I am having real trouble setting up outbound SIP calls from my Call Manager Express!
I am using a 2801 running IOS 'c2801-adventerprisek9-mz.124-24.T.bin'. I have around 8 phones running SCCP internally. The Call manager has been working fine for a while now, we recently wanted to make PSTN calls and subscribed to SIP SP for in and outbound calls. After much reading of documentation and forums I have got inbound calls from the SP working. However I can not get outbound calls to work and no amount of playing appears to be working..
voice rtp send-recv ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer fax protocol cisco stun sip bind control source-interface Loopback10 bind media source-interface Loopback10 session transport tcp registrar server expires max 300 min 60 transport switch udp tcp ! ! ! voice class codec 1 codec preference 3 g729r8 codec preference 4 g711alaw codec preference 5 g711ulaw !
! dial-peer voice 2001 voip description * Outbound calling to Landlines via Gradwell.com * destination-pattern 0[1-3]......... voice-class codec 1 session protocol sipv2 session target dns:sip.trunk.gradwell.com session transport tcp dtmf-relay rtp-nte sip-notify ip qos dscp cs3 signaling no vad ! dial-peer voice 2011 voip description * Outbound calling to Mobiles via Gradwell.com * destination-pattern 07......... voice-class codec 1 session protocol sipv2 session target dns:sip.trunk.gradwell.com session transport tcp dtmf-relay rtp-nte sip-notify ip qos dscp cs3 signaling no vad ! ! sip-ua nat symmetric role passive nat symmetric check-media-src no remote-party-id retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 timers connect 100 ! ! telephony-service sdspfarm transcode sessions 32 conference hardware fxo hook-flash max-ephones 24 max-dn 64 ip source-address 192.168.70.1 port 2000 calling-number initiator no service local-directory service phone displayOnWhenIncomingCall 1 service phone specialNumbers 999 system message PMD Inc. cnf-file location flash: cnf-file perphone network-locale GB network-locale 1 GB network-locale 2 GB network-locale 3 GB network-locale 4 GB load 7960-7940 P00308010200 load 7941 SCCP41.8-5-4S load 7942 SCCP42.8-5-4S load 7961 SCCP61.8-5-4S load 7962 SCCP62.8-5-4S load 7970 SCCP70.8-5-4S time-zone 21 time-format 24 date-format dd-mm-yy keepalive 15 max-conferences 4 gain -6 moh flash:/MOH/music-on-hold.au multicast moh 184.108.40.206 port 16834 route 172.17.2.2 172.17.2.6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 !
! ephone-dn 59 dual-line number 0844xxxxxx no-reg both !
My SP has not provided a username and password of the service and instead chooses to do authentication based on my source IP address and CLID.
thanks for the reply. I have tried 'debug ccsip messages' already and do not get any responces. I was thinking this indicates the problem is before the SIP messages are created???? I have enabled 'debug ccsip all' and get the following:
Jan 29 2010 14:06:29.700 GMT: //48/4B307EB28095/SIP/Call/sipSPICallInfo: The Call Setup Information is: Call Control Block (CCB) : 0x69153F8C State of The Call : STATE_DEAD TCP Sockets Used : YES Calling Number : 0844xxxxxxx Called Number : 07xxxxxxxxx Source IP Address (Sig ): 192.168.70.1 Destn SIP Req Addr:Port : 220.127.116.11:5060 Destn SIP Resp Addr:Port : 18.104.22.168:5060 Destination Name : sip.trunk.gradwell.com
Jan 29 2010 14:06:29.704 GMT: //48/4B307EB28095/SIP/Call/sipSPIMediaCallInfo: Number of Media Streams: 1 Media Stream : 1 Negotiated Codec : No Codec Negotiated Codec Bytes : 0 Nego. Codec payload : 255 (tx), 255 (rx) Negotiated Dtmf-relay : 0 Dtmf-relay Payload : 0 (tx), 0 (rx) Source IP Address (Media): 192.168.70.1 Source IP Port (Media): 17490 Destn IP Address (Media): - Destn IP Port (Media): 0 Orig Destn IP Address:Port (Media): [ - ]:0
Jan 29 2010 14:06:29.704 GMT: //48/4B307EB28095/SIP/Call/sipSPICallInfo: Disconnect Cause (CC) : 38 Disconnect Cause (SIP) : 503
I understand SIP code 503 is a server unavaliable. I have placed a call with the SP who just tell me that the server is up and its not their problem.
After one of those 'oh you say your using TCP' with the serivce provider the support guy was kind enough to tell me that they only support SIP over UDP. I removed all the TCP related commands and the outbound SIP trunk works.
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