One Way - 'Stopped Sounds'

[pablo2000]

Hi folks

I hope someone can give me a point in the right direction with a one-way audio problem.

I am dialling externally via an IAX2 trunk. All internal calls are absolutely fine (some echo but I'm not worried about that right now). All incoming calls are also OK. Also, if I connect my IAX softphone directly to the trunk, not via the * server, that's OK too.

When establishing an outgoing call via my telco partner the call setup is fine and connects to the other party, but no audio from the * extension goes to the other party. Audio back is fine however.

Console reports:
vici*CLI>
-- Accepting AUTHENTICATED call from 192.168.1.33:
> requested format = ulaw,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (ulaw),
> priority = mine
-- Executing AGI("IAX2/cc301-11185", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Set("IAX2/cc301-11185", "CALLERID(all)="*REMOVED*"") in new stack
-- Executing Dial("IAX2/cc301-11185", "IAX2/gradwell-out/0XXX850043|55|To") in new stack
-- Called gradwell-out/01XXX850043
-- Call accepted by 193.111.201.98 (format ulaw)
-- Format for call is ulaw
-- IAX2/gradwell-out-16384 is ringing
-- IAX2/gradwell-out-16384 is making progress passing it to IAX2/cc301-11185
-- IAX2/gradwell-out-16384 stopped sounds
-- IAX2/gradwell-out-16384 answered IAX2/cc301-11185
-- Hungup 'IAX2/gradwell-out-16384'
== Spawn extension (default, 901XXX850043, 3) exited non-zero on 'IAX2/cc301-11185'
-- Executing DeadAGI("IAX2/cc301-11185", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----8-----2") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---8-----2 completed, returning 0
-- Hungup 'IAX2/cc301-11185'

I have 'XX'd out some of the numbers, clearly, they are fully present in the output however.

Any ideas as to why the 'stopped sounds' is there - it clearly is the problem as it's the only real difference in the reporting from a correctly connected internal call.

All advice, however whacky, welcome

Pablo

[Op3r]

check if your firewall is up

[pablo2000]

Hi

Yes, checked firewall. Main firewall isn't an issue and stopped iptables running as well.

?????

P

[williamconley]

is the iax port through your firewall pointed to that soft phone you tested, and intercepting the sound on the way back and misleading it?

also: i note that during your description of the problem you referred to "the trunk" and "my telco partner" ... I would like to assume that the "trunk" in question was through the telco partner and you meant that you tested "the trunk" with a soft phone and it worked, then you tested "the trunk" with asterisk and it had one way audio ... but i'd like to verify it (before you get insulted, please remember that I am a help desk, and you have NO idea how many times I've had to clarify issues like this, clarity is extremely important).

[pablo2000]

William

Yes, your assumptions are spot on.

As to the port redirection, I have been careful to redirect between tests. Also, the sound 'back' works fine as do inbound calls which connect to the target extension without any issues at all.

Your help is greatly appreciated.

P

[williamconley]

Have you tried changing Codecs? What's your bandwidth? (Anyone ELSE using your bandwidth?)

[pablo2000]

I've tried ulaw and GSM both between the softphone and * and between the* and the trunk. When connecting to the trunk, the console output confirms that the selected codec is being used.

There are other people on the LAN and the WAN is 6Mbytes bandwidth. I don't think it's a traffic problem though.... It's too consistent.

What is the meaning of the 'sounds stopped' message - what is * doing at that point??

P

[pablo2000]

Help anybody????

[williamconley]

OK, Back to Basics:

1) Version (I know it's vicidialNOW, but let's do this right: which release of VicidialNOW and which Version of Vicidial is presently in it ...)

2) Hardware (just to be funny)

3) Any other software running on the box?

Then we can go into some "follow the white rabbit" stuff. Like: Can you post the sip/iax context you are using for this call (for the provider, it's only a few lines, please mask your user/pwd/host by replacing them with "USER" ... etc).

Also post your exten=> entry from the extensions for this dial pattern (once again, only a few lines, even shorter than the above context).

After that we'll dig in a little further, but these should always be covered because this is ordinarily where the problem lies.

[pablo2000]

William, many thanks for your offer of help. The problem is now resolved, however.

For the record, in case anyone searches this thread for a solution in the future. The problem was caused by having set

Trunk=yes

in the iax.conf entries for my outbound provider. Apparently my account was not setup to support trunking. If the setting is commented out or set to no the problem is resolved. I have now had my provider change my connection to support trunking - with 30 outbound lines open at any time I don't really want the overhead of individual streams!

Thanks again to all.

P

[williamconley]

That's new. Thanks for the "Closeout" post. Very interesting.