John Millican
2006-Apr-02 10:38 UTC
[Asterisk-Users] no audio between sip channels * 1.2.6
Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the other and no sound. Okay must have screwed something up. checked sip.conf all looked good. Okay good time to go to 1.2.6, still no audio. All phones ring and answer but no audio. the last thing that apears on the console is "attempting native bridge of sip/677-xxxx and sip/699-xxxx" below is a debug of a call and sip.conf. Each channel on the PAP2's is set to a different port 5060 through 5063. I can call in to any phone and all is good, use any phone to call to POTS line and back in on second POTS line and all is good. I have been looking through the archive of the mail list that I keep and have not found anything to fix my problems yet. i have transfered the registration of both PAP2's to a 1.2.0 system that I have and everything works as it should. moved 1.2.0 configs to 1.2.6 box and again no audio between sip channels. *CLI> sip debug SIP Debugging enabled *CLI> <-- SIP read from 192.168.1.200:5060: INVITE sip:699@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10> Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 101 INVITE Max-Forwards: 70 Contact: John Millican <sip:677@192.168.1.200:5060> Expires: 240 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 235 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 143361 143361 IN IP4 192.168.1.200 s=- c=IN IP4 192.168.1.200 t=0 0 m=audio 16410 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 12 lines)--- Using INVITE request as basis request - 23a92fd3-a3463cc9@192.168.1.200 Sending to 192.168.1.200 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941;received=192.168.1.200 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10>;tag=as0767a869 Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:699@192.168.1.10> Proxy-Authenticate: Digest realm="asterisk", nonce="6e91851e" Content-Length: 0 --- Scheduling destruction of call '23a92fd3-a3463cc9@192.168.1.200' in 15000 ms Found user '677' <-- SIP read from 192.168.1.200:5060: ACK sip:699@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10>;tag=as0767a869 Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 101 ACK Max-Forwards: 70 Contact: John Millican <sip:677@192.168.1.200:5060> User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 0 --- (10 headers 0 lines)--- <-- SIP read from 192.168.1.200:5060: INVITE sip:699@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10> Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="677",realm="asterisk",nonce="6e91851e",uri="sip:699@192.168.1.10",algorithm=MD5,response="121d27cf19808e8a097930f0f969d3d7" Contact: John Millican <sip:677@192.168.1.200:5060> Expires: 240 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 235 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 143361 143361 IN IP4 192.168.1.200 s=- c=IN IP4 192.168.1.200 t=0 0 m=audio 16410 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 12 lines)--- Using INVITE request as basis request - 23a92fd3-a3463cc9@192.168.1.200 Sending to 192.168.1.200 : 5060 (non-NAT) Found user '677' Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.200:16410 Found description format PCMU Found description format NSE Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 699 in pap2 (domain 192.168.1.10) list_route: hop: <sip:677@192.168.1.200:5060> Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1;received=192.168.1.200 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10> Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:699@192.168.1.10> Content-Length: 0 --- -- Executing Dial("SIP/677-bd65", "sip/699") in new stack We're at 192.168.1.10 port 12042 Adding codec 0x4 (ulaw) to SDP 13 headers, 8 lines Reliably Transmitting (no NAT) to 192.168.1.201:5062: INVITE sip:699@192.168.1.201:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0;rport From: "John Millican" <sip:677@pap2>;tag=as604497e3 To: <sip:699@192.168.1.201:5062> Contact: <sip:677@192.168.1.10> Call-ID: 400784d110e4e314373334be31ee3576@pap2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 02 Apr 2006 17:10:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 158 v=0 o=root 5051 5051 IN IP4 192.168.1.10 s=session c=IN IP4 192.168.1.10 t=0 0 m=audio 12042 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- -- Called 699 <-- SIP read from 192.168.1.201:5062: SIP/2.0 100 Trying To: <sip:699@192.168.1.201:5062> From: "John Millican" <sip:677@pap2>;tag=as604497e3 Call-ID: 400784d110e4e314373334be31ee3576@pap2 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 --- (8 headers 0 lines)--- <-- SIP read from 192.168.1.201:5062: SIP/2.0 180 Ringing To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0 From: "John Millican" <sip:677@pap2>;tag=as604497e3 Call-ID: 400784d110e4e314373334be31ee3576@pap2 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0 Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 --- (8 headers 0 lines)--- -- SIP/699-8c1a is ringing Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1;received=192.168.1.200 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10>;tag=as7abbbe62 Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:699@192.168.1.10> Content-Length: 0 --- <-- SIP read from 192.168.1.201:5062: SIP/2.0 200 OK To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0 From: "John Millican" <sip:677@pap2>;tag=as604497e3 Call-ID: 400784d110e4e314373334be31ee3576@pap2 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0 Contact: Sentinel Communications <sip:699@192.168.1.201:5062> Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 233 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 13282 13282 IN IP4 192.168.1.201 s=- c=IN IP4 192.168.1.201 t=0 0 m=audio 16444 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (12 headers 12 lines)--- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.201:16444 Found description format PCMU Found description format NSE Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:699@192.168.1.201:5062> set_destination: Parsing <sip:699@192.168.1.201:5062> for address/port to send to set_destination: set destination to 192.168.1.201, port 5062 Transmitting (no NAT) to 192.168.1.201:5062: ACK sip:699@192.168.1.201:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3db4b100;rport From: "John Millican" <sip:677@pap2>;tag=as604497e3 To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0 Contact: <sip:677@192.168.1.10> Call-ID: 400784d110e4e314373334be31ee3576@pap2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/699-8c1a answered SIP/677-bd65 We're at 192.168.1.10 port 11982 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1;received=192.168.1.200 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10>;tag=as7abbbe62 Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:699@192.168.1.10> Content-Type: application/sdp Content-Length: 214 v=0 o=root 5051 5051 IN IP4 192.168.1.10 s=session c=IN IP4 192.168.1.10 t=0 0 m=audio 11982 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/677-bd65 and SIP/699-8c1a <-- SIP read from 192.168.1.200:5060: ACK sip:699@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-e7eb5b44 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10>;tag=as7abbbe62 Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="677",realm="asterisk",nonce="6e91851e",uri="sip:699@192.168.1.10",algorithm=MD5,response="36f62fd39cac4190c11bc2b1fa7cb31b" Contact: John Millican <sip:677@192.168.1.200:5060> User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 0 --- (11 headers 0 lines)--- <-- SIP read from 192.168.1.200:5060: BYE sip:699@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-cd302779 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10>;tag=as7abbbe62 Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="677",realm="asterisk",nonce="6e91851e",uri="sip:699@192.168.1.10",algorithm=MD5,response="fe5a3ed0d6a5e2b18071b0784b4c0c80" User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 0 --- (10 headers 0 lines)--- Sending to 192.168.1.200 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-cd302779;received=192.168.1.200 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10>;tag=as7abbbe62 Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:699@192.168.1.10> Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing <sip:699@192.168.1.201:5062> for address/port to send to set_destination: set destination to 192.168.1.201, port 5062 Reliably Transmitting (no NAT) to 192.168.1.201:5062: BYE sip:699@192.168.1.201:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69a7640a;rport From: "John Millican" <sip:677@pap2>;tag=as604497e3 To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0 Contact: <sip:677@192.168.1.10> Call-ID: 400784d110e4e314373334be31ee3576@pap2 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (pap2, 699, 1) exited non-zero on 'SIP/677-bd65' <-- SIP read from 192.168.1.201:5062: SIP/2.0 200 OK To: <sip:699@192.168.1.201:5062>;tag=4eb3c3babbd8e5efi0 From: "John Millican" <sip:677@pap2>;tag=as604497e3 Call-ID: 400784d110e4e314373334be31ee3576@pap2 CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69a7640a Server: Linksys/PAP2-3.1.9(LSc) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '400784d110e4e314373334be31ee3576@pap2' Destroying call '23a92fd3-a3463cc9@192.168.1.200' *CLI> sip no debug [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes [677] type=friend ;username=John Millican ;fromuser=677 secret=hell99 host=dynamic callerid=John <677> fromdomain=pap2 context=pap2 dtmfmode=inband disallow=all allow=ulaw mailbox=4700 canreinvite=no nat=no [666] type=friend secret=xxxxxx host=dynamic callerid=Kelly <666> fromdomain=pap2 context=pap2 dtmfmode=inband disallow=all allow=ulaw mailbox=4800 canreinvite=no nat=no [699] type=friend secret=sentinel1 host=dynamic callerid=Sentinel 1 <603xxxxxxx> fromdomain=pap2 context=pap2 dtmfmode=inband disallow=all allow=ulaw mailbox=9010 canreinvite=no nat=no <flame suit> If I have missed something stupid feel free to rub my nose in it</flame suit> Any help would be greatly appreciated. John M