search for: iaxtest

Displaying 4 results from an estimated 4 matches for "iaxtest".

2012 Mar 09
2
dreaded one-way audio with nat=yes
I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten
2003 Aug 27
0
Registering via IAX2 succeeds, but bridging to the registered peer fails
...ate*] cannot register via IAX2 correctly while [public*] is running. Status remains UNKNOWN even after minutes, calls from [public*] to [private*] are not possible. Console output of [public*]: | *CLI> iax2 show peers | Name/Username Host Mask Port Status | iaxtest/iaxtest (Unspecified) (D) 255.255.255.255 0 UNKNOWN | -- Registered 'iaxtest' (AUTHENTICATED) at 217.187.160.28:4569 | *CLI> iax2 show peers | Name/Username Host Mask Port Status | iaxtest/iaxtest 217.187.160.28 (D) 255.255.255.255...
2007 Jun 26
1
Asterisk to Cisco 2600 GW DTMF Not Working
...8fa(1024) Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge: Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call: Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on 'SIP/53061-92e0' The call drops. If I enable ILBC codec with Asterisk, here is what I get: == Forcing Marker bit, because SSRC has changed Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multipl...
2007 Jun 27
0
Asterisk to Cisco 2600 GW DTMF Not Working, Working now
...:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge: > Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible > Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call: > Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa > == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on > 'SIP/53061-92e0' > > The call drops. > > If I enable ILBC codec with Asterisk, here is what I get: > > == Forcing Marker bit, because SSRC has changed > Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein: > Huh...