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naff
New Contributor

SIP to PRI handoff

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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1 Reply
naff
New Contributor

Re: SIP to PRI handoff

Hi Guys,

I managed to find a work around on the issue. I downgraded my AOS from TA900EA-16-05-00-E AOS to TA900EA-A4-11-00-E. Bi-directional traffic is now working.

Thanks.

Naf