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I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites.<BR>
<BR>
Got SIP response 500 "Internal Server Error" back from 10.1.3.28<BR>
SIP/alma-1b77 is circuit-busy<BR>
Everyone is busy/congested at this time (1:0/1/0)<BR>
<BR>
The strange thing is that one of the access servers is working fine with the exact same configs in every way. I have moved both routers to the same IOS version which is:<BR>
<BR>
IOS (tm) 5400 Software (C5400-JS-M), Version 12.2(2)XB15<BR>
<BR>
I have included a copy of the dial-peers that are specified on the non-functional access server, and I have double checked the configs against the circuitassignments and they are correct.<BR>
<BR>
!<BR>
dial-peer voice 63201 pots<BR>
destination-pattern 632....<BR>
no digit-strip<BR>
port 1/0:0<BR>
!<BR>
dial-peer voice 63202 pots<BR>
destination-pattern 632....<BR>
no digit-strip<BR>
port 1/2:0<BR>
!<BR>
dial-peer voice 63203 pots<BR>
destination-pattern 632....<BR>
no digit-strip<BR>
port 1/3:0<BR>
!<BR>
dial-peer voice 63204 pots<BR>
destination-pattern 632....<BR>
no digit-strip<BR>
port 1/4:0<BR>
!<BR>
dial-peer voice 63401 pots<BR>
destination-pattern 634....<BR>
no digit-strip<BR>
port 1/5:0<BR>
!<BR>
dial-peer voice 63402 pots<BR>
destination-pattern 634....<BR>
no digit-strip<BR>
port 1/6:0<BR>
!<BR>
dial-peer voice 99701 pots<BR>
destination-pattern 997....<BR>
no digit-strip<BR>
port 1/0:0<BR>
!<BR>
dial-peer voice 99702 pots<BR>
destination-pattern 997....<BR>
no digit-strip<BR>
port 1/2:0<BR>
!<BR>
dial-peer voice 99703 pots<BR>
destination-pattern 997....<BR>
no digit-strip<BR>
port 1/3:0<BR>
!<BR>
dial-peer voice 99704 pots<BR>
destination-pattern 997....<BR>
no digit-strip<BR>
port 1/4:0<BR>
!<BR>
dial-peer voice 43001 pots<BR>
destination-pattern 430....<BR>
no digit-strip<BR>
port 1/0:0<BR>
!<BR>
dial-peer voice 43002 pots<BR>
destination-pattern 430....<BR>
no digit-strip<BR>
port 1/2:0<BR>
!<BR>
dial-peer voice 43003 pots<BR>
destination-pattern 430....<BR>
no digit-strip<BR>
port 1/3:0<BR>
!<BR>
dial-peer voice 43004 pots<BR>
destination-pattern 430....<BR>
no digit-strip<BR>
port 1/4:0<BR>
!<BR>
dial-peer voice 67001 pots<BR>
destination-pattern 670....<BR>
no digit-strip<BR>
port 1/0:0<BR>
!<BR>
dial-peer voice 67002 pots<BR>
destination-pattern 670....<BR>
no digit-strip<BR>
port 1/2:0<BR>
!<BR>
dial-peer voice 67003 pots<BR>
destination-pattern 670....<BR>
no digit-strip<BR>
port 1/3:0<BR>
!<BR>
dial-peer voice 67004 pots<BR>
destination-pattern 670....<BR>
no digit-strip<BR>
port 1/4:0<BR>
!<BR>
sip-ua<BR>
max-forwards 15<BR>
retry invite 10<BR>
timers trying 1000<BR>
timers expires 300000<BR>
sip-server ipv4:XXX.XXX.XXX.XXX:5060<BR>
no transport tcp<BR>
!<BR>
<BR>
The following is the debugs I collected from the access server with the problem:<BR>
<BR>
6936: 006932: Dec 29 01:48:05.075: Received:<BR>
6937: INVITE sip:6329900@10.1.3.28 SIP/2.0<BR>
6938: Via: SIP/2.0/UDP 65.67.76.41:5060;branch=z9hG4bK72629db3<BR>
6939: From: "5462000" <sip:<A HREF="mailto:5462000@65.67.76.41"><U>5462000@65.67.76.41</U></A>>;tag=as3e9b26ba<BR>
6940: To: <sip:<A HREF="mailto:6329900@10.1.3.28"><U>6329900@10.1.3.28</U></A>><BR>
6941: Contact: <sip:<A HREF="mailto:5462000@65.67.76.41"><U>5462000@65.67.76.41</U></A>><BR>
6942: Call-ID: <A HREF="mailto:023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41"><U>023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41</U></A><BR>
6943: CSeq: 102 INVITE<BR>
6944: User-Agent: Asterisk PBX<BR>
6945: Date: Wed, 29 Dec 2004 01:47:54 GMT<BR>
6946: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>
6947: Content-Type: application/sdp<BR>
6948: Content-Length: 179<BR>
6949:<BR>
6950: v=0<BR>
6951: o=root 9671 9671 IN IP4 65.67.76.41<BR>
6952: s=session<BR>
6953: c=IN IP4 65.67.76.41<BR>
6954: t=0 0<BR>
6955: m=audio 11980 RTP/AVP 0 3<BR>
6956: a=rtpmap:0 PCMU/8000<BR>
6957: a=rtpmap:3 GSM/8000<BR>
6958: a=silenceSupp:off - - - -<BR>
6959:<BR>
6960: 006933: Dec 29 01:48:05.075: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 65.67.76.41:5060<BR>
6961: 006934: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: sipSPISipIncomingCall<BR>
6962: 006935: Dec 29 01:48:05.075: 0x63CEA7B8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)<BR>
6963: 006936: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: act_idle_new_message<BR>
6964: 006937: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: Converting TimeZone CST to SIP default timezone = GMT<BR>
6965: 006938: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: sact_idle_new_message_invite<BR>
6966: 006939: Dec 29 01:48:05.075: CCSIP-SPI-CONTROL: sip_stats_method<BR>
6967: 006940: Dec 29 01:48:05.075: sipSPIGetSdpBody : Parse incoming session description<BR>
6968: 006941: Dec 29 01:48:05.079: Info: Media ip address/domain name in c line: 65.67.76.41<BR>
6969:<BR>
6970: 006942: Dec 29 01:48:05.079: sact_idle_new_message_invite: non dial peer leg - using RTP Supported Codecs<BR>
6971:<BR>
6972: 006943: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 18<BR>
6973:<BR>
6974: 006944: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 0<BR>
6975:<BR>
6976: 006945: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 8<BR>
6977:<BR>
6978: 006946: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 4<BR>
6979:<BR>
6980: 006947: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 2<BR>
6981:<BR>
6982: 006948: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 15<BR>
6983:<BR>
6984: 006949: Dec 29 01:48:05.079: sact_idle_new_message_invite: RTP Preferred Codecs supported by GW 3<BR>
6985:<BR>
6986: 006950: Dec 29 01:48:05.079: sipSPIDoFaxMediaNegotiation()<BR>
6987: 006951: Dec 29 01:48:05.079: sipSPIDoMediaNegotiation: Codec Negotiation Successful<BR>
6988:<BR>
6989: 006952: Dec 29 01:48:05.079: sipSPIDoMediaNegotiation: DTMF Relay mode : Inband Voice<BR>
6990:<BR>
6991: 006953: Dec 29 01:48:05.079: sipSPIUpdCcbWithSdpInfo: SDP Media Information:<BR>
6992: Negotiated Codec : g711ulaw , bytes :160<BR>
6993: Early Media : 0<BR>
6994: Delayed Media : 0<BR>
6995: Bridge Done : 0<BR>
6996: New Media : 0<BR>
6997: DSP DNLD Reqd : 0<BR>
6998: Media Dest addr/Port : 65.67.76.41:11980<BR>
6999:<BR>
7000: 006954: Dec 29 01:48:05.079: sipSPIHandleInviteMedia:<BR>
7001: Negotiated Codec : g711ulaw, bytes :160<BR>
7002: Preferred Codec : g729r8, bytes :20<BR>
7003: Preferred DTMF relay : 0<BR>
7004: Negotiated DTMF relay : 0<BR>
7005: Preferred and Negotiated NTE payloads: 101 0<BR>
7006:<BR>
7007: 006955: Dec 29 01:48:05.079: sipSPIDoQoSNegotiation - SDP body with media description<BR>
7008: 006956: Dec 29 01:48:05.079: sipSPIDoQoSNegotiation: Best effort call<BR>
7009: 006957: Dec 29 01:48:05.079: sipSPIAddBillingInfoToCcb: sipCallId for billing records = <A HREF="mailto:023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41"><U>023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41</U></A><BR>
7010: 006958: Dec 29 01:48:05.079: ****Adding to UAS Request table<BR>
7011:<BR>
7012: 006959: Dec 29 01:48:05.079: adding call id B to table<BR>
7013:<BR>
7014: 006960: Dec 29 01:48:05.079: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE<BR>
7015: 006961: Dec 29 01:48:05.079: CCSIP-SPI-CONTROL: sip_stats_status_code<BR>
7016: 006962: Dec 29 01:48:05.079: ****Adding to UAS Response table<BR>
7017:<BR>
7018: 006963: Dec 29 01:48:05.079: Previous Hop 65.67.76.41:5060<BR>
7019:<BR>
7020: 006964: Dec 29 01:48:05.079: 0x63CEA7B8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_RECD_INVITE, SUBSTATE_NONE)<BR>
7021: 006965: Dec 29 01:48:05.083: Sent:<BR>
7022: SIP/2.0 100 Trying<BR>
7024: From: "5462000" <sip:<A HREF="mailto:5462000@65.67.76.41"><U>5462000@65.67.76.41</U></A>>;tag=as3e9b26ba<BR>
7025: To: <sip:<A HREF="mailto:6329900@10.1.3.28"><U>6329900@10.1.3.28</U></A>>;tag=6231D0-207E<BR>
7027: Call-ID: <A HREF="mailto:023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41"><U>023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41</U></A><BR>
7028: Server: Cisco-SIPGateway/IOS-12.x<BR>
7030: Content-Length: 0<BR>
7032:<BR>
7034: 006966: Dec 29 01:48:05.083: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING<BR>
7035: 006967: Dec 29 01:48:05.083: ccsip_report_digit_control: enable=0:<BR>
7036: 006968: Dec 29 01:48:05.083: ccsip_report_digit_control: disabled.<BR>
7038: 006970: Dec 29 01:48:05.083: ccsip_report_digit_control: enable=0:<BR>
7040: 006972: Dec 29 01:48:05.083: CCSIP-SPI-CONTROL: act_recdinvite_proceeding<BR>
7041: 006973: Dec 29 01:48:05.083: CCSIP-SPI-CONTROL: act_recdinvite_proceeding<BR>
7042: 006974: Dec 29 01:48:05.083: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING<BR>
7044: 006976: Dec 29 01:48:05.083: ccsip_report_digit_control: disabled.<BR>
7045: 006977: Dec 29 01:48:05.083: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING<BR>
7046: 006978: Dec 29 01:48:05.083: ccsip_report_digit_control: enable=0:<BR>
7050: 006982: Dec 29 01:48:05.087: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT<BR>
7052: 006984: Dec 29 01:48:05.087: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE<BR>
7053: 006985: Dec 29 01:48:05.087: CCSIP-SPI-CONTROL: sip_stats_status_code<BR>
7055: 006987: Dec 29 01:48:05.087: Sent:<BR>
7056: SIP/2.0 500 Internal Server Error<BR>
7057: Via: SIP/2.0/UDP 65.67.76.41:5060;branch=z9hG4bK72629db3<BR>
7058: From: "5462000" <sip:<A HREF="mailto:5462000@65.67.76.41"><U>5462000@65.67.76.41</U></A>>;tag=as3e9b26ba<BR>
7059: To: <sip:<A HREF="mailto:6329900@10.1.3.28"><U>6329900@10.1.3.28</U></A>>;tag=6231D0-207E<BR>
7060: Date: Wed, 29 Dec 2004 01:48:05 GMT<BR>
7061: Call-ID: <A HREF="mailto:023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41"><U>023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41</U></A><BR>
7062: Server: Cisco-SIPGateway/IOS-12.x<BR>
7063: CSeq: 102 INVITE<BR>
7064: Content-Length: 0<BR>
7065:<BR>
7066:<BR>
7067:<BR>
7068: 006988: Dec 29 01:48:05.119: Received:<BR>
7069: ACK sip:6329900@10.1.3.28 SIP/2.0<BR>
7070: Via: SIP/2.0/UDP 65.67.76.41:5060;branch=z9hG4bK72629db3<BR>
7071: From: "5462000" <sip:<A HREF="mailto:5462000@65.67.76.41"><U>5462000@65.67.76.41</U></A>>;tag=as3e9b26ba<BR>
7072: To: <sip:<A HREF="mailto:6329900@10.1.3.28"><U>6329900@10.1.3.28</U></A>>;tag=6231D0-207E<BR>
7073: Contact: <sip:<A HREF="mailto:5462000@65.67.76.41"><U>5462000@65.67.76.41</U></A>><BR>
7074: Call-ID: <A HREF="mailto:023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41"><U>023bbbe91f61ad7529d14ffb2be36b0d@65.67.76.41</U></A><BR>
7076: User-Agent: Asterisk PBX<BR>
7078:<BR>
7080:<BR>
7081: 006989: Dec 29 01:48:05.119: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 65.67.76.41:5060<BR>
7083:<BR>
7084: 006991: Dec 29 01:48:05.119: CCSIP-SPI-CONTROL: act_disconnecting_new_message<BR>
7087: 006994: Dec 29 01:48:05.119: CCSIP-SPI-CONTROL: sip_stats_method<BR>
7089: 006996: Dec 29 01:48:05.119: sipSPIIcpifUpdate :CallState: 2 Playout: 0 DiscTime:643532 ConnTime 0<BR>
7090:<BR>
7092: 006998: Dec 29 01:48:05.119: The Call Setup Information is :<BR>
7094: State of The Call : STATE_DEAD<BR>
7096: Calling Number : 5462000<BR>
7097: Called Number : 6329900<BR>
7098: Negotiated Codec : g711ulaw<BR>
7099: Negotiated Codec Bytes : 160<BR>
7102:<BR>
7103: 006999: Dec 29 01:48:05.119:<BR>
7104: Source IP Address (Sig ): 10.1.3.28<BR>
7105: Source IP Address (Media): 0.0.0.0<BR>
7106: Source IP Port (Media): 0<BR>
7107: Destn IP Address (Media): 65.67.76.41<BR>
7108: Destn IP Port (Media): 11980<BR>
7109: Destn SIP Req Addr:Port : 65.67.76.41:0<BR>
7110: Destn SIP Resp Addr:Port : 65.67.76.41:5060<BR>
7111: Destination Name : 65.67.76.41<BR>
7112:<BR>
7113: 007000: Dec 29 01:48:05.119:<BR>
7114: Disconnect Cause (CC) : 63<BR>
7115: Disconnect Cause (SIP) : 500<BR>
7116:<BR>
7117: 007001: Dec 29 01:48:05.119: ****Deleting from UAS Request table<BR>
7118:<BR>
7119: 007002: Dec 29 01:48:05.119: ****Deleting from UAS Response table<BR>
7120:<BR>
7121: 007003: Dec 29 01:48:05.119: Removing call id B<BR>
7122:<BR>
7123: 007004: Dec 29 01:48:05.123: freeing ccb 63CEA7B8<BR>
7124:<BR>
<BR>
The only difference that I can see is that the non-functional access server is 3 hops from the VoIP server and the ones that are functional are only 2 hops.... I do not think this makes any difference, but I thought I would include it.<BR>
<BR>
Thanks for all your assistance in advance.<BR>
<BR>
Stephen
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