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REMOVED DYNAMIC REGISTRAR FAILED


gifti
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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

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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.

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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

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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.

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