nathans Posted January 9, 2012 Report Share Posted January 9, 2012 This one is trance as I'm pretty sure it was working and somehow we did something and it is not longer working. Basically we are trying to use the default star code for call forwarding *77 or call pick up and we get a recording for "service is not available" On the snomone trace below extension 9999 is trying to forward a call to extension 45. 7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024: INVITE sip:*7745@192.168.30.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone> Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1 X-Serialnumber: 000413412E6B P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom870/8.4.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:187.174.100.120>;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 528 v=0 o=root 100582088 100582088 IN IP4 75.149.181.121 s=call c=IN IP4 75.149.181.121 t=0 0 m=audio 53466 RTP/AVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XzhHQKdn2K8ILVlcwvTbc6XTO0iWl7pmUhQ4ijEM a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport=1024 From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 1 INVITE Content-Length: 0 [7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport=1024 From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 1 INVITE User-Agent: snom-PBX/2011-4.2.1.4025 WWW-Authenticate: Digest realm="192.168.30.2",nonce="059ba8b53b1b0982fd128db9b5ef8e58",domain="sip:*7745@192.168.30.2:5060;user=phone",algorithm=MD5 Content-Length: 0 [7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024: ACK sip:*7745@192.168.30.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1 Content-Length: 0 [7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024: INVITE sip:*7745@187.174.100.120;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9 To: <sip:*7745@187.174.100.120;user=phone> Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:9999@192.168.88.107:3072;line=4r9hwkd9>;reg-id=1 X-Serialnumber: 000413412E6B P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom870/8.4.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:187.174.100.120>;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="9999",realm="192.168.30.2",nonce="059ba8b53b1b0982fd128db9b5ef8e58",uri="sip:*7745@192.168.30.2:5060;user=phone",response="7d4d1b35a0321805252b0311df97c9e3",algorithm=MD5 Content-Type: application/sdp Content-Length: 528 v=0 o=root 100582088 100582088 IN IP4 192.168.88.107 s=call c=IN IP4 192.168.88.107 t=0 0 m=audio 53466 RTP/AVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XzhHQKdn2K8ILVlcwvTbc6XTO0iWl7pmUhQ4ijEM a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport=1024;received=75.149.181.121 From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9 To: <sip:*7745@187.174.100.120;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 2 INVITE Content-Length: 0 [7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport=1024;received=75.149.181.121 From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9 To: <sip:*7745@187.174.100.120;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 2 INVITE Contact: <sip:9999@192.168.30.2:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Type: application/sdp Content-Length: 396 v=0 o=- 405392631 405392631 IN IP4 192.168.30.2 s=- c=IN IP4 192.168.30.2 t=0 0 m=audio 53968 RTP/AVP 0 8 9 18 99 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2012/01/09 17:42:38: SIP Rx udp:75.149.181.121:1024: ACK sip:9999@192.168.30.2:5060 SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-lh9yrb0744rt;rport From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1 Content-Length: 0 [7] 2012/01/09 17:42:40: SIP Rx udp:75.149.181.121:1024: BYE sip:9999@192.168.30.2:5060 SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-78xiabcq4825;rport From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1 User-Agent: snom870/8.4.33 RTP-RxStat: Total_Rx_Pkts=106,Rx_Pkts=106,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=165,Tx_Pkts=121,Remote_Tx_Pkts=0 Content-Length: 0 [7] 2012/01/09 17:42:40: SIP Tx udp:75.149.181.121:1024: SIP/2.0 200 Ok Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-78xiabcq4825;rport=1024 From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9 To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193 Call-ID: 8633353c102b-7vv98fylg9ww CSeq: 3 BYE Contact: <sip:9999@192.168.30.2:5060> User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia support Posted January 12, 2012 Report Share Posted January 12, 2012 Can you check if the star code has been changed? We have just tested 4.2.1.4025 and was able to transfer the call. 1. 40 calls 41, 2. 41 puts ext 40 on hold 3. 41 press *7742 4. pbx disconnects 41 and transfers 42 and 40. Quote Link to comment Share on other sites More sharing options...
nathans Posted January 26, 2012 Author Report Share Posted January 26, 2012 We rebooted both systems and the *codes are working again. Very, very strange as nothing was changed. Just a reboot and now it work. Strange. Quote Link to comment Share on other sites More sharing options...
nathans Posted February 21, 2012 Author Report Share Posted February 21, 2012 And after 3 weeks the same problem again. The call redirect or pick ceased to work. No changes done to the system. Any ideas? It is very frustrating having to reboot the system every 3 weeks. Quote Link to comment Share on other sites More sharing options...
Vodia support Posted February 21, 2012 Report Share Posted February 21, 2012 it's rather rare for this to not work can you post a sip-trace of the call redirect? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 22, 2012 Report Share Posted February 22, 2012 Before you reboot, can you take a screen shot from the status screen? It looks to me more like there are some dangling call objects there that stop the PBX from making more calls. is it only the *77 code?! I dont see what is so special about the code, there must be also other things be affected. Quote Link to comment Share on other sites More sharing options...
nathans Posted February 22, 2012 Author Report Share Posted February 22, 2012 HI, Actually most of the codes requiring an extension like *83 and *77 are the ones not working anymore. I just found out also that the call forwarding also is not working any more. Is this the screen status you wanted? Let me know if you need more infer before rebooting the system. I will try to get a a log later on . Quote Link to comment Share on other sites More sharing options...
nathans Posted February 22, 2012 Author Report Share Posted February 22, 2012 screen shoot Quote Link to comment Share on other sites More sharing options...
nathans Posted February 22, 2012 Author Report Share Posted February 22, 2012 some log on the snomone of a failed *83 call pick attempt: [7] 2012/02/22 13:49:17: SIP Rx tls:192.168.30.77:4422: INVITE sip:*877121@pbx.bondojito.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay To: <sip:*877121@pbx.bondojito.com;user=phone> Call-ID: 17c0263cd8d7-yvp7g9nueny5 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1 X-Serialnumber: 000413457659 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 524 v=0 o=root 367130050 367130050 IN IP4 192.168.30.77 s=call c=IN IP4 192.168.30.77 t=0 0 m=audio 64274 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UKfZKYOjMYhdvmxL3Uk8T2mUEmznx5sPoPLZ0/HE a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/02/22 13:49:17: SIP Tx tls:192.168.30.77:4422: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport=4422 From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54 Call-ID: 17c0263cd8d7-yvp7g9nueny5 CSeq: 1 INVITE Content-Length: 0 [7] 2012/02/22 13:49:17: SIP Tx tls:192.168.30.77:4422: SIP/2.0 404 Not Found Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport=4422 From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54 Call-ID: 17c0263cd8d7-yvp7g9nueny5 CSeq: 1 INVITE Contact: <sip:7120@192.168.30.2:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [7] 2012/02/22 13:49:17: SIP Rx tls:192.168.30.77:4422: ACK sip:*877121@pbx.bondojito.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54 Call-ID: 17c0263cd8d7-yvp7g9nueny5 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [7] 2012/02/22 13:49:29: SIP Rx tls:192.168.30.77:4422: INVITE sip:*877121@pbx.bondojito.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi To: <sip:*877121@pbx.bondojito.com;user=phone> Call-ID: 25c0263c5251-7ubabgx00lt8 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1 X-Serialnumber: 000413457659 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 526 v=0 o=root 1685612402 1685612402 IN IP4 192.168.30.77 s=call c=IN IP4 192.168.30.77 t=0 0 m=audio 56568 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+gqsKVDkqLc0eaydfsXtEcnTYUf1dCTo+h/mTo34 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/02/22 13:49:29: SIP Tx tls:192.168.30.77:4422: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport=4422 From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c Call-ID: 25c0263c5251-7ubabgx00lt8 CSeq: 1 INVITE Content-Length: 0 [7] 2012/02/22 13:49:29: SIP Tx tls:192.168.30.77:4422: SIP/2.0 404 Not Found Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport=4422 From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c Call-ID: 25c0263c5251-7ubabgx00lt8 CSeq: 1 INVITE Contact: <sip:7120@192.168.30.2:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [7] 2012/02/22 13:49:29: SIP Rx tls:192.168.30.77:4422: ACK sip:*877121@pbx.bondojito.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c Call-ID: 25c0263c5251-7ubabgx00lt8 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1 Proxy-Require: buttons Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 22, 2012 Report Share Posted February 22, 2012 Okay, now we know that this is not specific to *77. The version that you are running is somewhat outdated; what could be a problem is that the number of G.729A codecs is exhausted. There was a bug in the older version that could cause this problem. Any chance to upgrade? 4.5 (Beta Corodindis) is available. If you backup your configuration and see if the trunks are working fine after the upgrade (try this after hours to minimize the disturbance), it would be worth giving it a try. If not, you can move back to the previous version. Quote Link to comment Share on other sites More sharing options...
nathans Posted February 23, 2012 Author Report Share Posted February 23, 2012 Are the G729 licenses recycled? I'm guessing the error happens if I have X number of simultaneous calls going on and I try to pick up one? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 24, 2012 Report Share Posted February 24, 2012 Are the G729 licenses recycled? I'm guessing the error happens if I have X number of simultaneous calls going on and I try to pick up one? Well all I remember that there was a bug some versions ago. IMHO it is worth trying to upgrade. Quote Link to comment Share on other sites More sharing options...
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