1 Reply Latest reply on Aug 14, 2015 9:12 AM by naff

    SIP to PRI handoff

    naff New Member

      Hi Guys,

       

      I am new on this forum and this is my first time to post a thread. I hope I am on the right page.

       

      I currently have a TA908e that has a SIP trunk to SIP server and will deliver a PRI to PBX. I successfully managed to do a test call from a phone behind the PBX to a phone behind the SIP server.
      But I am getting a SIP/2.0 603 Decline when calling from a phone behind the SIP server to a phone behind the PBX.

       

      I was hoping that someone might be able to help me on troubleshooting the issue. TIA!

       

      Regards,

      Naf

       

      --------------------------------------------------------------------

      Voice configs on the TA:

       

      !

      voice feature-mode network

      voice forward-mode network

      !

      voice codec-list TEST

        codec g711ulaw

      !

      voice trunk T01 type sip

        no reject-external

        sip-server primary 192.168.0.200

        codec-group TEST

      !

      voice trunk T02 type isdn

        resource-selection circular descending

        connect isdn-group 1

        rtp delay-mode adaptive

      !

      !

      voice grouped-trunk SIP

        no description

        trunk T01

        accept $ cost 0

      !

      !

      voice grouped-trunk PRI

        no description

        trunk T02

        accept $ cost 0

      !

      -----------------------------------------------------------------------------------------------------

       

      Debug sip stack message output when a call from SIP server was made.

       

      16:37:53 SIP.STACK MSG     Rx: UDP src=192.168.0.200:5060 dst=10.1.3.62:5060

      16:37:53 SIP.STACK MSG         INVITE sip:9542271703@10.1.3.62 SIP/2.0

      16:37:53 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK631969cb;rport

      16:37:53 SIP.STACK MSG         Max-Forwards: 70

      16:37:53 SIP.STACK MSG         From: "1000" <sip:1000@192.168.0.200>;tag=as6e75bf04

      16:37:53 SIP.STACK MSG         To: <sip:9542271703@10.1.3.62>

      16:37:53 SIP.STACK MSG         Contact: <sip:1000@192.168.0.200:5060>

      16:37:53 SIP.STACK MSG         Call-ID: 05a5ecb00a2d47572bc66a356e084044@192.168.0.200:5060

      16:37:53 SIP.STACK MSG         CSeq: 102 INVITE

      16:37:53 SIP.STACK MSG         User-Agent: WcS-SoNuS-LaB

      16:37:53 SIP.STACK MSG         Date: Fri, 14 Aug 2015 13:47:46 GMT

      16:37:53 SIP.STACK MSG         Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

      16:37:53 SIP.STACK MSG         Supported: replaces, timer

      16:37:53 SIP.STACK MSG         Content-Type: application/sdp

      16:37:53 SIP.STACK MSG         Content-Length: 237

      16:37:53 SIP.STACK MSG

      16:37:53 SIP.STACK MSG         v=0

      16:37:53 SIP.STACK MSG         o=root 245192180 245192180 IN IP4 192.168.0.200

      16:37:53 SIP.STACK MSG         s=Asterisk PBX 1.8.32.3

      16:37:53 SIP.STACK MSG         c=IN IP4 192.168.0.200

      16:37:53 SIP.STACK MSG         t=0 0

      16:37:53 SIP.STACK MSG         m=audio 13900 RTP/AVP 0 101

      16:37:53 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

      16:37:53 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

      16:37:53 SIP.STACK MSG         a=fmtp:101 0-16

      16:37:53 SIP.STACK MSG         a=ptime:20

      16:37:53 SIP.STACK MSG         a=sendrecv

      16:37:53 SIP.STACK MSG

      16:37:53 SIP.STACK MSG     Tx: UDP src=10.1.3.62:5060 dst=192.168.0.200:5060

      16:37:53 SIP.STACK MSG         SIP/2.0 100 Trying

      16:37:53 SIP.STACK MSG         From: "1000"<sip:1000@192.168.0.200>;tag=as6e75bf04

      16:37:53 SIP.STACK MSG         To: <sip:9542271703@10.1.3.62>

      16:37:53 SIP.STACK MSG         Call-ID: 05a5ecb00a2d47572bc66a356e084044@192.168.0.200:5060

      16:37:53 SIP.STACK MSG         CSeq: 102 INVITE

      16:37:53 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.0.200:5060;rport=5060;branch=z9hG4bK631969cb

      16:37:53 SIP.STACK MSG         Contact: <sip:10.1.3.62:5060;transport=UDP>

      16:37:53 SIP.STACK MSG         Supported: 100rel,replaces

      16:37:53 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

      16:37:53 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e/16.05.00.E

      16:37:53 SIP.STACK MSG         Content-Length: 0

      16:37:53 SIP.STACK MSG

      16:37:53 SIP.STACK MSG     Tx: UDP src=10.1.3.62:5060 dst=192.168.0.200:5060

      16:37:53 SIP.STACK MSG         SIP/2.0 603 Decline

      16:37:53 SIP.STACK MSG         From: "1000"<sip:1000@192.168.0.200>;tag=as6e75bf04

      16:37:53 SIP.STACK MSG         To: <sip:9542271703@10.1.3.62>;tag=2fe8320-a01033e-13c4-e9e1-48840da9-e9e1

      16:37:53 SIP.STACK MSG         Call-ID: 05a5ecb00a2d47572bc66a356e084044@192.168.0.200:5060

      16:37:53 SIP.STACK MSG         CSeq: 102 INVITE

      16:37:53 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.0.200:5060;rport=5060;branch=z9hG4bK631969cb

      16:37:53 SIP.STACK MSG         Supported: 100rel,replaces

      16:37:53 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

      16:37:53 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e/16.05.00.E

      16:37:53 SIP.STACK MSG         Content-Length: 0

      16:37:53 SIP.STACK MSG

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