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I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G phones. Unit obtains SIP trunking
from an ITSP (server has no PCI boards). *8 Call Pickup works fine from any of the phones connected using the Linksys SPA2102. *8 Call Pickup does not work from the Cisco 7940G phones (chan_sip.c:13977 handle_request_invite: Nothing to pick up for [email protected]) What could the difference be? Below you will find: (a) the "sip show peer nnn" for an ATA extension and a Cisco extension (b) the SIP debug trace for (i) a successful call pickup from the ATA and (ii) an unsuccessful call pickup from the Cisco Any light anyone can shed on the perplexing problem would be most appreciated. I have a forehead-shaped dent in the wall that is growing larger. Linksys SPA2102 ATA "sip show peer 101": * Name : 101 Secret : <Set> MD5Secret : <Not set> Context : numberplan-custom-1 Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : 1 Pickupgroup : 1 Mailbox : 101 VM Extension : asterisk LastMsgsSent : 0/3 Call limit : 0 Dynamic : Yes Callerid : "Dxx Gxxx" <604-123-4444> MaxCallBR : 384 kbps Expire : 1861 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.0.205 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 101 SIP Options : replaces replace Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing: No Status : OK (5 ms) Useragent : Linksys/SPA2102-5.2.3 Reg. Contact : sip:[email protected]:5060 Cisco 7940G Phone "sip show peer 106": * Name : 106 Secret : <Set> MD5Secret : <Not set> Context : numberplan-custom-1 Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : 1 Pickupgroup : 1 Mailbox : 106 VM Extension : asterisk LastMsgsSent : 3/1 Call limit : 0 Dynamic : Yes Callerid : "Cxxx Nxxx" <6041234567> MaxCallBR : 384 kbps Expire : 247 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.0.211 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 106 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) Auto-Framing: No Status : OK (195 ms) Useragent : Cisco-CP7940G/8.0 Reg. Contact : sip:[email protected]:5060;transport=udp Successful *8 Call Pickup (SIP Trace) <--- SIP read from 192.168.0.205:5060 ---> INVITE sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c From: 101 <sip:[email protected]>;tag=22b459c8b65178bco0 To: <sip:*[email protected]> Remote-Party-ID: 101 <sip:[email protected]>;screen=yes;party=calling Call-ID: [email protected] CSeq: 101 INVITE Max-Forwards: 70 Contact: 101 <sip:[email protected]:5060> Expires: 240 User-Agent: Linksys/SPA2102-5.2.3 Content-Length: 444 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 922981 922981 IN IP4 192.168.0.205 s=- c=IN IP4 192.168.0.205 t=0 0 m=audio 16412 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (15 headers 20 lines) --- Sending to 192.168.0.205 : 5060 (no NAT) Using INVITE request as basis request - [email protected] Found peer '101' tg2*CLI> <--- Reliably Transmitting (no NAT) to 192.168.0.205:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c;received=192.168.0.205 From: 101 <sip:[email protected]>;tag=22b459c8b65178bco0 To: <sip:*[email protected]>;tag=as4bf7113a Call-ID: [email protected] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="637bf838" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE) tg2*CLI> <--- SIP read from 192.168.0.205:5060 ---> ACK sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c From: 101 <sip:[email protected]>;tag=22b459c8b65178bco0 To: <sip:*[email protected]>;tag=as4bf7113a Call-ID: [email protected] CSeq: 101 ACK Max-Forwards: 70 Contact: 101 <sip:[email protected]:5060> User-Agent: Linksys/SPA2102-5.2.3 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- tg2*CLI> <--- SIP read from 192.168.0.205:5060 ---> INVITE sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-4be14d40 From: 101 <sip:[email protected]>;tag=22b459c8b65178bco0 To: <sip:*[email protected]> Remote-Party-ID: 101 <sip:[email protected]>;screen=yes;party=calling Call-ID: [email protected] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="101",realm="asterisk",nonce="637bf838",uri="sip:*[email protected]",algorithm=MD5,response="8064fac79e90815539b2ee1aa33df011" Contact: 101 <sip:[email protected]:5060> Expires: 240 User-Agent: Linksys/SPA2102-5.2.3 Content-Length: 444 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 922981 922981 IN IP4 192.168.0.205 s=- c=IN IP4 192.168.0.205 t=0 0 m=audio 16412 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 =rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (16 headers 20 lines) --- Sending to 192.168.0.205 : 5060 (no NAT) Using INVITE request as basis request - [email protected] Found peer '101' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.205:16412 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G723 for ID 4 Found audio description format PCMA for ID 8 Found audio description format G729a for ID 18 Found unknown media description format G726-40 for ID 96 Found unknown media description format G726-24 for ID 97 Found unknown media description format G726-16 for ID 98 Found unknown media description format NSE for ID 100 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.205:16412 Looking for *8 in numberplan-custom-1 (domain 192.168.0.12) list_route: hop: <sip:[email protected]:5060> tg2*CLI> <--- Transmitting (no NAT) to 192.168.0.205:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-4be14d40;received=192.168.0.205 From: 101 <sip:[email protected]>;tag=22b459c8b65178bco0 To: <sip:*[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*[email protected]> Content-Length: 0 <------------> Audio is at 192.168.0.12 port 17634 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP tg2*CLI> <--- Reliably Transmitting (no NAT) to 192.168.0.205:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-4be14d40;received=192.168.0.205 From: 101 <sip:[email protected]>;tag=22b459c8b65178bco0 To: <sip:*[email protected]>;tag=as49b9fd26 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*[email protected]> Content-Type: application/sdp Content-Length: 262 v=0 o=root 1941 1941 IN IP4 192.168.0.12 s=session c=IN IP4 192.168.0.12 t=0 0 m=audio 17634 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> tg2*CLI> <--- SIP read from 192.168.0.205:5060 ---> ACK sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-58c05ba7 From: 101 <sip:[email protected]>;tag=22b459c8b65178bco0 To: <sip:*[email protected]>;tag=as49b9fd26 Call-ID: [email protected] CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="101",realm="asterisk",nonce="637bf838",uri="sip:*[email protected]",algorithm=MD5,response="8064fac79e90815539b2ee1aa33df011" Contact: 101 <sip:[email protected]:5060> User-Agent: Linksys/SPA2102-5.2.3 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.0.191:5060: CANCEL sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK7548dfb8;rport From: "New User" <sip:[email protected]>;tag=as29c3bf9e To: <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE) -- SIP/101-08196f30 answered SIP/nv2.xxxl01.a-081c91e0 Audio is at 192.168.0.12 port 12632 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP tg2*CLI> <--- Reliably Transmitting (NAT) to 64.34.49.82:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.34.49.82:5060;branch=z9hG4bK23f3f87a;received=64.34.49.82;rport=5060 From: "6041112222" <sip:[email protected]>;tag=as637ef124 To: <sip:[email protected]>;tag=as36bb1ac1 Call-ID: [email protected] CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 306 v=0 o=root 1941 1941 IN IP4 192.168.0.12 s=session c=IN IP4 192.168.0.12 t=0 0 m=audio 12632 RTP/AVP 3 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/nv2.tevl01.a-081c91e0 and SIP/101-08196f30Successful *8 Call Pickup (SIP Trace) x<--- SIP read from 192.168.0.211:50386 ---> INVITE sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK29b51f96 From: "Cxxx Nxxxx" <sip:[email protected]>;tag=000d6556eeb302bd081e4f74-20817e6e To: <sip:*[email protected]> Call-ID: [email protected] Max-Forwards: 70 Date: Wed, 08 Apr 2009 00:54:50 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7940G/8.0 Contact: <sip:[email protected]:5060;transport=udp> Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Supported: replaces,join,norefersub Content-Length: 275 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 25698 0 IN IP4 192.168.0.211 s=SIP Call t=0 0 m=audio 19514 RTP/AVP 0 8 18 101 c=IN IP4 192.168.0.211 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (17 headers 13 lines) --- Sending to 192.168.0.211 : 5060 (no NAT) Using INVITE request as basis request - [email protected] Found no matching peer or user for '192.168.0.211:50386' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.211:19514 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.211:19514 Looking for *8 in numberplan-custom-1 (domain 192.168.0.12) list_route: hop: <sip:[email protected]:5060;transport=udp> tg2*CLI> <--- Transmitting (no NAT) to 192.168.0.211:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK29b51f96;received=192.168.0.211 From: "Cxxx Nxxxx" <sip:[email protected]>;tag=000d6556eeb302bd081e4f74-20817e6e To: <sip:*[email protected]> Call-ID: [email protected] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*[email protected]> Content-Length: 0 <------------> [Apr 7 17:58:50] NOTICE[2097]: chan_sip.c:13977 handle_request_invite: Nothing to pick up for [email protected] <--- Reliably Transmitting (no NAT) to 192.168.0.211:5060 ---> SIP/2.0 503 Unavailable Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK29b51f96;received=192.168.0.211 From: "Cxxx Nxxxx" <sip:[email protected]>;tag=000d6556eeb302bd081e4f74-20817e6e To: <sip:*[email protected]>;tag=as52f770b6 Call-ID: [email protected] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*[email protected]> Content-Length: 0 <------------> tg2*CLI> <--- SIP read from 192.168.0.211:50387 ---> ACK sip:*[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK29b51f96 From: "Cxxx Nxxxx" <sip:[email protected]>;tag=000d6556eeb302bd081e4f74-20817e6e To: <sip:*[email protected]>;tag=as52f770b6 Call-ID: [email protected] Date: Wed, 08 Apr 2009 00:54:50 GMT CSeq: 101 ACK Content-Length: 0 <-------------> -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) |
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