Cisco 2800 sip trunk

[leon1e]

Op een Cisco 2800 met software versie 12.4 met daarachter een Call Manager 6.5 probeer ik een sip trunk naar een provider werkend te krijgen. De calls naar buiten lopen al wel goed via de trunk, maar inkomend is het nog een probleem. Zelf heb ik het idee dat er een verkeerd nummer binnen komt (zie hieronder), maar misschien is het ook wel een config foutje graag jullie advies.

Sip log:

From: "06525*****" <sip:06525*****@89.***.172.**:5080>;tag=as4ffc27fb
To: <sip:6782vc01@89.***.172.**:5080>

Config router:

(onder de **** staat het nummer volledig uitgeschreven)

code:

voice translation-rule 1
 rule 1 /053760****/ /314/
!
voice translation-profile Inkomend
 translate calling 2
 translate called 1
!

dial-peer voice 10 voip
 translation-profile outgoing Outgoing
 destination-pattern 0T
 voice-class codec 1
 session protocol sipv2
 session target ipv4:****
 dtmf-relay rtp-nte
!
dial-peer voice 20 voip
 translation-profile incoming Inkomend
 voice-class codec 1
 session protocol sipv2
 session target ipv4:192.168.6.50
 incoming called-number 053760****
 dtmf-relay rtp-nte
!


sip-ua
 credentials username 6782vc01 password ***** realm sip.****.nl
 authentication username 6782vc01 password ****
 no remote-party-id
 retry invite 3
 retry register 3
 timers register 150
 registrar dns:sip.****.nl expires 3600
 sip-server dns:sip.****.nl
!

#show sip-ua register status
Line peer expires(sec) registered
============ ============= ============ ===========
6782vc01 -1 2343 yes

[Bart]

Ik had ooit het volgende draaien:

dial-peer voice 31 voip
destination-pattern ****
voice-class codec 1
session protocol sipv2
session target ipv4:*.*.*.*
dtmf-relay rtp-nte sip-notify
no vad
!

Die destination-pattern moet dan matchen met je To header op inkomende INVITE verwacht ik.

'incoming called-number' zou ik weghalen want dat matched denk ik niet door de wat "vreemde" To header.

[leon1e]

Incoming called-number heb ik nu weggehaald, en het destination pattern even op: destination-pattern 6782.* gezet, zonder resultaat. Ik ga morgen de provider maar eens bellen om de aparte header na te kijken.

[Bart]

Op zich is de header niet vreemd al verwacht je bij trunking eerder dat daar een telefoonnummer staat.

Zet eens wat debugging aan op die Cisco om te kijken wat er gebeurd als er een call binnenkomt.

En maak van dat destination-pattern gewoon eens 6782vc01

[ Voor 12% gewijzigd door Bart op 23-11-2011 20:36 ]

[leon1e]

Destination-pattern veranderen naar 6782vc01 werkt niet, de router verwacht een regular expression of een correct E.164 number.

001333: Nov 23 19:42:06.809: //10/0F6171718044/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x65007D78
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 057555****
Called Number : 6782vc01
Source IP Address (Sig ): 94.****
Destn SIP Req Addr:Port : 89.****
Destn SIP Resp Addr:Port : 89.***
Destination Name : 89.***

code:

001339: Nov 23 19:45:46.897: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 89.***.***.*4:5060
001340: Nov 23 19:45:46.897: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone gmt1 to SIP default timezone = GMT
001341: Nov 23 19:45:46.897: //-1/92934AFB8056/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
001342: Nov 23 19:45:46.897: //-1/92934AFB8056/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
001343: Nov 23 19:45:46.897: //-1/92934AFB8056/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
001344: Nov 23 19:45:46.897: //-1/92934AFB8056/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
001345: Nov 23 19:45:46.897: //-1/92934AFB8056/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
001346: Nov 23 19:45:46.897: //-1/92934AFB8056/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
001347: Nov 23 19:45:46.897: //-1/92934AFB8056/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
001348: Nov 23 19:45:46.897: //-1/92934AFB8056/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 3
001349: Nov 23 19:45:46.901: //-1/92934AFB8056/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
001350: Nov 23 19:45:46.901: //-1/92934AFB8056/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 64FE5438
001351: Nov 23 19:45:48.293: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 89.***.***.*4:5060
001352: Nov 23 19:45:48.297: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone gmt1 to SIP default timezone = GMT
001353: Nov 23 19:45:48.297: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetGtdBody: No valid GTD body found.
001354: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
001355: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x64FE76CC key=6505ec981498e2c62202372a1569b528@89.***.***.*5:50806782vc01
001356: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
001357: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: 6782vc01
001358: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: 057555****
001359: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name 057555****, number 057555****, Calling oct3 0x00, oct_3a 0x80, Called number 6782vc01
001360: Nov 23 19:45:48.301: //-1/9368E57F8057/SIP/Info/sipSPIGetCallConfig: Peer tag 10 matched for incoming call
001361: Nov 23 19:45:48.301: //-1/9368E57F8057/SIP/Info/sipSPIGetCallConfig: Using Voice Class Codec, tag = 1
001362: Nov 23 19:45:48.301: //-1/9368E57F8057/SIP/Info/sipSPICopyPeerDataToCCB:
From CLI: Modem NSE payload = 100, Passthrough = 0, Modem relay = 0, Gw-Xid = 1
SPRT latency 200, SPRT Retries = 12, Dict Size = 1024
 String Len = 32, Compress dir = 3
001363: Nov 23 19:45:48.301: //-1/9368E57F8057/SIP/Info/sipSPIContinueNewMsgInvite: Calling name 057555****, number 057555****, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number 6782vc01, oct3 0x00
001364: Nov 23 19:45:48.301: //-1/9368E57F8057/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
001365: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
001366: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1
001367: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:20
001368: Nov 23 19:45:48.301: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :20, codecbytes: 160
001369: Nov 23 19:45:48.301: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
001370: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
001371: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved.
001372: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPIDoDTMFRelayNegotiation: Payload type (101) is reserved for requested dtmf relay mode.
001373: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named event(NE) match in fmtp list of events.
001374: Nov 23 19:45:48.301: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0
001375: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
001376: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0
001377: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
        payload_type=8, codec_bytes=160, codec=g711alaw, dtmf_relay=rtp-nte
        stream_type=voice+dtmf (1), dest_ip_address=89.***.***.***, dest_port=10898
001378: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPI_ipip_report_media_to_peer:
 callId 12 peer 0 flags 0x201
001379: Nov 23 19:45:48.301: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CallID 12, sdp 0x659DEBD4 channels 0x64FE8F94
001380: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 8 mline 1
001381: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw
001382: Nov 23 19:45:48.305: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
001383: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160, coverted ptime=20 stream->mline_index=1, media_ndx=1
001384: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 6 ptype 8 time 20, bytes 160  as channel 0 mline 1 ss 1 89.***.***.***:10898
001385: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 0 mline 1
001386: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711ulaw
001387: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation NOT done, get ptime from sdp: ptime=20, media_ndx=1
001388: Nov 23 19:45:48.305: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :20, codecbytes: 160
001389: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 5 ptype 0 time 20, bytes 160  as channel 1 mline 1 ss 1 89.***.***.***:10898
001390: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 111 mline 1
001391: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 112 mline 1
001392: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 3 mline 1
001393: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec gsmfr
001394: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation NOT done, get ptime from sdp: ptime=20, media_ndx=1
001395: Nov 23 19:45:48.305: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: gsmfr ptime :20, codecbytes: 66
001396: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 10 ptype 3 time 20, bytes 66  as channel 2 mline 1 ss 1 89.***.***.***:10898
001397: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 101 mline 1
001398: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_report_media_to_peer:
Report initial call media
001399: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/copy_channels:
 callId 12 size 224 ptr 0x65B54188)
001400: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPI_ipip_report_media_to_peer:
CCSIP: Unable to report channel ind
001401: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPIHandleInviteMedia:
Negotiated Codec        : g711alaw, bytes :160
Preferred Codec         : g711alaw, bytes :160
Preferred  DTMF relay 1 : 6
Preferred  DTMF relay 2 : 0
Negotiated DTMF relay   : 6
Preferred and Negotiated NTE payloads: 101 101
Preferred and Negotiated NSE payloads: 100 0
Preferred and Negotiated Modem Relay: 0 0
Preferred and Negotiated Modem Relay GwXid: 1 0

001402: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPIDoQoSNegotiation: SDP body with media description
001403: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
001404: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=16770
001405: Nov 23 19:45:48.305: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
001406: Nov 23 19:45:48.305: //12/9368E57F8057/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 16770
001407: Nov 23 19:45:48.309: //12/9368E57F8057/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Negotiated method of dtmf relayand pyld: 6 101

001408: Nov 23 19:45:48.309: //12/9368E57F8057/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = 6505ec981498e2c62202372a1569b528@89.***.***.*5:5080
001409: Nov 23 19:45:48.309: //12/9368E57F8057/SIP/Info/sipSPIAddInviteDataToCcbCont: Headers from INVITE added to callInfo container
001410: Nov 23 19:45:48.309: ccDumpTdRequestDataSip:
001411: Nov 23 19:45:48.309: reqURI=sip:6782vc01@94.**.*.**:5060
001412: Nov 23 19:45:48.309: calling_urip=sip:057555****@89.***.***.*5:5080
001413: Nov 23 19:45:48.309: url_dump_header_line_avpair:
001414: Nov 23 19:45:48.309:   num_headers = 15
001415: Nov 23 19:45:48.309:     headers[0].linep = From: "057555****" <sip:057555****@89.***.***.*5:5080>;tag=as37a636b4, len = 69
001416: Nov 23 19:45:48.309:       data.attr.datap = From: "057555****" <sip:057555****@89.***.***.*5:5080>;tag=as37a636b4, len = 4
001417: Nov 23 19:45:48.309:       data.value.datap = "057555****" <sip:057555****@89.***.***.*5:5080>;tag=as37a636b4, len = 63
001418: Nov 23 19:45:48.309:     headers[1].linep = To: <sip:6782vc01@89.***.***.*5:5080>, len = 37
001419: Nov 23 19:45:48.309:       data.attr.datap = To: <sip:6782vc01@89.***.***.*5:5080>, len = 2
001420: Nov 23 19:45:48.309:       data.value.datap = <sip:6782vc01@89.***.***.*5:5080>, len = 33
001421: Nov 23 19:45:48.309:     headers[2].linep = Via: SIP/2.0/UDP 89.***.***.*4;branch=z9hG4bKc58d.6f102534.0,SIP/2.0/UDP 89.***.***.***:5070;branch=z9hG4bK6e9d747d, len = 115
001422: Nov 23 19:45:48.309:       data.attr.datap = Via: SIP/2.0/UDP 89.***.***.*4;branch=z9hG4bKc58d.6f102534.0,SIP/2.0/UDP 89.***.***.***:5070;branch=z9hG4bK6e9d747d, len = 3
001423: Nov 23 19:45:48.309:       data.value.datap = SIP/2.0/UDP 89.***.***.*4;branch=z9hG4bKc58d.6f102534.0,SIP/2.0/UDP 89.***.***.***:5070;branch=z9hG4bK6e9d747d, len = 110
001424: Nov 23 19:45:48.309:     headers[3].linep = Call-ID: 6505ec981498e2c62202372a1569b528@89.***.***.*5:5080, len = 60
001425: Nov 23 19:45:48.309:       data.attr.datap = Call-ID: 6505ec981498e2c62202372a1569b528@89.***.***.*5:5080, len = 7
001426: Nov 23 19:45:48.309:       data.value.datap = 6505ec981498e2c62202372a1569b528@89.***.***.*5:5080, len = 51
001427: Nov 23 19:45:48.309:     headers[4].linep = CSeq: 102 INVITE, len = 16
001428: Nov 23 19:45:48.309:       data.attr.datap = CSeq: 102 INVITE, len = 4
001429: Nov 23 19:45:48.309:       data.value.datap = 102 INVITE, len = 10
001430: Nov 23 19:45:48.309:     headers[5].linep = Contact: <sip:057555****@89.***.***.***:5070>, len = 45
001431: Nov 23 19:45:48.309:       data.attr.datap = Contact: <sip:057555****@89.***.***.***:5070>, len = 7
001432: Nov 23 19:45:48.309:       data.value.datap = <sip:057555****@89.***.***.***:5070>, len = 36
001433: Nov 23 19:45:48.309:     headers[6].linep = Content-Length: 377, len = 19
001434: Nov 23 19:45:48.313:       data.attr.datap = Content-Length: 377, len = 14
001435: Nov 23 19:45:48.313:       data.value.datap = 377, len = 3
001436: Nov 23 19:45:48.313:     headers[7].linep = Content-Type: application/sdp, len = 29
001437: Nov 23 19:45:48.313:       data.attr.datap = Content-Type: application/sdp, len = 12
001438: Nov 23 19:45:48.313:       data.value.datap = application/sdp, len = 15
001439: Nov 23 19:45:48.313:     headers[8].linep = Record-Route: <sip:89.***.***.*4;r2=on;lr=on;ftag=as37a636b4>,<sip:89.***.***.*5:5080;r2=on;lr=on;ftag=as37a636b4>, len = 114
001440: Nov 23 19:45:48.313:       data.attr.datap = Record-Route: <sip:89.***.***.*4;r2=on;lr=on;ftag=as37a636b4>,<sip:89.***.***.*5:5080;r2=on;lr=on;ftag=as37a636b4>, len = 12
001441: Nov 23 19:45:48.313:       data.value.datap = <sip:89.***.***.*4;r2=on;lr=on;ftag=as37a636b4>,<sip:89.***.***.*5:5080;r2=on;lr=on;ftag=as37a636b4>, len = 100
001442: Nov 23 19:45:48.313:     headers[9].linep = NULL, len = 0
001443: Nov 23 19:45:48.313:       data.attr.datap = , len = 0
001444: Nov 23 19:45:48.313:       data.value.datap = , len = 0
001445: Nov 23 19:45:48.313:     headers[10].linep = NULL, len = 0
001446: Nov 23 19:45:48.313:       data.attr.datap = , len = 0
001447: Nov 23 19:45:48.313:       data.value.datap = , len = 0
001448: Nov 23 19:45:48.313:     headers[11].linep = Max-Forwards: 69, len = 16
001449: Nov 23 19:45:48.313:       data.attr.datap = Max-Forwards: 69, len = 12
001450: Nov 23 19:45:48.313:       data.value.datap = 69, len = 2
001451: Nov 23 19:45:48.313:     headers[12].linep = User-Agent: Asterisk PBX 1.6.0.15, len = 33
001452: Nov 23 19:45:48.313:       data.attr.datap = User-Agent: Asterisk PBX 1.6.0.15, len = 10
001453: Nov 23 19:45:48.313:       data.value.datap = Asterisk PBX 1.6.0.15, len = 21
001454: Nov 23 19:45:48.313:     headers[13].linep = Date: Wed, 23 Nov 2011 19:45:48 GMT, len = 35
001455: Nov 23 19:45:48.313:       data.attr.datap = Date: Wed, 23 Nov 2011 19:45:48 GMT, len = 4
001456: Nov 23 19:45:48.313:       data.value.datap = Wed, 23 Nov 2011 19:45:48 GMT, len = 29
001457: Nov 23 19:45:48.313:     headers[14].linep = Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, len = 72
001458: Nov 23 19:45:48.313:       data.attr.datap = Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, len = 5
001459: Nov 23 19:45:48.313:       data.value.datap = INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, len = 65num_bodies = 0
001460: Nov 23 19:45:48.313: //12/9368E57F8057/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
001461: Nov 23 19:45:48.313: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed.
001462: Nov 23 19:45:48.313: //12/9368E57F8057/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
001463: Nov 23 19:45:48.313: //12/9368E57F8057/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
001464: Nov 23 19:45:48.313: //12/9368E57F8057/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x64FE76CC key=6505ec981498e2c62202372a1569b528@89.***.***.*5:508031BD94-14D5
001465: Nov 23 19:45:48.313: //12/9368E57F8057/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id C to table
001466: Nov 23 19:45:48.317: //12/9368E57F8057/SIP/Info/sipSPIProcessContactInfo: Previous Hop 89.***.***.*4:5060
001467: Nov 23 19:45:48.321: //12/9368E57F8057/SIP/Info/act_recdinvite_disconnect: Performing disconnect
001468: Nov 23 19:45:48.325: //12/9368E57F8057/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI
001469: Nov 23 19:45:48.325: //12/9368E57F8057/SIP/Info/sentErrResDisconnecting: Sent an 3456XX Error Response
001470: Nov 23 19:45:48.329: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 89.***.***.*4:5060
001471: Nov 23 19:45:48.329: //12/9368E57F8057/SIP/Info/sipSPIFindCcbUASReqTable: *****CCB found in UAS Request table. ccb=0x64FE76CC
001472: Nov 23 19:45:48.333: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone gmt1 to SIP default timezone = GMT
001473: Nov 23 19:45:48.333: //12/9368E57F8057/SIP/Info/sipSPIIcpifUpdate: CallState: 2 Playout: 0 DiscTime:325983 ConnTime 0
001474: Nov 23 19:45:48.333: //12/9368E57F8057/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x64FE76CC
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 057555****
Called Number            : 6782vc01
Source IP Address (Sig  ): 94.**.*.**
Destn SIP Req Addr:Port  : 89.***.***.*4:5060
Destn SIP Resp Addr:Port : 89.***.***.*4:5060
Destination Name         : 89.***.***.*4

001475: Nov 23 19:45:48.333: //12/9368E57F8057/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101
Source IP Address (Media): 94.**.*.**
Source IP Port    (Media): 16770
Destn  IP Address (Media): 89.***.***.***
Destn  IP Port    (Media): 10898
Orig Destn IP Address:Port (Media): 0.0.0.0:0

001476: Nov 23 19:45:48.333: //12/9368E57F8057/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 500

Hier gaat het dus fout:
001356: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
001357: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: 6782vc01
001358: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: 057555****
001359: Nov 23 19:45:48.297: //-1/9368E57F8057/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name 057555****, number 057555****, Calling oct3 0x00, oct_3a 0x80, Called number 6782vc01
001360: Nov 23 19:45:48.301: //-1/9368E57F8057/SIP/Info/sipSPIGetCallConfig: Peer tag 10 matched for incoming call

Peer 10 is de outgoing kant van de trunk:

code:

dial-peer voice 10 voip
 translation-profile outgoing Outgoing
 destination-pattern 0T
 voice-class codec 1
 session protocol sipv2
 session target ipv4:89.***.***.**
 dtmf-relay rtp-nte
!

[ Voor 96% gewijzigd door leon1e op 23-11-2011 21:03 ]

[Bart]

Het lijkt hem dus echt hier in te zitten:

Called number 6782vc01

Pakt de 'incoming called-number' parameter wel deze waarde? Of moet dit ook een regex/E164 nummer zijn? Als dat zo is, is het misschien makkelijk om te kijken of de trunk leverancier niet gewoon een telefoonnummer in de To kan aanleveren. Of anders misschien een voorbeeld heeft qua Cisco config.

[leon1e]

Bart schreef op woensdag 23 november 2011 @ 21:57:
Het lijkt hem dus echt hier in te zitten:

Called number 6782vc01

Pakt de 'incoming called-number' parameter wel deze waarde? Of moet dit ook een regex/E164 nummer zijn? Als dat zo is, is het misschien makkelijk om te kijken of de trunk leverancier niet gewoon een telefoonnummer in de To kan aanleveren. Of anders misschien een voorbeeld heeft qua Cisco config.

Ook niet, Incorrect format for E.164 Number. De leverancier komt er vandaag op terug, ik ben benieuwd. Op de andere trunk (zonder authenticatie, en andere leverancier) is het To field wel het telefoon nummer zie ik net.

[DJSmiley]

leon1e schreef op donderdag 24 november 2011 @ 09:04:
[...]


Ook niet, Incorrect format for E.164 Number. De leverancier komt er vandaag op terug, ik ben benieuwd. Op de andere trunk (zonder authenticatie, en andere leverancier) is het To field wel het telefoon nummer zie ik net.

Dat is een instelling op het voiceconnect kanaal. De betreffende leverancier kan dit instellen via een website bij de betreffende voip partij in Almere (ik weet niet of je direct daar inkoopt of t via een reseller doet)

Onder eigenschappen van voip kanaal dan geavanceerd daar kunnen ze opgeven hoe het nr moet aangeboden worden enz (Als je zelf ook een login hebt kun je volgens mij dat ook zelf wijzigen)



Uit mn hoofd moet je didToHeader op aan zetten (zie screenshot), zodat een oproep binnenkomt als <afzendernr>@sipserverip ipv als <voicekanaalnaam>@sipserverip

Kan ook zijn dat je even moet testen met didLenght, maar ik gok dat je 9 cijferig moet hebben.

Ps: een wijziging doorvoeren duurt een minuut of 5 voordat ie actief is

[ Voor 3% gewijzigd door DJSmiley op 25-11-2011 08:04 ]