Francesco Peeters wrote:
NM! Found out this only happens on a single extension, and that one was using IAX... Changed it to SIP as well and got ringing there too!Francesco Peeters wrote:Does anybody else see the same behavior for VoipBuster connections?When I trace one of the other SIP peers, I see it sends this message: ---------------------------------------------------------------------- <--- SIP read from 82.101.62.99:5060 ---> SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: [email protected] Contact: <sip:82.101.62.99:5060> Content-Type: application/sdp CSeq: 103 INVITE From: "**********" <sip:*******[email protected]>;tag=as70e84199 Record-Route: <sip:82.101.62.115;lr;r2=on;ftag=as70e84199>,<sip:82.101.63.5;lr;r2=on;ftag=as70e84199> Server: Cirpack/v4.41b (gw_sip) To: <sip:0031*****[email protected]>;tag=00-08168-044b6f36-245cd72c7 Via: SIP/2.0/UDP ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92 Content-Length: 182 v=0 o=cp10 125193221174 125193221174 IN IP4 82.101.62.66 s=SIP Call c=IN IP4 194.109.8.2 t=0 0 m=audio 36984 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=sendrecv <-------------> --- (12 headers 10 lines) --- Found RTP audio format 8 Peer audio RTP is at port 194.109.8.2:36984 Found audio description format PCMA for ID 8 Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.109.8.2:36984 -- SIP/*********-089ca9b8 is ringing -- SIP/*********-089ca9b8 is making progress passing it to IAX2/2104-2287 Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 82.101.62.99:5060: CANCEL sip:0031******[email protected] SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport From: "**********" <sip:******[email protected]>;tag=as70e84199 To: <sip:0031******[email protected]> Call-ID: [email protected] CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ---------------------------------------------------------------------- However when I dial exactly the same from VoipBuster, I see this instead: ---------------------------------------------------------------------- <--- SIP read from 77.72.169.129:5060 ---> SIP/2.0 183 Session progress Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport From: "*********" <sip:******[email protected]>;tag=as1374705a To: <sip:0031******[email protected]>;tag=120113ac4a54a269af9e2c Contact: sip:0031******[email protected]:5060 Call-ID: [email protected] CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 162 v=0 o=********* 1251932194 1251932194 IN IP4 194.221.62.33 s=SIP Call c=IN IP4 194.221.62.33 t=0 0 m=audio 8958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 194.221.62.33:8958 Found audio description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.221.62.33:8958 -- SIP/********-089dc538 is making progress passing it to IAX2/2104-8077 == Connect attempt from '127.0.0.1' unable to authenticate Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 77.72.169.129:5060: CANCEL sip:0031******[email protected] SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport From: "**********" <sip:*******[email protected]>;tag=as1374705a To: <sip:0031******[email protected]> Call-ID: [email protected] CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ---------------------------------------------------------------------- As you can see, there are different packets being sent, and in the 2nd case, there is no "is ringing" message, which is rather irritating... Any suggestions would be appreciated... TIABTW: I am talking about the ringtone the caller should hear... The other side is ringing, and calls are established just fine, but it is very irritating to hear nothing until the call either fails or gets picked up... -- FP |
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