Mark Fisher
2006-May-10 04:49 UTC
[Asterisk-Users] No audio in either direction on Zap -> SIP or SIP -> Zap calls
Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and zap. At the packet level, I can see the packets leave the sip phone and get to asterisk, but asterisk doesnt attempt to send any rtp back toward the phone. This is the same with iptables running or not. If anyone could give any suggestions they would be gratefully received. Ive included sections of relevant config files below. At the very bottom Ive pasted an extract from the * CLI with real numbers changed for fake ones, and I notice that a native bridge doesnt actually get mentioned. zapata.conf: [channels] language=en context=zap signalling=pri_cpe switchtype=euroisdn pridialplan=unknown calleridusecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes musiconhold=default group = 3 channel => 63-77,79-93 sip.conf: disallow=all allow=alaw allow=ulaw [sip_proxy] type=friend context=foobar host=sip.proxy.net defualtip=x.x.x.x port=5060 disallow=all allow=ulaw canreinvite=no nat=no extensions.conf: [sip] exten => _X.,1,Dial(Zap/g2/01234567890) [zap] exten => _X.,1,Dial(SIP/1234@sip_proxy) Asterisk CLI: -- Accepting call from '1234' to '5678' on channel 0/23, span 3 -- Executing SetTransferCapability("Zap/85-1", "SPEECH") in new stack -- Setting transfer capability to: 0x00 - SPEECH. -- Executing Dial("Zap/85-1", "SIP/3456@sip_proxy") in new stack -- Called 3456@sip_proxy -- SIP/sip_proxy-57c5 is ringing -- SIP/sip_proxy-57c5 answered Zap/85-1 -- Channel 0/23, span 3 got hangup request == Spawn extension (zap, 5678, 2) exited non-zero on 'Zap/85-1' -- Hungup 'Zap/85-1' -- Mark Fisher