voice using Singnet Broadband connection via mio box(2wire). Thats why I am asking to see if anyone can try out at the wireless@sg hotspots or via a Starhub broadband connection or even overseas internet connection(maybe we can therefore save on IDD). Lol.
Yes. It will work as i mentioned on my previous post. :s8:

[liangtam]

mio Voice on voip.singtel.com or 10.161.129.36??

[alanchia67]

Hi,
sorry to join the thread this late and my question may have been answered.
Question:
Do I need to sign up anything to have VoIP? I have Mio plan now.

mioplan is mobile + snbb + fixed line in a bill plan. the fixed line could be voip or traditional analogue line - by default it would be converted to voip (you receive a letter stating your mio home phone number and the corresponding system id).

[deckcard]

mioplan is mobile + snbb + fixed line in a bill plan. the fixed line could be voip or traditional analogue line - by default it would be converted to voip (you receive a letter stating your mio home phone number and the corresponding system id).
Thanks for explaining. We still kept our "old" phone number so does it mean we are analogue? The day of installation, we got a few hours of downtime with only static noise on the phone line. What does that mean? Btw, have not received any letters from Singtel.

[alanchia67]

even with mio voip (aka mio voice) you can retain your old number. if you didnt receive the letter at all, this means you're still using analogue voice. i supposed you didnt even setup mio voice configuration in the 2wire right?

[deckcard]

even with mio voip (aka mio voice) you can retain your old number. if you didnt receive the letter at all, this means you're still using analogue voice. i supposed you didnt even setup mio voice configuration in the 2wire right?

Well, the tech came down and fix everything. I looked at the voice setting on the router and it says:


Telephone Number 1 (Red Port) : my phone number
System ID 1 : *******

The tech guy also says we should get a letter tell our login password but for some reason, I did not receive anything. Anyway, he went online and got the password so he could set things up.. Guess I have VoIP?

[alanchia67]

yup, that is mio voice (aka voip).

[infinity2030]

mio Voice on voip.singtel.com or 10.161.129.36??
voip.singtel.com works throughout. Where 10.161.129.36 only works within snbb env.

[liangtam]

voip.singtel.com works throughout. Where 10.161.129.36 only works within snbb env.
Really??
mio Voice can auth and work with voip.singtel.com!?

Can edit first post if true :D

[lwq2000]

Yah, can anyone try mio Voice outside home or use starhub's connection, so that if really voip.singtel.com can work... than there is a chance to save on local calls while overseas... :)

[infinity2030]

Really??
mio Voice can auth and work with voip.singtel.com!?

Can edit first post if true :D
Yes, it will work. I've tried it since together with mio Mobile when i got the mio Voice back when it just launched.

[infinity2030]

Yah, can anyone try mio Voice outside home or use starhub's connection, so that if really voip.singtel.com can work... than there is a chance to save on local calls while overseas... :)
Yes. It works. It was tried and tested by me. Works without a hitch. =)

[deckcard]

Yay, I tried using C730 WM6 with hsdpa using Fring and it works! Thanks guys. I wondered what is IDD rate tho..

[infinity2030]

Yay, I tried using C730 WM6 with hsdpa using Fring and it works! Thanks guys. I wondered what is IDD rate tho..
Hmm.. You don't really need to worry about IDD rate while calling back to singapore through internet via this method while you're overseas..

[liangtam]

Updated 1st post. Anyone with mio Voice can try without a mio box/2Wire 2700HGV in place??

[infinity2030]

Updated 1st post. Anyone with mio Voice can try without a mio box/2Wire 2700HGV in place??
How about me? I've tried. No issues in anyway.

[deckcard]

Hmm.. You don't really need to worry about IDD rate while calling back to singapore through internet via this method while you're overseas..
But I want to call oversea like HK or China if the rate is reasonable.

I can use pc software to call Sing when in HK but wont be as mobile as using a cell phone to call via HSDPA or 3G. The hsdpa plan here is cheaper than HK.

[liangtam]

You shld look for alternative if wan call China/HK ba. mio Voice/mobile more for calling local numbers.

[alanchia67]

ok, am using WM6 native VoIP and I manage to receive the phone call using my mobile phone (connected to home wifi/or hsdpa) for the mio voice number. also managed to do a callout to normal 8-digit local phone number.

cool!

all thanks to infinity2030 for the observation for mio mobile. =:p

[deckcard]

ok, am using WM6 native VoIP and I manage to receive the phone call using my mobile phone (connected to home wifi/or hsdpa) for the mio voice number. also managed to do a callout to normal 8-digit local phone number.

cool!

all thanks to infinity2030 for the observation for mio mobile. =:p
After using VoIP on my mobile, my home phone is no longer working :s8:
I also lost internet connection and have to reboot the MIO box, wondered what happened.
I'm guessing the software is not doing a proper "logout" so the MIO box can regain the phone control?

[liangtam]

After using VoIP on my mobile, my home phone is no longer working :s8:
I also lost internet connection and have to reboot the MIO box, wondered what happened.
I'm guessing the software is not doing a proper "logout" so the MIO box can regain the phone control?
Yar, if timeout does not work to regain phone access to 2Wire, then you really have to force the 2Wire to re-auth, so far I did not see such feature other than in mdc, so maybe you rly have to reboot.
Most VoIP provider only allows 1 VoIP client connected and in use at any time, so the previous client will not be able to receive or make calls.

Internet conn dc maybe is of another matter??

[alanchia67]

have not tried, but shouldnt you be able to untick, and retick the voice setting to re-enable authentication (w/o using mdc)?

no i didnt lost internet connection at all.

[alanchia67]

hey, am wondering - since i can use wm6 native voip feature to connect to mio voice, WHY isnt wm6 a supported platform for mio mobile? instead, only selected nokia and sony ericsson phone are supported. strange isnt it?

[deckcard]

Well, have to do more testing to see what went wrong.

Anyhow, voip is native to wm6 but some wm6 phones do not have voip feature. Like on c730, i do not see voip feature hence i'm using fring. Go figure.

[liangtam]

So they will have less items to 'support' :)

[alanchia67]

Well, have to do more testing to see what went wrong.

Anyhow, voip is native to wm6 but some wm6 phones do not have voip feature. Like on c730, i do not see voip feature hence i'm using fring. Go figure.

wm6 has voip in the core. you just need to install additional s/w (not found in the phone, but is supplied by MS for wm6) to enable voip support.

[alanchia67]

So they will have less items to 'support' :)

walau, wm6 issues will be handled by MS mah... furthermore wm6 is voip ready - just need to know how to enable it that's all.

[deckcard]

wm6 has voip in the core. you just need to install additional s/w (not found in the phone, but is supplied by MS for wm6) to enable voip support.
Ahh.. Do you have any links so I can check out? The closest I found is this:

http://forum.xda-developers.com/showthread.php?t=299950

And I cant find the files stated in the post so I guess I'm out of luck using native support

[alanchia67]

Ahh.. Do you have any links so I can check out? The closest I found is this:

http://forum.xda-developers.com/showthread.php?t=299950

And I cant find the files stated in the post so I guess I'm out of luck using native support

yup that's the correct forum/link! the files are still in xda, you have account there? pm me if you need help getting them.

[deckcard]

yup that's the correct forum/link! the files are still in xda, you have account there? pm me if you need help getting them.

I dont have account on xda, I meant I do not have the files in \Windows directory:

* ipdialplan.xml
* dnsapi.dll
* voipphonecanvas.dll
* rtcdll.dll

I cant find these files on my c730

[alanchia67]

you need to install wm6voip.cab to get those files. reboot, and install and run sipconfigtool_2_0_1.cab to enter the voip configuration. have fun!

[deckcard]

Ahh.. I found wm6voip.cab in the first page but cannot find sipconfigtoo_2_0_1.cab.. That thread is over 100 pages.....

[deckcard]

Also, I read the sound comes out from rear speaker using the native support?

[deckcard]

Ok, found sipconfigtool_2_0_1.cab, for completness (if anyone wanna try) the link is:
http://forum.xda-developers.com/showthread.php?t=308931

[alanchia67]

huh? the sound comes from my normal front speaker.

anyway we've went alittle offthread. maybe should create another thread for this purpose.

[deckcard]

you need to install wm6voip.cab to get those files. reboot, and install and run sipconfigtool_2_0_1.cab to enter the voip configuration. have fun!
Ok, I installed the sipconfigtool, but I still dont see where to do the voip configuration.. :s11:

[deckcard]

huh? the sound comes from my normal front speaker.

anyway we've went alittle offthread. maybe should create another thread for this purpose.

Oh ya.. Sorry OP for hijacking :D

[liangtam]

huh? the sound comes from my normal front speaker.

anyway we've went alittle offthread. maybe should create another thread for this purpose.
I dun mind actually. Do discuss abit more on mio Voice or mobile related tho :)

[alanchia67]

run the configtool:
enter anything for description (i use "mio voice")
enter voip.singtel.com for sip server
enter your home number for user name 6xxxxxxx
enter the system id for password.

save configuration, and you may want to enable sip over 3g under tools.
reboot.
make sure you can see the internet phone option in 'today' screen. and select appropriate option in settings->phone->internet.

the internet phone option in today screen toggles should you use voip or regular gsm/3g for the NEXT phone call.

have fun.

the above should be both applicable for mio voice or mio mobile for wm6.

too bad wm6 doesnt have option to change to different voip profiles.

[deckcard]

The configtool crashed when I ran it. Ahh well, will try it later...

[echoz]

Try this one instead,

http://forum.xda-developers.com/showthread.php?t=317070

worked on pfingo for me. WM6 Native. Am using a Dopod 838Pro with a custom ROM by Schap.

[Yunnnn]

Hi,

Thank you for this post :)!

Trying to find another VOIP software now because x-lite doesn't seem to work for me.

[hangyong]

hey, am wondering - since i can use wm6 native voip feature to connect to mio voice, WHY isnt wm6 a supported platform for mio mobile? instead, only selected nokia and sony ericsson phone are supported. strange isnt it?

eh, you using which phone got native VoIP on WM6???

my Kaiser also need to hack one... and the sound comes out from the speaker, unless using headset...

I am using SJPhone, now

[liangtam]

Hi,

Thank you for this post :)!

Trying to find another VOIP software now because x-lite doesn't seem to work for me.
???

You mean PC? If Xlite does not work, then you got firewall or related issues already.

[Yunnnn]

???

You mean PC? If Xlite does not work, then you got firewall or related issues already.

Very choppy and weird noises...:(.

[deckcard]

Try this one instead,

http://forum.xda-developers.com/showthread.php?t=317070

worked on pfingo for me. WM6 Native. Am using a Dopod 838Pro with a custom ROM by Schap.
Tried this one too and also crashed.

Well, I'm sticking to fring for now, at least it works.

[alanchia67]

eh, you using which phone got native VoIP on WM6???

my Kaiser also need to hack one... and the sound comes out from the speaker, unless using headset...

I am using SJPhone, now

dopod838pro with HTC's original wm6 rom. just need to install additional drivers (wm6voip) to enable the voip sip provisioning. the sound comes out from the front like normal phone for me.

ok offthread liao, i'll create a new thread for this.

[deckcard]

Did you guys install SDA application unlock?

[hangyong]

you guys with MIO Voice wanna try this out??


VOIP Service Type: SIP
Server IP Address: 10.161.129.68
Server Port: 5060
Domain: moip.dect
Register Expire Time: 300
Register Retry Interval: 300

seems like there is a change in the IP of the SIP server/gateway

[liangtam]

See my first post on update 2.
If voip.singtel.com works, then no point using the internal ip addr of segmented voice serbice.

[bizcon]

I am testing PAP2. Voice seems quite okie. But I cannot fax. Any1 know the setting?

[cloudbp]

so any other modem can support the mio Voice?? i saw this modem http://www.dlink.com/products/?sec=0&pid=446
but i dunno whether can it work with moi Voice

[liangtam]

so any other modem can support the mio Voice?? i saw this modem http://www.dlink.com/products/?sec=0&pid=446
but i dunno whether can it work with moi Voice

Generally, if softphone works with 2Wire out of the picture, ....
Yes, SIP ATA boxes shld as well.

PLS try with a softphone with 2Wire either removed in setup or bridge mode before you go order a external SIP ata.

[cloudbp]

okay thx i will ttry out

[xieliwei]

Is it me or is it that voip.singtel.com is filtering house numbers now?

I can't register using voip.singtel.com; 10.161.129.68 still works though:

Reliably Transmitting (no NAT) to 202.166.85.196:5060:
REGISTER sip:voip.singtel.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=xxxxxxxxxxxxxx;rport
From: <sip:xxxxxxxx@voip.singtel.com>;tag=xxxxxxxxxx
To: <sip:xxxxxxxx@voip.singtel.com>
Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx@xxx.xxx.xxx.xxx
CSeq: 102 REGISTER
User-Agent: Asterisk
Max-Forwards: 70
Expires: 3600
Contact: <sip:s@xxx.xxx.xxx.xxx>
Event: registration
Content-Length: 0
---
<-- SIP read from 202.166.85.196:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=xxxxxxxxxxxxxx;rport=5060
Call-ID: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx@xxx.xxx.xxx.xxx
From: <sip:xxxxxxxx@voip.singtel.com>;tag=xxxxxxxxxx
To: <sip:xxxxxxxx@voip.singtel.com>;tag=xxxxxxxx
CSeq: 102 REGISTER
Content-Length: 0

[HoGnix]

I have checked, 202.166.85.196 is the same as voip.singtel.com

[xieliwei]

Yes. But I'm not recognised on their side as a valid user now.

Does yours still work?

[efencer]

Got the same problem, seems voip.singtel.com does not work properly for me anymore.
I was using Sipura 3000 ATA to register and it was working fine until this week, somehow I did not want to register anymore. The ATA is working fine as I can register for my none local calls without a problem to other SIP servers.

Seems they are really made some changes.

[liangtam]

How about softphone?

[xieliwei]

Same. Tells me user not found.

I can't use the internal IP 10.161.129.36 either, it fails to register after a while (no response) and won't come back until I stop registering for at least an hour and restart. I sense the 2Wire's SIP ALG causing this.

This is bad, I have to revert to using the 2Wire ATA, and its horrible.

Plus all the IP telephony infrastructure I've invested in becomes useless.

[liangtam]

IF you're still using the 2Wire in gateway mode, the internal IP shld work.

[xieliwei]

Yes, the internal IP works, but somehow I'm getting intermittent responses when registering.

Below I log my attempts to troubleshoot the problem, hopefully someone experienced can help me.

I will be able to register initially. However, after a while, the telephony server stops responding to everything I send it. I'll then have to stop trying to register for a while; sometimes for half an hour, sometimes a few hours; and try registering again.

From this I can have three possibilities:
1. Data sent from me to the server cannot be reached.
2. Data sent from the server to me cannot be received.
3. Data from both sides are lost.

To further investigate the problem, I tried calling in immediately after the server stops responding. If I receive at least an incoming call sip message, it means that possibility 2 is out. I did not receive any messages from the server side, so the test was inconclusive.

Thus, I decided to focus on the possible points of failure:
1. The 2Wire's built in SIP ALG
2. Asterisk's lack of STUN or other forms of UDP transversal
3. SingTel's side possibly banning certain clients

The most probable cause could be the ALG. The ALG (from what I think I know, correct me if I'm wrong) is supposed to automatically translate the source and destination IP of every SIP and packet it sees passing through it so that LAN addresses are substituted with WAN addresses when the packets passes through to the internet. That is the reason why SIP ALG must be turned off if you wish to use mioMobile at home; as if I'm not wrong mioMobile tries to communicate to the internal server address first before trying voip.singtel.com (or is it? Someone confirm!). Thus the ALG's automated substitution of WAN address will cause the return address of all packets sent by the handset to be invalid.

This leads me to guess that the handset on mioMobile will be assigned an address in the server's subnet, thus allowing direct addressing between the handset and server without using the ALG. From here, one possibility of resolving this problem could be to try to place Asterisk on the same subnet as the server.

However, a contradiction occurs. So please, if you can provide any details, please do so:

Based on the above deductions, the ALG may seem useless as it will only be useful when the SIP server is on the WAN (internet) side. However, it does not seem so as turning on the ALG allows incoming packets from the server to be received properly in Asterisk when properly registered. I will need somebody with mioMobile to test a few scenarios out before I can resolve this contradiction, most probably its because the ALG is smart enough to differentiate between different network segments instead of just WAN and LAN.

However, assuming the contradiction is resolved, and based on the fact that turning on ALG allows successful passage of SIP packets from the server to Asterisk, I can weakly conclude that the cause of the sudden loss of response is due to the ALG losing track of where everything is.

For the ALG to be able to translate SIP packets properly, it must be able to determine which local address the SIP packets should go to. This cannot be fixed as the ALG is supposed to be able to handle more than one client on more than one address/machine (unlike NAT). Thus, one method of keeping track of which packet goes where is to keep track of the SIP session IDs and/or call IDs, as well as registration information, and binding them to the originating address (similar to UPnP). Thus, my conclusion is based on the assumption that either by error or design flaw on the ALG or Asterisk or the server's side, the tracking is lost, and the ALG no longer knows how to forward the packets to the proper places. This is what I believe to be the most likely cause for the problems.

Then why do softphones such as EyeBeam and SJPhone work? The reason is that they support STUN (Simple Transversal of UDP over NAT), which makes use of UDP hole punching to initiate a connection between the server and the client. Asterisk does not have support for this, and thus relies on SIP ALG or proper addressing. A resolution would be to try out CallWeaver and see if STUN solves everything. Otherwise, I'll have to properly address Asterisk.

The final point of failure is SingTel attempting to ban non 2Wire clients. I don't think this had happened yet, but just in case, I've modified asterisk to report itself as a 2Wire client. However, differences in packet formats may still give me away. Anyway, it may already be happening as they've started banning residential phone numbers from registering on voip.singtel.com.

TLDR: Possible causes of the problem I experience are, problems with the SIP ALG, addressing issues related to the SIP ALG, lack of STUN support in Asterisk, SingTel banning Asterisk.
TLDR2: Possible respective solutions are, make use of STUN to replace SIP ALG, proper addressing of Asterisk, make use of third party implementations of STUN in Asterisk, pretend to be an official 2Wire client.
TLDR3: Please help me if you can, any amount of information would help.

[liangtam]

Have you disabled 2Wire's inbuilt VoIP feature from trying to connect and gain foothold in managing of the voice?

[seowbin]

1st, disable the ALG and try see how?

singtel is user of broadsoft SBC, i help out in one of the pbx testing before

not sure whether this apply to home user, but for their business sip trunk user they are using that, likely it's the same case

inside the SBC , it actually support different type of 'profile', like generic , avaya, etc etc...
since it happen suddenly, it might be due to the applying of the '2wire' profile which behave differently compare to generic sip equipment

if so, you need to tweak ur asterisk to behave like 1 liao :s13:
---------------------------

2ndly, do make sure ur sip gateway have correct domain configure,etc cause softphone actually support this some sort of sip dns lookup, maybe you can obverse how the software react, maybe you can tweak ur asterisk.

Lastly i think asterisk support some sort of NAT if me not wrong,whereby you can tell your server what's ur wan ip is, etc.

i'm not very good at it, if you are good. HELP me on it also :s34: now i face some problem when i try to connect to some cisco callmanager product :(

[xieliwei]

Just thought I'll update this on how I solved my problem two posts above, since I had to waste time writing a patch when a simple configuration statement could have done it.

The reason why authentication fails for voip.singtel.com is because of Asterisk's default lack of support for multi line statements. After tracing the SIP conversation between the two, it seems that voip.singtel.com has spread its authentication statement over a few lines, the first split is right before the nonce. Thus Asterisk was unable to read the next few lines and thus could not obtain the correct nonce required to generate the challenge response for authentication. Thus the response generated is wrong and authentication fails.

This is where I wrote a patch to deal with the multiple lines. Luckily for you, if you are using a recent version of Asterisk, pedantic checking of SIP packets will deal with the multiple lines. Just add "pedantic = yes" (without quotes) to the [general] context of sip.conf.

Still can't figure out why the internal IP doesn't work though. Probably the complex layout of my network, or they're really trying to filter me. (Or because I didn't really bother looking into that after fixing this)

Hope this helps.

[xieliwei]

1st, disable the ALG and try see how?
Tried that, didn't change anything. The ALG only works in translating SIP packets going out to the internet, but like you mentioned, it can be done in Asterisk.

singtel is user of broadsoft SBC, i help out in one of the pbx testing before
For the mio systems, they're using Huawei SoftX3000 softswitches. Those things are well known to be common AND non-standards-compliance.

2ndly, do make sure ur sip gateway have correct domain configure,etc cause softphone actually support this some sort of sip dns lookup, maybe you can obverse how the software react, maybe you can tweak ur asterisk.
I suppose you mean STUN and domain support? I know in the 2Wire ATA they use moip.dect as domain, but setting it or not didn't change anything. Also, singtel runs a STUN server for their mioMobile users.

Lastly i think asterisk support some sort of NAT if me not wrong,whereby you can tell your server what's ur wan ip is, etc.
Yes, specify your IP address or hostname with externhost and optionally a refresh time with externrefresh. Then use nat = yes to activate it for the required peer.

i'm not very good at it, if you are good. HELP me on it also :s34: now i face some problem when i try to connect to some cisco callmanager product :(
If its not too late, I'll see if my limited knowledge is of any use :P

[seowbin]

lose interest in trying that out liao :s13:

so long already :D

[liangtam]

Moi ish more interested in... whether if mio Voice does actually work with normal ATAs instead.
But I guess not?

[bizcon]

pap2 can support mio voice. Does any1 tried using pap2 to fax via mio account??..... i cannot get it to work.

[liangtam]

You mean u can still use external sip ata to make and receive call, especially without the 2wire in normal gateway mode?

[bizcon]

pap2 still can work with mio voice with normal gateway. with or without using mio line. but I cannot get fax to work well.

My intention is to bring pap2 to another location and use it for fax. And suggestion how to get it work?

[outer92]

pap2 still can work with mio voice with normal gateway. with or without using mio line. but I cannot get fax to work well.

My intention is to bring pap2 to another location and use it for fax. And suggestion how to get it work?

how to config the PAP2 ?

[liangtam]

was wondering actually, was it the codec or some sip ata cannot handle fax.

same setting
voip.singtel.com:5060
user
pass

[seowbin]

it should be the codec... need g711
and also support for t.38 protocol i think.... most importantly, the analog fax need to support that protocol too :s13:

[bizcon]

I tried using g711. Most of the time it just hang during negotiating. Sometimes, it manage to transmit, but halfway fail.

I am using my PC via modem to fax out. Using 2wire no problem at all. So not the modem problem. I even tried tunning the modem speed.

Does any1 know if upgrading pap2t firmware helps? What is the latest firmware?

[bizcon]

how to config the PAP2 ?

Hi, you might want to read

http://cgi.ebay.com.sg/FREE-GUIDE-LINKSYS-PAP2T-PAP2-ATA-2FXS-Phone-Adapter_W0QQitemZ200153179165QQihZ010QQcategoryZ3706QQssPageNameZWDVWQQrdZ1QQcmdZViewItem

[seowbin]

go google for pap2t t.38 support or something like that..

likely 2wire supports that, which is fax over ip

[bizcon]

the strange thing is pap2t support fax but did not indicate it support t.38. probably is some passthru.

I also happen to find http://www.kapanga.net/ip/home.cfm when I google for some answer.

This softphone claim to support t.38 fax. But I cannot manage to register my account. Any1 wanna confirm?

[james_zc]

Conclusion:

Has anyone setup singnet BB & mio voice successfully without mio box??

[xieliwei]

Hello people, just noticed a response in this thread and checked it out. It seems that you have problems using fax on the PAP2?

I do not have a PAP2 with me, but you may want to consider these points:
1. T.38 requires support on the server side of the system and as far as I know, SingNet does not and has no need to support this.
2. Use G.711, and if you can specify, ulaw, since fax is sensitive to artifacts and frequency losses due to compression. In fact, I don't think any codec other than ulaw is usable with mioVoice as I don't see their server advertising other codecs.
3. Fax is also very sensitive to latency and jitter as well, this might be the problem you're facing.
* Try using the internal 10.x.x.x address and not voip.singtel.com
* Try closing all bandwidth intensive applications, and if you have mioTV, turn it off too. If it works after that, then its a bandwidth issue.
* The bandwidth issue might be a bit hard to tackle since the mio box has no provisions of QoS, it is only coded to prioritise traffic for mio services (I think)

I've had no problems sending or receiving faxes (even sending and receiving multiple faxes at the same time) with my asterisk setup, and I'm using voip.singtel.com, so I'm not very sure why there's a problem on your side, but I hope those points can help isolate the problem.

A little reading up tells me that there were problems with fax pass-through before firmware 2.0.13(LSb), but I don't think your PAP2 is that old (2005).

Conclusion:

Has anyone setup singnet BB & mio voice successfully without mio box??
Do you mean totally without the box or using an external ATA? The answer to both is yes anyway.

If you have mioTV however, its recommended to keep the mio box, the steps to configure the required ATM circuits and network bridges requires at least a linux-based router. I may document it here if I have the time.

I have a question however, why do you all want to get rid of the mio box totally?

[liangtam]

mio Box not lasting nor cheap mah.

[ngbc]

I have a question however, why do you all want to get rid of the mio box totally?

For me, I feel that the mio box's lack of uPnP, DDNS & url blocking support, which are sort of standard in other routers, is a turn-off. I can't even upgrade firmware myself. Furthermore, there were reports by users here that their power adapter became faulty after 1 year+ usage. I just recontracted in March this year and the modem is 1.5 yr old, so I'm worry....

If you have time, please do a detailed write-up here. Thanks in advance. It would be useful when my modem/adapter go up lorry, and give me a reason to play with new toy :look:

[cozi]

Just thought I'll update this on how I solved my problem two posts above, since I had to waste time writing a patch when a simple configuration statement could have done it.

The reason why authentication fails for voip.singtel.com is because of Asterisk's default lack of support for multi line statements. After tracing the SIP conversation between the two, it seems that voip.singtel.com has spread its authentication statement over a few lines, the first split is right before the nonce. Thus Asterisk was unable to read the next few lines and thus could not obtain the correct nonce required to generate the challenge response for authentication. Thus the response generated is wrong and authentication fails.

This is where I wrote a patch to deal with the multiple lines. Luckily for you, if you are using a recent version of Asterisk, pedantic checking of SIP packets will deal with the multiple lines. Just add "pedantic = yes" (without quotes) to the [general] context of sip.conf.

Still can't figure out why the internal IP doesn't work though. Probably the complex layout of my network, or they're really trying to filter me. (Or because I didn't really bother looking into that after fixing this)

Hope this helps.

I managed to get Asterisk to register with voip.singtel.com. However, i am still not able to route a call out using Asterisk (X-lite works with the same credential) I am looking that the debug information for the SIP communications. Attached is the debug information between singtel and Asterisk.

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
<-- SIP read from 202.166.85.196:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 116.15.29.XXX:5060;branch=z9hG4bK44c18643;rport=5060
Call-ID: 6531bebf1028ca32341c0ea53202b394@116.15.29.XXX
From: "Mio Mobile"<sip:XXXXXXXX@116.15.29.XXX>;tag=as4d9f14a7
To: <sip:XXXXXXXX@voip.singtel.com>
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---

<-- SIP read from 202.166.85.196:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 116.15.29.XXX:5060;branch=z9hG4bK44c18643;rport=5060
Call-ID: 6531bebf1028ca32341c0ea53202b394@116.15.29.XXX
From: "Mio Mobile"<sip:XXXXXXXX@116.15.29.XXX>;tag=as4d9f14a7
To: <sip:XXXXXXXX@voip.singtel.com>;tag=8c6835fe
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="singtel",nonce="1:25:35:34127", stale=false,algorithm=MD5
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0


--- (9 headers 0 lines) ---

Transmitting (NAT) to 202.166.85.196:5060:
ACK sip:XXXXXXXX@voip.singtel.com SIP/2.0
Via: SIP/2.0/UDP 116.15.29.XXX:5060;branch=z9hG4bK44c18643;rport
From: "Mio Mobile" <sip:XXXXXXXX@116.15.29.XXX>;tag=as4d9f14a7
To: <sip:XXXXXXXX@voip.singtel.com>;tag=8c6835fe
Contact: <sip:XXXXXXXX@116.15.29.XXX>
Call-ID: 6531bebf1028ca32341c0ea53202b394@116.15.29.XXX
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

We're at 116.15.29.XXX port 5066

Adding codec 0x2 (gsm) to SDP

Adding codec 0x4 (ulaw) to SDP

Adding codec 0x400 (ilbc) to SDP

Adding codec 0x8 (alaw) to SDP

Adding codec 0x200 (speex) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (NAT) to 202.166.85.196:5060:
INVITE sip:XXXXXXXX@voip.singtel.com SIP/2.0
Via: SIP/2.0/UDP 116.15.29.XXX:5060;branch=z9hG4bK6c1b07e9;rport
From: "Mio Mobile" <sip:XXXXXXXX@116.15.29.XXX>;tag=as4d9f14a7
To: <sip:XXXXXXXX@voip.singtel.com>
Contact: <sip:XXXXXXXX@116.15.29.XXX>
Call-ID: 6531bebf1028ca32341c0ea53202b394@116.15.29.XXX
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="XXXXXXXX", realm="singtel", algorithm=MD5, uri="sip:XXXXXXXX@voip.singtel.com", nonce="1:25:35:34127", response="c5765a71713b931f77221f5d708a57cc", opaque=""
Date: Thu, 18 Dec 2008 17:25:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 3936 3937 IN IP4 116.15.29.XXX
s=session
c=IN IP4 116.15.29.XXX
t=0 0
m=audio 5066 RTP/AVP 3 0 97 8 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 202.166.85.196:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 116.15.29.XXX:5060;branch=z9hG4bK6c1b07e9;rport=5060
Call-ID: 6531bebf1028ca32341c0ea53202b394@116.15.29.XXX
From: "Mio Mobile"<sip:XXXXXXXX@116.15.29.XXX>;tag=as4d9f14a7
To: <sip:XXXXXXXX@voip.singtel.com>
CSeq: 103 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---

<-- SIP read from 202.166.85.196:5060:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 116.15.29.XXX:5060;branch=z9hG4bK6c1b07e9;rport=5060
Call-ID: 6531bebf1028ca32341c0ea53202b394@116.15.29.XXX
From: "Mio Mobile"<sip:XXXXXXXX@116.15.29.XXX>;tag=as4d9f14a7
To: <sip:XXXXXXXX@voip.singtel.com>;tag=d798fb38
CSeq: 103 INVITE
Reason: Q.850;cause=98;text="Message not compatible with call state or message type non-existent or not implemented"
Content-Length: 0


--- (8 headers 0 lines) ---

-- Got SIP response 500 "Server Internal Error" back from 202.166.85.196

Transmitting (NAT) to 202.166.85.196:5060:
ACK sip:XXXXXXXX@voip.singtel.com SIP/2.0
Via: SIP/2.0/UDP 116.15.29.XXX:5060;branch=z9hG4bK6c1b07e9;rport
From: "Mio Mobile" <sip:XXXXXXXX@116.15.29.XXX>;tag=as4d9f14a7
To: <sip:XXXXXXXX@voip.singtel.com>;tag=d798fb38
Contact: <sip:XXXXXXXX@116.15.29.XXX>
Call-ID: 6531bebf1028ca32341c0ea53202b394@116.15.29.XXX
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

-- SIP/XXXXXXXX-00100dc8 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

Anyone have any idea what "Message not compatible with call state or message type non-existent or not implemented" means?

And how to fix it?

[limlaw]

Just thought I'll update this on how I solved my problem two posts above, since I had to waste time writing a patch when a simple configuration statement could have done it.

The reason why authentication fails for voip.singtel.com is because of Asterisk's default lack of support for multi line statements. After tracing the SIP conversation between the two, it seems that voip.singtel.com has spread its authentication statement over a few lines, the first split is right before the nonce. Thus Asterisk was unable to read the next few lines and thus could not obtain the correct nonce required to generate the challenge response for authentication. Thus the response generated is wrong and authentication fails.

This is where I wrote a patch to deal with the multiple lines. Luckily for you, if you are using a recent version of Asterisk, pedantic checking of SIP packets will deal with the multiple lines. Just add "pedantic = yes" (without quotes) to the [general] context of sip.conf.

Still can't figure out why the internal IP doesn't work though. Probably the complex layout of my network, or they're really trying to filter me. (Or because I didn't really bother looking into that after fixing this)

Hope this helps.
HI xieliwei, I believe you are guru in asterisk, guess you managed to get MIO voice to work asterisk, care to share with me the magic which portions to edit so it can make out going call successfully, I am having SIP/2.0 500 Server Internal Error. Reason: Q.850;cause=98;text="Message not compatible with call state or message type non-existent or not implemented" as what cozi had mentioned... You help would be greatly appreciated.

[ch1234]

Hi,
I just got a linksys ADSL2 Gateway with 2 phone port from SLS. after doing the configuration shown I can make phone call out. However when a call comes in I can hear the other party but the other party cannot hear me. Can any expert tell me what could be the problem?? Thanks??
Wireless-G ADSL2 Gateway With 2 Phone Ports
Wi
Wireless-G ADSL2 Gateway With 2 Phone Ports
reless-G ADSL2 Gateway With 2 Phone Ports

[PDA_lover]

friends,

i am gonna jump ship from LaggyHub to Singtel and need some help from the experts here.

i have a CIsco 1841 router which i want to use for BB (preferrably 25MB) from singtel and at the same time if i can use their MIO voice (Voip for landline and not mobile) it would be great. reading this thread i saw some one telling that we can remove 2wire completely if cisco router is in place, pls enlighten me if this is possible

[liangtam]

What has a cisco router got to do with whether if you can use mio Voice :s11:

[PDA_lover]

it is not possible to take the 2wire out of the picture unless you have a fully configurable router like cisco.

my 2wire is now in bridged mode, and connected to the D-Link DIR-655, which serves the rest of the network.

The MioTV, however, is connected to the 2Wire.

The interesting thing is, regardless of which port I plug the DIR or the MIO into on the 2Wire, its able to detect them and work accordingly, so seems like they have managed to make it dynamic.

So I am wondering, if I plug an ATA into the 2Wire, will it be able to detect automatically, since the phone port is simply another physical/logical port to the 2Wire.

Bro hangyong, perhanps you could help me answer my question

[seowbin]

your 1841 got atm module or not?

if not, how to totally remove ur 2wire modem?

1841 don't support cme also if me not wrong,so don't expect much voice feature in it...

[PDA_lover]

oh, that mean i will have to live with stuppid 2 Wire :-(

[liangtam]

oh, that mean i will have to live with stuppid 2 Wire :-(

No
*10 char*

[PDA_lover]

No means? is there any way i can avoid 2 wire?

[liangtam]

No means? is there any way i can avoid 2 wire?

..........

[ch1234]

I also wanna get rid of the 2wire. I have a linksys router with VOIP which I wanna replace the 2wire. Can any expert teach me how to??

[liangtam]

I also wanna get rid of the 2wire. I have a linksys router with VOIP which I wanna replace the 2wire. Can any expert teach me how to??

Look for firmware update, since it may be unlikely on the codec.

[ch1234]

ok will try. thanks

[ch1234]

Look for firmware update, since it may be unlikely on the codec.

Tried updated to the latest firmware liao. Problem still there. Guess still got to stick to the stupid 2wire liao. Haiz......:(:(

[Icyen~]

so this means after following the steps, my phone can still work without having to turn on the modem?

[liangtam]

you still need internet connection of cos.

[bizcon]

yes, please throw me some light to get asterik to register with MIO.

I am able to to register with voipuser......just do not know the combination outgoing string for MIO.

[slowstatic]

Upz for more advise.... :(

[tanyj82]

I try to connect using xlite, it keep giving me service unavailable.

Can someone share the xlite setting here.
I cant find the place to insert the server ip

[weiyang76]

it works for me with xlite. (:

[OKK77]

mioVoice will certainly not work behind another all-in-one router? *sigh*