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gerkhs
New Contributor III

Outbound through SIP trunk DID

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Hello all. On a 7060, I got 2 FXO lines that are used by several users. That works fine. However, I have a separate user that I need to allow outbound calls for. When this user, who's not part of the operator (I don't want him to be) dials out, his phone uses one of those FXO lines. I need him to instead use the DID (on a SIP trunk) to dial out. Is this possible or are outbound calls only going to go out the FXO lines? I created a separate COS for this specific use, allowing outbout.

These DIDs and FXO Lines are both local for the country.

Thank you very much for any help.

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gouc
New Contributor III

Re: Outbound through SIP trunk DID

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You may want to look at this document. https://supportforums.adtran.com/docs/DOC-1862

 

SABR is a feature on AOS voice products that enhances call routing services by routing calls based on either source trunk or ANI information. By routing calls based on this information, rather than the standard dialed number identification service (DNIS) information, more flexible and user-centered networks can be created. For example, using SABR allows faxes and modems to be limited to user-specified trunks for connections, as well as restricts the types of calls certain users are allowed to dial, while maintaining full access for others. SABR can allow certain users (for example, hotel guests) to be able to only dial certain numbers out a specified trunk group (for example, 911) while allowing other users (for example, front desk personnel) full access to the trunk group. In essence, SABR is a feature that can restrict the access of certain trunks (sources) and certain users (ANI) to a configured trunk group.

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Anonymous
Not applicable

Re: Outbound through SIP trunk DID

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For calls to go out the SIP trunk, the trunk group for the SIP trunk has to allow those calls.  Also, for the user to be able to place specific types of calls, the user's CoS must allow those call types.  Do you have an outbound accept template on the SIP trunk that allows (matches) the call patterns the user is dialing outbound?

You could run a "debug voice summary", and see what number is hitting the switchboard and what message the system is generating and what trunk it is trying to send the call out to.

gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

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Thank you jbicknell. I did it and I see the call going out the wrong trunk still. I replaced numbers and letters with **** for privacy reasons. Here's the output. I actually need the call to go out trunk T01 and not T03. I'm including the COS config for that user and the T01 permit template config.

12:18:34.556 VOICE.SUMMARY voice user 4047 cos allowed the call to Local

12:18:34.559 VOICE.SUMMARY 4047 is calling T03 (6112****).

12:18:38.338 VOICE.SUMMARY RTP for Call from 4047 to 6112****:  Codec PCMU

12:18:38.339 VOICE.SUMMARY 4047 is connected to T03 (6112****)

12:19:08.957 VOICE.SUMMARY Call from 4047 to T03 (6112****) ended by 4047: normal clearing

------

voice class-of-service Re******

  call-privilege extensions

  call-privilege local

  call-privilege long-distance

  overhead-paging

  permit-template NXX-XXXX

-------

voice grouped-trunk SIP_TO_*****

  trunk T01

  accept NXX-XXXX cost 0

  reject 976-XXXX

Anonymous
Not applicable

Re: Outbound through SIP trunk DID

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The class of service looks to be working correctly as the call was placed and allowed by the user.  I'm assuming that is a 7-digit pattern they are dialing for local calls?  What does the trunk group config look like for T03's trunk group?

gouc
New Contributor III

Re: Outbound through SIP trunk DID

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You may want to look at this document. https://supportforums.adtran.com/docs/DOC-1862

 

SABR is a feature on AOS voice products that enhances call routing services by routing calls based on either source trunk or ANI information. By routing calls based on this information, rather than the standard dialed number identification service (DNIS) information, more flexible and user-centered networks can be created. For example, using SABR allows faxes and modems to be limited to user-specified trunks for connections, as well as restricts the types of calls certain users are allowed to dial, while maintaining full access for others. SABR can allow certain users (for example, hotel guests) to be able to only dial certain numbers out a specified trunk group (for example, 911) while allowing other users (for example, front desk personnel) full access to the trunk group. In essence, SABR is a feature that can restrict the access of certain trunks (sources) and certain users (ANI) to a configured trunk group.

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gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

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Here it is for T03.

voice grouped-trunk "LOCAL TRUNKS"

  trunk T03

  trunk T04

  accept NXX-XXXX cost 0

  accept 1-NXX-NXX-XXXX cost 0

  accept 1-800-NXX-XXXX cost 0

  accept 1-888-NXX-XXXX cost 0

  accept 1-877-NXX-XXXX cost 0

  accept 1-866-NXX-XXXX cost 0

  accept 1-855-NXX-XXXX cost 0

  accept 011-$ cost 0

  accept 911 cost 0

  accept 0-NXX-NXX-XXXX cost 0

  accept 6$ cost 0

  accept 001-$ cost 0

  reject 976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

Anonymous
Not applicable

Re: Outbound through SIP trunk DID

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The reason the call is going out trunk group with T03 (and out trunk T03) is because that trunk group's accept digit pattern is a more specific match (6$) to the number dialed than the Nxx-xxxx pattern.  Trunk group accept patterns, combined with associated costs for that type of call, determine which trunk is selected for a given outbound call.  The lower the cost and the more specific match and lowest trunk number will all give that trunk group (trunk) preference.

If you want the outbound trunk selected per call to be based on the user placing the call, then the SABR configuration guide mentioned in another comment to this post would assist you in setting that up.

gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

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Thank you both. I will take a look at the doc, play with it and let you know how it goes.

Regards

gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

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All, I've configured SABR and still cannot force this 4047 extension to go out the specific T01 Trunk. Here's the config

!

voice ani-list AR***

  ani 4047

!

voice trunk-list AR***Trunk

  trunk T01

--

Here's the Trunk

voice grouped-trunk SIP_TO_C***

  trunk T01

  accept NXX-XXXX cost 0

  accept NXXX-XXXX cost 0 (The number I'm calling from the extension is 8 digits) It's a cell and it helps me confirm whether it works or not)

  reject 976-XXXX

  permit list AR***

  !deny all other trunks

  !deny all other ani

Anonymous
Not applicable

Re: Outbound through SIP trunk DID

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The other trunk group could still be accepting the call because it has a more specific accept match for the number you are dialing.  (6$ is more specific than NXXX-XXXX).  So, you can make the other trunk group accept number more generic OR you could do a "deny list AR***" on the other trunk group.  Let us know if that helps!

gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

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Thank you Jbicknell. I did both, now. I see the user hitting the right trunk now. Here's the debug voice summary output. It's just not going out. I'm wondering if the trunk is not setup for outbound on the carrier's side. The user get's a busy tone. 

09:16:18.157 VOICE.SUMMARY voice user 4047 cos allowed the call to Local

09:16:18.160 VOICE.SUMMARY 4047 is calling T01 (611*****).

09:16:18.174 VOICE.SUMMARY Call from 4047 to T01 (611*****) ended by T01: no user responding

Anonymous
Not applicable

Re: Outbound through SIP trunk DID

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You said you are dialing 8 digits, right?  The carrier might be expecting a 7-digit number.  If so, you can configure DNIS substitution on the Trunk Account to replace the 8-digit number with a 7-digit number.

In the trunk group, you would enter:

match dnis "6XXX-XXXX" substitute "XXX-XXXX"

To see more information on what is happening on the SIP communication, you could run a "debug sip stack messages".

gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

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Nothing yet. Here's the output. I really appreciate all the help.

13:57:39.190 VOICE.SUMMARY voice user 4047 cos allowed the call to Local

13:57:39.193 VOICE.SUMMARY 4047 is calling T01 (611&&&&&).

13:57:39.194 VOICE.SUMMARY DNIS Substitution: dialed number 611&&&&& -> 11&&&&&

13:57:39.207 VOICE.SUMMARY Call from 4047 to T01 (611&&&&&) ended by T01: no user responding

%%%%%%_7060#

%%%%%%_7060#debug sip stack messages

%%%%%%_7060#

13:59:22.483 SIP.STACK MSG     Rx: UDP src=192.168.XX.64:5060 dst=192.168.XX.1:5060

13:59:22.483 SIP.STACK MSG         INVITE sip:9611&&&&&@192.168.XX.1:5060 SIP/2.0

13:59:22.483 SIP.STACK MSG         Max-Forwards: 70

13:59:22.483 SIP.STACK MSG         Content-Length: 436

13:59:22.483 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.XX.64:5060;branch=z9hG4bKca84ea30c

13:59:22.484 SIP.STACK MSG         Call-ID: a58c4be7025bc94d806e0aeff17f8135@192.168.XX.64

13:59:22.484 SIP.STACK MSG         From: AR**** r***** - rec**** <sip:4047@192.168.XX.1:5060>;tag=ffe2a75e9b4af94

13:59:22.484 SIP.STACK MSG         To: 9611&&&&& <sip:9611&&&&&@192.168.XX.1:5060>

13:59:22.484 SIP.STACK MSG         CSeq: 877744988 INVITE

13:59:22.484 SIP.STACK MSG         Supported: 100rel

13:59:22.484 SIP.STACK MSG         Supported: timer

13:59:22.485 SIP.STACK MSG         Allow: SUBSCRIBE, NOTIFY, REFER, OPTIONS, MESSAGE, INVITE, ACK, CANCEL, BYE, INFO

13:59:22.485 SIP.STACK MSG         Content-Type: application/sdp

13:59:22.485 SIP.STACK MSG         Contact: AR**** r***** - rec**** <sip:4047@192.168.XX.64:5060>

13:59:22.485 SIP.STACK MSG         Supported: replaces

13:59:22.485 SIP.STACK MSG         User-Agent: Adtran-SIP-IP706/v1.3.13

13:59:22.485 SIP.STACK MSG

13:59:22.486 SIP.STACK MSG         v=0

13:59:22.486 SIP.STACK MSG         o=MxSIP 0 356616556 IN IP4 192.168.XX.64

13:59:22.486 SIP.STACK MSG         s=SIP Call

13:59:22.486 SIP.STACK MSG         c=IN IP4 192.168.XX.64

13:59:22.486 SIP.STACK MSG         t=0 0

13:59:22.486 SIP.STACK MSG         m=audio 3000 RTP/AVP 0 18 103 102 107 104 105 101 8

13:59:22.486 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

13:59:22.487 SIP.STACK MSG         a=rtpmap:18 G729/8000

13:59:22.487 SIP.STACK MSG         a=rtpmap:103 BV16/8000

13:59:22.487 SIP.STACK MSG         a=rtpmap:102 BV32/16000

13:59:22.487 SIP.STACK MSG         a=rtpmap:107 L16/16000

13:59:22.487 SIP.STACK MSG         a=rtpmap:104 PCMU/16000

13:59:22.487 SIP.STACK MSG         a=rtpmap:105 PCMA/16000

13:59:22.488 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

13:59:22.488 SIP.STACK MSG         a=rtpmap:8 PCMA/8000

13:59:22.488 SIP.STACK MSG         a=fmtp:101 0-15

13:59:22.488 SIP.STACK MSG         a=ptime:20

13:59:22.488 SIP.STACK MSG         a=sendrecv

13:59:22.488 SIP.STACK MSG         a=silenceSupp:off - - - -

13:59:22.489 SIP.STACK MSG

13:59:22.492 SIP.STACK MSG     Tx: UDP src=192.168.XX.1:5060 dst=192.168.XX.64:5060

13:59:22.493 SIP.STACK MSG         SIP/2.0 100 Trying

13:59:22.493 SIP.STACK MSG         From: AR**** r***** - rec**** <sip:4047@192.168.XX.1:5060>;tag=ffe2a75e9b4af94

13:59:22.493 SIP.STACK MSG         To: 9611&&&&& <sip:9611&&&&&@192.168.XX.1:5060>

13:59:22.493 SIP.STACK MSG         Call-ID: a58c4be7025bc94d806e0aeff17f8135@192.168.XX.64

13:59:22.493 SIP.STACK MSG         CSeq: 877744988 INVITE

13:59:22.493 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.XX.64:5060;branch=z9hG4bKca84ea30c

13:59:22.494 SIP.STACK MSG         Contact: <sip:9611&&&&&@192.168.XX.1:5060>

13:59:22.494 SIP.STACK MSG         Supported: 100rel,replaces

13:59:22.494 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

13:59:22.494 SIP.STACK MSG         User-Agent: ADTRAN_NetVanta_7060/A5.02.00.E

13:59:22.494 SIP.STACK MSG         Content-Length: 0

13:59:22.494 SIP.STACK MSG

13:59:22.496 VOICE.SUMMARY voice user 4047 cos allowed the call to Local

13:59:22.499 VOICE.SUMMARY 4047 is calling T01 (611&&&&&).

13:59:22.500 VOICE.SUMMARY DNIS Substitution: dialed number 611&&&&& -> 11&&&&&

13:59:22.503 SIP.STACK MSG     Tx: UDP src=172.^^.55.34:5060 dst=172.^^.50.50:5060

13:59:22.503 SIP.STACK MSG         INVITE sip:11&&&&&@172.^^.50.50:5060 SIP/2.0

13:59:22.503 SIP.STACK MSG         From: "AR**** r***** - rec****" <sip:304****@172.^^.50.50:5060;transport=UDP>;tag=5530fd0-7f000001-13c4-2043f0-d1366302-2043f0

13:59:22.504 SIP.STACK MSG         To: <sip:11&&&&&@172.^^.50.50:5060>

13:59:22.504 SIP.STACK MSG         Call-ID: 55c1ed0-7f000001-13c4-2043f0-ba6d5f1e-2043f0@172.^^.50.50

13:59:22.504 SIP.STACK MSG         CSeq: 1 INVITE

13:59:22.504 SIP.STACK MSG         Via: SIP/2.0/UDP 172.^^.55.34:5060;branch=z9hG4bK-2043f0-7e096352-6ca46020

13:59:22.504 SIP.STACK MSG         Max-Forwards: 70

13:59:22.504 SIP.STACK MSG         Supported: 100rel,replaces

13:59:22.505 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

13:59:22.505 SIP.STACK MSG         User-Agent: ADTRAN_NetVanta_7060/A5.02.00.E

13:59:22.505 SIP.STACK MSG         Contact: <sip:304****@172.^^.55.34:5060;transport=UDP>

13:59:22.505 SIP.STACK MSG         Content-Type: application/sdp

13:59:22.505 SIP.STACK MSG         Content-Length: 313

13:59:22.505 SIP.STACK MSG

13:59:22.505 SIP.STACK MSG         v=0

13:59:22.506 SIP.STACK MSG         o=MxSIP 0 356616556 IN IP4 172.^^.55.34

13:59:22.506 SIP.STACK MSG         s=SIP Call

13:59:22.506 SIP.STACK MSG         c=IN IP4 172.^^.55.34

13:59:22.506 SIP.STACK MSG         t=0 0

13:59:22.506 SIP.STACK MSG         m=audio 50106 RTP/AVP 0 18 8 101

13:59:22.506 SIP.STACK MSG         a=ptime:20

13:59:22.507 SIP.STACK MSG         a=sendrecv

13:59:22.507 SIP.STACK MSG         a=silenceSupp:off - - - -

13:59:22.507 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

13:59:22.507 SIP.STACK MSG         a=rtpmap:18 G729/8000

13:59:22.507 SIP.STACK MSG         a=fmtp:18 annexb=no

13:59:22.507 SIP.STACK MSG         a=rtpmap:8 PCMA/8000

13:59:22.508 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

13:59:22.508 SIP.STACK MSG         a=fmtp:101 0-15

13:59:22.508 SIP.STACK MSG

13:59:22.512 SIP.STACK MSG     Rx: UDP src=172.^^.50.50:5060 dst=172.^^.55.34:5060

13:59:22.512 SIP.STACK MSG         SIP/2.0 100 Trying

13:59:22.512 SIP.STACK MSG         From: "AR**** r***** - rec****" <sip:304****@172.^^.50.50:5060;transport=UDP>;tag=5530fd0-7f000001-13c4-2043f0-d1366302-2043f0

13:59:22.512 SIP.STACK MSG         To: <sip:11&&&&&@172.^^.50.50:5060>

13:59:22.512 SIP.STACK MSG         Call-ID: 55c1ed0-7f000001-13c4-2043f0-ba6d5f1e-2043f0@172.^^.50.50

13:59:22.512 SIP.STACK MSG         CSeq: 1 INVITE

13:59:22.513 SIP.STACK MSG         Via: SIP/2.0/UDP 172.^^.55.34:5060;branch=z9hG4bK-2043f0-7e096352-6ca46020

13:59:22.513 SIP.STACK MSG

13:59:22.515 SIP.STACK MSG     Rx: UDP src=172.^^.50.50:5060 dst=172.^^.55.34:5060

13:59:22.515 SIP.STACK MSG         SIP/2.0 480 No Routes Found

13:59:22.515 SIP.STACK MSG         Via: SIP/2.0/UDP 172.^^.55.34:5060;branch=z9hG4bK-2043f0-7e096352-6ca46020

13:59:22.515 SIP.STACK MSG         From: "AR**** r***** - rec****" <sip:304****@172.^^.50.50:5060;transport=UDP>;tag=5530fd0-7f000001-13c4-2043f0-d1366302-2043f0

13:59:22.515 SIP.STACK MSG         To: <sip:11&&&&&@172.^^.50.50:5060>;tag=aprqngfrt-m6dmqq1000020

13:59:22.515 SIP.STACK MSG         Call-ID: 55c1ed0-7f000001-13c4-2043f0-ba6d5f1e-2043f0@172.^^.50.50

13:59:22.516 SIP.STACK MSG         CSeq: 1 INVITE

13:59:22.516 SIP.STACK MSG

13:59:22.518 SIP.STACK MSG     Tx: UDP src=172.^^.55.34:5060 dst=172.^^.50.50:5060

13:59:22.518 SIP.STACK MSG         ACK sip:11&&&&&@172.^^.50.50:5060;transport=UDP SIP/2.0

13:59:22.518 SIP.STACK MSG         From: "AR**** r***** - rec****" <sip:304****@172.^^.50.50:5060;transport=UDP>;tag=5530fd0-7f000001-13c4-2043f0-d1366302-2043f0

13:59:22.518 SIP.STACK MSG         To: <sip:11&&&&&@172.^^.50.50:5060>;tag=aprqngfrt-m6dmqq1000020

13:59:22.519 SIP.STACK MSG         Call-ID: 55c1ed0-7f000001-13c4-2043f0-ba6d5f1e-2043f0@172.^^.50.50

13:59:22.519 SIP.STACK MSG         CSeq: 1 ACK

13:59:22.519 SIP.STACK MSG         Via: SIP/2.0/UDP 172.^^.55.34:5060;branch=z9hG4bK-2043f0-7e096352-6ca46020

13:59:22.519 SIP.STACK MSG         Max-Forwards: 70

13:59:22.519 SIP.STACK MSG         Supported: 100rel,replaces

13:59:22.519 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

13:59:22.520 SIP.STACK MSG         User-Agent: ADTRAN_NetVanta_7060/A5.02.00.E

13:59:22.520 SIP.STACK MSG         Contact: <sip:304****@172.^^.55.34:5060;transport=UDP>

13:59:22.520 SIP.STACK MSG         Content-Length: 0

13:59:22.520 SIP.STACK MSG

13:59:22.521 VOICE.SUMMARY Call from 4047 to T01 (611&&&&&) ended by T01: no user responding

13:59:22.523 SIP.STACK MSG     Tx: UDP src=192.168.XX.1:5060 dst=192.168.XX.64:5060

13:59:22.523 SIP.STACK MSG         SIP/2.0 480 Temporarily Unavailable

13:59:22.523 SIP.STACK MSG         From: AR*** R***** - rec**** <sip:4047@192.168.XX.1:5060>;tag=ffe2a75e9b4af94

13:59:22.523 SIP.STACK MSG         To: 9611&&&&& <sip:9611&&&&&@192.168.XX.1:5060>;tag=552aef0-7f000001-13c4-2043f0-a5d67c6d-2043f0

13:59:22.523 SIP.STACK MSG         Call-ID: a58c4be7025bc94d806e0aeff17f8135@192.168.XX.64

13:59:22.523 SIP.STACK MSG         CSeq: 877744988 INVITE

13:59:22.524 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.XX.64:5060;branch=z9hG4bKca84ea30c

13:59:22.524 SIP.STACK MSG         Supported: 100rel,replaces

13:59:22.524 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

13:59:22.524 SIP.STACK MSG         User-Agent: ADTRAN_NetVanta_7060/A5.02.00.E

13:59:22.524 SIP.STACK MSG         Content-Length: 0

13:59:22.524 SIP.STACK MSG

13:59:22.578 SIP.STACK MSG     Rx: UDP src=192.168.XX.64:5060 dst=192.168.XX.1:5060

13:59:22.578 SIP.STACK MSG         ACK sip:9611&&&&&@192.168.XX.1:5060 SIP/2.0

13:59:22.578 SIP.STACK MSG         Max-Forwards: 70

13:59:22.578 SIP.STACK MSG         Content-Length: 0

13:59:22.578 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.XX.64:5060;branch=z9hG4bKca84ea30c

13:59:22.578 SIP.STACK MSG         Call-ID: a58c4be7025bc94d806e0aeff17f8135@192.168.XX.64

13:59:22.579 SIP.STACK MSG         From: AR**** r***** - rec**** <sip:4047@192.168.XX.1:5060>;tag=ffe2a75e9b4af94

13:59:22.579 SIP.STACK MSG         To: 9611&&&&& <sip:9611&&&&&@192.168.XX.1:5060>;tag=552aef0-7f000001-13c4-2043f0-a5d67c6d-2043f0

13:59:22.579 SIP.STACK MSG         CSeq: 877744988 ACK

13:59:22.579 SIP.STACK MSG         User-Agent: Adtran-SIP-IP706/v1.3.13

13:59:22.579 SIP.STACK MSG

Anonymous
Not applicable

Re: Outbound through SIP trunk DID

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It looks like there is no route to that number as presented out that trunk (11&&).  So, either the number you are sending in that format is not the number expected or the number is not reachable through that circuit/connection.

gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

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Jbicknell, I talked to the provider and they say I have to send the first 7 digits. They have also confirmed that this SIP Trunk allows outbound calls so they claim the problem is on my end. I entered this on the trunk, like you suggested, but the call does not go out

"match dnis "6XXX-XXXX" substitute "XXX-XXXX"

Anonymous
Not applicable

Re: Outbound through SIP trunk DID

Jump to solution

What is the actual dialable number of the cell phone you are calling, that it can be reached at through the provider (6$$$, etc.)?  (you don't have to send us the exact number, but the pattern would be helpful).  You want to make sure you are sending the right digits to match the actual phone number so the provider can route the call.  For example, if the public number is 645-5111, and you are dialing 64551112 and only sending 455-1112, it won't match.  With the DNIS substitution we discussed, it will strip the first digit (6) and send the rest of the number. 

gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

Jump to solution

Here's the thing. We cannot drop the 6 since all cell phones need a 6 in front where the trunk is located. we need to send the first 6 digits including the 6. However, I just tested calling a 7 digit number and I get the same output. It seems like there's something else preventing outbound through this trunk.

12:28:11.296 VOICE.SUMMARY voice user 4047 cos allowed the call to Local

12:28:11.300 VOICE.SUMMARY 4047 is calling T01 (344####).

12:28:11.312 VOICE.SUMMARY Call from 4047 to T01 (344####) ended by T01: no user responding

Anonymous
Not applicable

Re: Outbound through SIP trunk DID

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If you need the 6, then the DNIS subsitution would be match dnis "6xxx-xxxx" substitute "6xx-xxxx".  But, your debug earlier does show the system is trying to send a call out the SIP trunk via the SIP messaging you saw, and you are receiving messages back (far end being the 172.X.X.50).  Could the provider verify that they are seeing the signaling/messages come in and then what happens on their end?  Are the numbers you are dialing reachable via other trunks?

gerkhs
New Contributor III

Re: Outbound through SIP trunk DID

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Thank you so much for the help!!! After talking to the provider and doing some traces, they determined I needed to change the IP of the SIP server on their network, in my 7060. SABR did help me with call routing.