gifti Posted March 18, 2013 Report Share Posted March 18, 2013 Hello, I've got the following configuration: 3 x ISDN PMP <> Patton 4638 <> snomONE mini 5.0.5 Up to 5% of incoming calls won't be connectet to the incoming SIP-Interface to snomONE (IF_SIP_730730_PHONE). I get an error message in the patton syslog: <195>1 2013-03-18T14:30:53+01:00 192.168.0.220 SIP - - - SIP: [EP IF_SIP_730730_PHONE-00ad5190 SES 0x106c4e8] REMOVED DYNAMIC REGISTRAR FAILED After that, the call uses the 2nd destination (IF_ISDN_00_DEFAULT). It is an Default-ISDN-Phon which works even when the power fail. Config of the incoming HG on the Patton Gateway: service hunt-group HG_SIP_ISDN_730730_IN_PHONE timeout 2 allows-push-back drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable drop-cause unallocated-number unavailable drop transparent route call 1 dest-interface IF_SIP_730730_PHONE route call 2 dest-interface IF_ISDN_00_DEFAULT Why is the first destination-interface skipped sporadically ? Trunk Config snomONE: # Trunk 10 in domain pbx.ggizef.lokal Name: BRI_2_3_4_Bidirectional Type: gateway To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: RegRegistrar: RegKeep: RegUser: Icid: Require: OutboundProxy: 192.168.0.220:5060 Ani: DialExtension: !73073([0-9]{1,10}$)!!t!0 Trusted: false AcceptRedirect: false RfcRtp: true Analog: true RtpBegin: RtpEnd: Prack: false SendEmail: UseUuid: false Ring180: true Failover: never HeaderRequestUri: {request-uri} HeaderFrom: {from} HeaderTo: {to} HeaderPai: {trunk} HeaderPpi: HeaderRpi: HeaderPrivacy: HeaderRpiCharging: BlockCidPrefix: Glob: RequestTimeout: Codecs: CodecLock: true DtmfMode: Expires: 3600 Fraction: 128 Minimum: 10 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: UseEpid: false CidUpdate: Ignore18xSDP: true UserHdr: Diversion: {rfc} CoBusy: 500 Line Unavailable Colines: DialogPermission: And the SIP-Interface on the Patton Gateway: interface sip IF_SIP_730730_PHONE bind context sip-gateway GW_SIP_730730_PHONE route call dest-table RT_TO_ISDN_CLIP remote 192.168.0.200 early-connect no call-transfer pull-in call-reroute accept call-reroute emit privacy address-translation outgoing-call diversion-header host-part call regards gift Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 18, 2013 Report Share Posted March 18, 2013 The "DialExtension" looks suspicious to me, there should be something like \1 between the two !! there. Also, I would put sip: in from of the outbound proxy, so that it is a SIP URI. The message "REMOVED DYNAMIC REGISTRAR FAILED" is somewhat cryptical. If this is a gateway trunk, the PBX should not be a registrar at all. Does the Patton try to register? Maybe there is a time when that fails and then it redirects the calls. Quote Link to comment Share on other sites More sharing options...
gifti Posted March 19, 2013 Author Report Share Posted March 19, 2013 The \1 part of the DialExtension pattern was escaped after copy & paste . It really looks like this: !73073([0-9]{1,10}$)!\1!t!0 Quote Link to comment Share on other sites More sharing options...
Vodia support Posted March 19, 2013 Report Share Posted March 19, 2013 I believe we use the trunk gateway and not registration with Patton similar to Sangoma. http://www.patton.com/voip/appnotes/How_to_use_SmartNodes_with_SNOMOne.PDF Quote Link to comment Share on other sites More sharing options...
gifti Posted March 19, 2013 Author Report Share Posted March 19, 2013 It is a "no registration" PSTN Gateway configuration for Patton!? I've figured out, that the issue happens in both directions and it's not only a problem with the trunk configuration. I also sometimes get the message "Unsupported Media Type" when i try to dial my mailbox or an internal/external number. When I dial my own mailbox a few times, I sporadic get an "Unsupported Media Type" displayed on the SNOM370 like this: <133>1 2013-03-19T12:43:41+01:00 snomonemini SIP - - - SIP Rx tls:192.168.0.202:1057: INVITE sip:21@pbx.ggizef.lokal;user=phone SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.0.202:1057;branch=z9hG4bK-mw2zkbllida2;rport#015#012From: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=aum8svrlcy#015#012To: <sip:21@pbx.ggizef.lokal;user=phone>#015#012Call-ID: 51485d7c70dc-m9xajgf82xni#015#012CSeq: 1 INVITE#015#012Max-Forwards: 70#015#012Contact: <sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy>;reg-id=1#015#012X-Serialnumber: 0004133A8410#015#012P-Key-Flags: resolution="31x13", keys="4"#015#012User-Agent: snom370/8.7.3.19#015#012Accept: application/sdp#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE#015#012Allow-Events: talk, hold, refer, call-info#015#012Supported: timer, 100rel, replaces, from-change#015#012Session-Expires: 3600;refresher=uas#015#012Min-SE: 90#015#012Proxy-Require: buttons#015#012Content-Type: application/sdp#015#012Content-Length: 426#015#012#015#012v=0#015#012o=root 579784838 579784838 IN IP4 192.168.0.202#015#012s=call#015#012c=IN IP4 192.168.0.202#015#012t=0 0#015#012m=audio 58122 RTP/AVP 0 8 18 101#015#012a=crypto: <135>1 2013-03-19T12:43:41+01:00 snomonemini Packet - - - Packet authenticated by transport layer <135>1 2013-03-19T12:43:41+01:00 snomonemini Using - - - Using outbound proxy sip:192.168.0.202:1057;transport=tls because of flow-label <135>1 2013-03-19T12:43:41+01:00 snomonemini Last - - - Last message repeated 3 times <133>1 2013-03-19T12:43:41+01:00 snomonemini SIP - - - SIP Tx tls:192.168.0.202:1057: SIP/2.0 100 Trying#015#012Via: SIP/2.0/TLS 192.168.0.202:1057;branch=z9hG4bK-mw2zkbllida2;rport=1057#015#012From: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=aum8svrlcy#015#012To: <sip:21@pbx.ggizef.lokal;user=phone>;tag=73e7bda774#015#012Call-ID: 51485d7c70dc-m9xajgf82xni#015#012CSeq: 1 INVITE#015#012Content-Length: 0#015#012#015 <135>1 2013-03-19T12:43:41+01:00 snomonemini Incoming - - - Incoming call: Request URI sip:21@pbx.ggizef.lokal;user=phone, To is <sip:21@pbx.ggizef.lokal;user=phone> <135>1 2013-03-19T12:43:41+01:00 snomonemini Set - - - Set the To domain based on From user 21@pbx.ggizef.lokal <134>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: set_codecs for 51485d7c70dc-m9xajgf82xni codecs "", codec_preference count 4 <135>1 2013-03-19T12:43:41+01:00 snomonemini Play - - - Play audio_de/mb_main_menu.wav audio_de/mb_main_menu1.wav audio_de/mb_main_menu2.wav audio_de/mb_main_menu3.wav audio_de/mb_main_menu4.wav audio_de/mb_enter_choice2.wav audio_de/bi_press_5.wav audio_de/mb_main_menu9.wav space50, caching false <135>1 2013-03-19T12:43:41+01:00 snomonemini call - - - call port 201: state code from 0 to 200 <135>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: Other side does not support codec PCMU/8000 <135>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: Other side does not support codec PCMA/8000 <135>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: Other side does not support codec G729/8000 <135>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: Other side does not support codec telephone-event/8000 <131>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: codec_preference size 4, available codecs list is empty <133>1 2013-03-19T12:43:41+01:00 snomonemini send_connected - - - send_connected: available codec list is empty for 51485d7c70dc-m9xajgf82xni <133>1 2013-03-19T12:43:41+01:00 snomonemini SIP - - - SIP Tx tls:192.168.0.202:1057: SIP/2.0 415 Unsupported Media Type#015#012Via: SIP/2.0/TLS 192.168.0.202:1057;branch=z9hG4bK-mw2zkbllida2;rport=1057#015#012From: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=aum8svrlcy#015#012To: <sip:21@pbx.ggizef.lokal;user=phone>;tag=73e7bda774#015#012Call-ID: 51485d7c70dc-m9xajgf82xni#015#012CSeq: 1 INVITE#015#012Contact: <sip:21@192.168.0.200:5061;transport=tls>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.5#015#012Content-Length: 0#015#012#015 <133>1 2013-03-19T12:43:41+01:00 snomonemini The - - - The call port 201 - 30 seconds callback set for force cleanup The same issue happens spordic with incoming calls on the trunk BRI_2_3_4_Bidirectional. <133>1 2013-03-19T11:35:41+01:00 snomonemini SIP - - - SIP Rx udp:192.168.0.220:5060: INVITE sip:730730@192.168.0.200 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5f3fe8836cb72b10c#015#012Max-Forwards: 70#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=91f426b026#015#012To: <sip:730730@192.168.0.200>#015#012Call-ID: 6b7f622466197438#015#012CSeq: 21336 INVITE#015#012Contact: <sip:00493617315293@192.168.0.220:5060>#015#012Supported: replaces#015#012User-Agent: Patton SN4638 5BIS 00A0BA076E36 R6.2 2012-07-13 H323 SIP BRI M5T SIP Stack/4.0.30.30#015#012Content-Type: application/sdp#015#012Content-Length: 269#015#012#015#012v=0#015#012o=MxSIP 0 163 IN IP4 192.168.0.220#015#012s=SIP Call#015#012c=IN IP4 192.168.0.220#015#012t=0 0#015#012m=audio 5038 RTP/AVP 0 8 18 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:18 annexb=no#015#012a=fmtp:101 0-16#015#012a=sendrecv#015 <133>1 2013-03-19T11:35:41+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5f3fe8836cb72b10c#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=91f426b026#015#012To: <sip:730730@192.168.0.200>;tag=927ca7a609#015#012Call-ID: 6b7f622466197438#015#012CSeq: 21336 INVITE#015#012Content-Length: 0#015#012#015 <134>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: set_codecs for 6b7f622466197438 codecs "", codec_preference count 4 <135>1 2013-03-19T11:35:41+01:00 snomonemini Play - - - Play audio_moh/noise.wav, caching false <135>1 2013-03-19T11:35:41+01:00 snomonemini call - - - call port 155: state code from 0 to 180 <135>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: Other side does not support codec PCMU/8000 <135>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: Other side does not support codec PCMA/8000 <135>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: Other side does not support codec G729/8000 <135>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: Other side does not support codec telephone-event/8000 <131>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: codec_preference size 4, available codecs list is empty <133>1 2013-03-19T11:35:41+01:00 snomonemini Available - - - Available codec list is empty for 6b7f622466197438 <133>1 2013-03-19T11:35:41+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 415 Unsupported Media Type#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5f3fe8836cb72b10c#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=91f426b026#015#012To: <sip:730730@192.168.0.200>;tag=927ca7a609#015#012Call-ID: 6b7f622466197438#015#012CSeq: 21336 INVITE#015#012Contact: <sip:730730@192.168.0.200:5060;transport=udp>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.5#015#012Content-Length: 0#015#012#015 Any ideas? I've changed the ProxyAddress into sip:patton.ggizef.lokal:5060 but the problem still persists. Here is one of the 95% bug-free calls. <133>1 2013-03-19T13:56:46+01:00 snomonemini SIP - - - SIP Rx udp:192.168.0.220:5060: INVITE sip:7307321@192.168.0.200 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5e5b883ef5b2aa7e6#015#012Max-Forwards: 70#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=d03e8bd9ed#015#012To: <sip:7307321@192.168.0.200>#015#012Call-ID: e8e5ad9640944b34#015#012CSeq: 9171 INVITE#015#012Contact: <sip:00493617315293@192.168.0.220:5060>#015#012Supported: replaces#015#012User-Agent: Patton SN4638 5BIS 00A0BA076E36 R6.2 2012-07-13 H323 SIP BRI M5T SIP Stack/4.0.30.30#015#012Content-Type: application/sdp#015#012Content-Length: 269#015#012#015#012v=0#015#012o=MxSIP 0 193 IN IP4 192.168.0.220#015#012s=SIP Call#015#012c=IN IP4 192.168.0.220#015#012t=0 0#015#012m=audio 5072 RTP/AVP 0 8 18 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:18 annexb=no#015#012a=fmtp:101 0-16#015#012a=sendrecv#015 <135>1 2013-03-19T13:56:46+01:00 snomonemini Resolve - - - Resolve 792009: aaaa udp 192.168.0.220 5060 <135>1 2013-03-19T13:56:46+01:00 snomonemini Resolve - - - Resolve 792009: a udp 192.168.0.220 5060 <135>1 2013-03-19T13:56:46+01:00 snomonemini Resolve - - - Resolve 792009: udp 192.168.0.220 5060 <133>1 2013-03-19T13:56:46+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5e5b883ef5b2aa7e6#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=d03e8bd9ed#015#012To: <sip:7307321@192.168.0.200>;tag=32eb02a52a#015#012Call-ID: e8e5ad9640944b34#015#012CSeq: 9171 INVITE#015#012Content-Length: 0#015#012#015 <133>1 2013-03-19T13:56:46+01:00 snomonemini Call-leg - - - Call-leg 223: Sending RTP for e8e5ad9640944b34 to 192.168.0.220:5072, codec not set yet <135>1 2013-03-19T13:56:46+01:00 snomonemini Incoming - - - Incoming call: Request URI sip:7307321@192.168.0.200, To is <sip:7307321@192.168.0.200> <135>1 2013-03-19T13:56:46+01:00 snomonemini Set - - - Set the To domain based on To user 21@pbx.ggizef.lokal <134>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: set_codecs for e8e5ad9640944b34 codecs "0 8 18", codec_preference count 4 <134>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: set_codecs for 875a3b0b@pbx codecs "", codec_preference count 4 <135>1 2013-03-19T13:56:46+01:00 snomonemini Using - - - Using outbound proxy sip:192.168.0.202:1057;transport=tls because of flow-label <135>1 2013-03-19T13:56:46+01:00 snomonemini call - - - call port 224: state code from 0 to 100 <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: update_codecs for 875a3b0b@pbx: adding codec PCMU/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: update_codecs for 875a3b0b@pbx: adding codec PCMA/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: update_codecs for 875a3b0b@pbx: adding codec G729/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: update_codecs for 875a3b0b@pbx: codec_preference size 4, available codecs size 4 <133>1 2013-03-19T13:56:46+01:00 snomonemini SIP - - - SIP Tx tls:192.168.0.202:1057: INVITE sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.0.200:5061;branch=z9hG4bK-bae6ffc35cbb42fcd5ff49ca0ca62d4e;rport#015#012From: <sip:7315293@pbx.ggizef.lokal:5060;user=phone>;tag=232636715#015#012To: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>#015#012Call-ID: 875a3b0b@pbx#015#012CSeq: 2695 INVITE#015#012Max-Forwards: 70#015#012Contact: <sip:21@192.168.0.200:5061;transport=tls>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.5#015#012Alert-Info: <http://127.0.0.1/Bellcore-dr3>#015#012Content-Type: application/sdp#015#012Content-Length: 406#015#012#015#012v=0#015#012o=- 861259913 861259913 IN IP4 192.168.0.200#015#012s=-#015#012c=IN IP4 192.168.0.200#015#012t=0 0#015#012m=audio 59610 RTP/AVP 0 8 18 101#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:RokvG5jvSh3iKEgiYMZ+WA71XRjICZl7NCAeGHnD#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18 annexb=no#015 <135>1 2013-03-19T13:56:46+01:00 snomonemini Play - - - Play audio_moh/noise.wav, caching true <135>1 2013-03-19T13:56:46+01:00 snomonemini call - - - call port 223: state code from 0 to 100 <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: update_codecs for e8e5ad9640944b34: adding codec PCMU/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: update_codecs for e8e5ad9640944b34: adding codec PCMA/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: update_codecs for e8e5ad9640944b34: adding codec G729/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: update_codecs for e8e5ad9640944b34: codec_preference size 4, available codecs size 4 <133>1 2013-03-19T13:56:46+01:00 snomonemini SIP - - - SIP Rx tls:192.168.0.202:1057: SIP/2.0 100 Trying#015#012Via: SIP/2.0/TLS 192.168.0.200:5061;branch=z9hG4bK-bae6ffc35cbb42fcd5ff49ca0ca62d4e;rport=5061#015#012From: <sip:7315293@pbx.ggizef.lokal:5060;user=phone>;tag=232636715#015#012To: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=oirh8epwhw#015#012Call-ID: 875a3b0b@pbx#015#012CSeq: 2695 INVITE#015#012Contact: <sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy>;reg-id=1#015#012Content-Length: 0#015#012#015 <135>1 2013-03-19T13:56:46+01:00 snomonemini Message - - - Message repetition, packet dropped <133>1 2013-03-19T13:56:47+01:00 snomonemini SIP - - - SIP Rx tls:192.168.0.202:1057: SIP/2.0 180 Ringing#015#012Via: SIP/2.0/TLS 192.168.0.200:5061;branch=z9hG4bK-bae6ffc35cbb42fcd5ff49ca0ca62d4e;rport=5061#015#012From: <sip:7315293@pbx.ggizef.lokal:5060;user=phone>;tag=232636715#015#012To: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=oirh8epwhw#015#012Call-ID: 875a3b0b@pbx#015#012CSeq: 2695 INVITE#015#012Contact: <sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy>;reg-id=1#015#012Require: 100rel#015#012RSeq: 1#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE#015#012Allow-Events: talk, hold, refer, call-info#015#012Content-Length: 0#015#012#015 <133>1 2013-03-19T13:56:47+01:00 snomonemini SIP - - - SIP Tx tls:192.168.0.202:1057: PRACK sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.0.200:5061;branch=z9hG4bK-68fd95afa82a61df3ae3c6dd654816cc;rport#015#012From: <sip:7315293@pbx.ggizef.lokal:5060;user=phone>;tag=232636715#015#012To: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=oirh8epwhw#015#012Call-ID: 875a3b0b@pbx#015#012CSeq: 2696 PRACK#015#012Max-Forwards: 70#015#012Contact: <sip:21@192.168.0.200:5061;transport=tls>#015#012RAck: 1 2695 INVITE#015#012Content-Length: 0#015#012#015 <135>1 2013-03-19T13:56:47+01:00 snomonemini Play - - - Play audio_de/ringback.wav, caching true <135>1 2013-03-19T13:56:47+01:00 snomonemini call - - - call port 223: state code from 100 to 180 <135>1 2013-03-19T13:56:47+01:00 snomonemini Resolve - - - Resolve 792012: aaaa udp 192.168.0.220 5060 <135>1 2013-03-19T13:56:47+01:00 snomonemini Resolve - - - Resolve 792012: a udp 192.168.0.220 5060 <135>1 2013-03-19T13:56:47+01:00 snomonemini Resolve - - - Resolve 792012: udp 192.168.0.220 5060 <133>1 2013-03-19T13:56:47+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 180 Ringing#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5e5b883ef5b2aa7e6#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=d03e8bd9ed#015#012To: <sip:7307321@192.168.0.200>;tag=32eb02a52a#015#012Call-ID: e8e5ad9640944b34#015#012CSeq: 9171 INVITE#015#012Contact: <sip:7307321@192.168.0.200:5060;transport=udp>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.5#015#012Content-Length: 0#015#012#015 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 19, 2013 Report Share Posted March 19, 2013 Hmm... The PSTN gateway is not even involved in the 51485d7c70dc-m9xajgf82xni call. I guess if you make 100 calls to the mailbox, 5 will fail? I suspect the problem is related to the codec names which we have changed from pcmu to PCMU; we'll put some extra normalization rules into 5.0.7. Quote Link to comment Share on other sites More sharing options...
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