Trixbox - no incoming audio with Voiptalk

Sep 01, 2006 11 Replies

Can anyone shed any light on this? I've set up 3 trunks - Voiptalk, Voipfone and Sipgate. With Voipfone and Sipgate I get 2 way audio but with Voiptalk the other party can hear me but I can't hear them.



Voiptalk CS won't support trixbox at the moment, although they helpfully sent me a config that another customer had used successfully. If the problem was across all Carriers I'd suspect a NAT or port forwarding issue but with just one I'm scratching my head a bit.



TIA



Liam


Are you using SIP on Sipgate & Voipfone but IAX with Voiptalk?

If so, IAX requires a different (single) port to operate. Have you forwarded TCP / UDP or both?

Looking at their site it appears they support Asterisk, tell them your using Asterisk and get them to tell you what should be in the *.conf files and what ports should be set. Make a note, then add it to the freepbx interface later (will overight *.conf files).

Let us know how you get on.

Calling Jono, What did you do to get Voiptalk working?

Liam's tried my settings which work fine here, but gets this one-way audio issue.

Funnily, I tried my settings from my normal Trixbox in my experimental Vmware version, and I'm getting this same issue......

Sorry to answer my own post, but it now works on the vmware version too.

In config edit->sip_nat.conf I had to add:

externip=myfixedIP (not my dyndnsname) localnet=192.168.1.0/255.255.255.0 (this would be the same on your

716g)

Calls in and out are working fine

Hi Paul,

I had the above in sip.conf under [general] but with the dyndns name. I removed it and put it in sip_nat.conf with the IP address. Still no joy. Re-booted linux/asterisk and made 1 call on Voiptalk - still one-way audio but subsequent attempts to call on this trunk get the asterisk lady telling me it's busy...

I'm going out to buy some Strowger switches...

Liam

on 01/09/2006, paul123 supposed :

Alas, nothing. It's not a provider I use.

Aha, never mind, it's just that I had assumed you'd got it working - voiptalk had come up in an earlier thread (about the free signup for a geo number), where you'd had some probs.

Not to worry, we'll get there in the end.

Bugger. Yes, I was thinking "voipfone". You're right. Duh!

I had to play around with getting the inbound DID working, I can't say there was much wrong with the audio, however, here're my trunk settings:

PEER: allow=ulaw authuser=12345678 canreinvite=no context=from-pstn defaultexpirey=160 disallow=all fromdomain=voiptalk.org fromuser=12345678 host=voiptalk.org insecure=very maxexpirey=180 nat=yes secret=abcDefg type=friend username=12345678

User Context: ext-did

Incoming settings: allow=ulaw authuser=12345678 canreinvite=no context=ext-did defaultexpirey=160 disallow=all fromdomain=voiptalk.org fromuser=12345678 host=voiptalk.org insecure=very maxexpirey=180 nat=yes secret=abcDefg type=peer username=12345678

I've also got this in extensions_custom.conf: [custom-voiptalk] exten => 12345678,1,Goto(ext-did,12345678,1)

I have also created an inbound route for 12345678.

I think I have a bit more in sip.conf:

port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm context = from-trunk ;defaultexpirey = 600 ; include this only if necessary ;maxexpirey = 3600 ; include this only if necessary progressinband = yes dtmfmode=auto

callerid = Unknown externip = my.dyndns.hostname localnet=192.168.1.0/255.255.255.0 nat=yes

#include sip_nat.conf #include sip_custom.conf #include sip_additional.conf

You're not the only one to mix those two up. Easily done! I was recently in email contact with someone and I sent them to voipfone instead of voiptalk. Not surprisingly, they couldn't find what I was talking about.

Hmm.

I didn't write my configs myself! I got them here from.....paul123

......paul123, why are you asking me how I got voiptalk working, when you told me?

lol, :) yes, I know, but I thought that you had initial audio problems too and that maybe you'd had to adjust something to get it going OK. But now, looking back at that thread, I see that it was dialling format that was the problem.

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