Dial Tone issue

[MBTC]

Hi all. I am using CM/SM and SMGR 8.1 with all user using SIP phone (J series). We have two PRIs, one H.323 Trunk with CISCO. We are not getting the dial tone after dialing the TAC of any trunk, it is dead silence. However through normal dial plan IN/OUT calls from PRI and H.323 working fine. Its is the TAC giving problem. We also tried dialing out different number using the TAC but we get busy tone. I also tried adding the TAC in session manager but still no dial tone. DIAL ACCESS=Y in all trunk groups. Any suggestions?

 

[kyle555]

I'd guess it's a CM bug. There's always the question of who's supposed to give ringback - the far end in early media, or the near end to the caller, but I've never had that question with dial tone before

What does the SIP invite look like? Presuming your TAC code is *101 and you're calling 9-1-212-555-1234, is the INVITE to *101 and then the 9-1-212-555-1234 is RFC2833 RTP DTMF to CM? If so, then it's conceivable CM would be able to setup media for that call to the TAC and provide dial tone.

If you see INVITE *101912125551234, then it's impossible because the phone is waiting for a digit timeout. If you traceSM with PPM enabled and "reload complete" on the phone from the SM registration page, you can see the dial patterns sent to the phone in GetAllEndpointConfigResponse or something.

Usually it's just the feature code, but depending on what dialplan analysis and UDP look like, that could be slightly different.

In any case, if you see an INVITE to *101 and the rest of the dialing happens with DTMFs, then I'd think CM should be able to play dial tone and I'd try opening it as a bug.

 

[MBTC]

Hi Kyle, I don see TAC+dialed number or only TAC in traceSM. The TAC of trunk is #667. I entered TAC #667 in dial pattern but still no dial tone. I get busy tone if i dial only TAC and wait, i also get busy tome if if dial TAC+dialout number after few milisecond. One thing i was considering to add TAC in AAR and route to SM but aar table is not accepting the #.
Please guide.

 

[MBTC]

anymore help?

 

[G van Hamburg]

SIP users do not use SM dial patterns, they use the Application Sequenses. When you dial the TAC, the providers should give dialtone. I’d trace CM to see what is going on.

Freelance Certified Avaya Aura Engineer

 

[MBTC]

Hi G Van,
I tested below
1- I dialed a TAC and waited for a tone, after few milisecond i get busy tone.
2- I dialed a TAC along with a mobile number, i again got busy tome after few milisecond, I also dont see the mobile
number in below capture


14:35:24 TRACE STARTED 03/08/2021 CM Release String cold-01.0.890.0-25578
14:35:26 SIP<INVITE sip:8519@uaeat.com;avaya-cm-fnu=off-hook SIP/2.
14:35:26 SIP<0
14:35:26 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:26 9
14:35:26 SIP>SIP/2.0 183 Session Progress
14:35:26 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:26 9
14:35:26 active station 8519 cid 0xa5c
14:35:26 Calling party uses public-unknown-numbering
14:35:28 SIP>SIP/2.0 484 Address Incomplete
14:35:28 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:28 9
14:35:28 SIP<INVITE sip:%23802@uaeat.com SIP/2.0
14:35:28 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:28 9
14:35:28 SIP>SIP/2.0 100 Trying
14:35:28 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:28 9
14:35:28 dial #802
14:35:28 term trunk-group 2 cid 0xa5c
14:35:28 dial #802
14:35:28 seize trunk-group 2 member 14 cid 0xa5c
14:35:28 Setup digits
14:35:28 Calling Number & Name 9733148519 UAE_Test_ext
14:35:28 SIP<ACK sip:8519@uaeat.com;avaya-cm-fnu=off-hook SIP/2.0
14:35:28 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:28 9
14:35:28 dial #802
14:35:28 seize trunk-group 2 member 14 cid 0xa5c
14:35:38 Proceed trunk-group 2 member 14 cid 0xa5c
14:35:38 SIP>SIP/2.0 183 Session Progress
14:35:38 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:38 9
14:35:38 G711MU ssff ps:20
rgn:1 [192.17.2.12]:5004
rgn:3 [192.17.5.21]:2086
14:35:38 xoip options: fax:Relay modemff tty:US uid:0x50003f
xoip ip: [192.17.5.21]:2086
14:35:38 SIP<PRACK sip:%23802@192.17.1.18;transport=tcp;asm=1 SIP/2.0
14:35:38 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:38 9
14:35:38 SIP>SIP/2.0 200 OK
14:35:38 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:38 9
14:35:41 SIP<CANCEL sip:%23802@uaeat.com SIP/2.0
14:35:41 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:41 9
14:35:41 SIP>SIP/2.0 200 OK
14:35:41 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:41 9
14:35:41 SIP>SIP/2.0 487 Request Terminated
14:35:41 Call-ID: fd_6045efdc1a30920d5x3j5a3l1z5ja56y544d73_I851
14:35:41 9
14:35:41 idle station 8519 cid 0xa5c
14:35:44 SIP<INVITE sip:8519@uaeat.com;avaya-cm-fnu=off-hook SIP/2.
14:35:44 SIP<0
14:35:44 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:44 519
14:35:44 SIP>SIP/2.0 183 Session Progress
14:35:44 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:44 519
14:35:44 active station 8519 cid 0xaaf
14:35:44 Calling party uses public-unknown-numbering
14:35:48 SIP>SIP/2.0 484 Address Incomplete
14:35:48 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:48 519
14:35:48 SIP<INVITE sip:%23803@uaeat.com SIP/2.0
14:35:48 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:48 519
14:35:48 SIP>SIP/2.0 100 Trying
14:35:48 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:48 519
14:35:48 dial #803
14:35:48 term trunk-group 3 cid 0xaaf
14:35:48 dial #803
14:35:48 seize trunk-group 3 member 5 cid 0xaaf
14:35:48 Setup digits
14:35:48 Calling Number & Name 8519
14:35:48 SIP<ACK sip:8519@uaeat.com;avaya-cm-fnu=off-hook SIP/2.0
14:35:48 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:48 519
14:35:48 dial #803
14:35:48 seize trunk-group 3 member 5 cid 0xaaf
14:35:58 Proceed trunk-group 3 member 5 cid 0xaaf
14:35:58 SIP>SIP/2.0 183 Session Progress
14:35:58 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:58 519
14:35:58 G711MU ssff ps:20
rgn:1 [192.17.2.12]:5004
rgn:1 [192.17.1.20]:2122
14:35:58 xoip options: fax:Relay modemff tty:US uid:0x500002
xoip ip: [192.17.1.20]:2122
14:35:58 SIP<PRACK sip:%23803@192.17.1.8;transport=tcp;asm=1 SIP/2.0
14:35:58 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:58 519
14:35:58 SIP>SIP/2.0 200 OK
14:35:58 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:35:58 519
14:36:03 SIP<CANCEL sip:%23803@uaeat.com SIP/2.0
14:36:03 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:36:03 519
14:36:03 SIP>SIP/2.0 200 OK
14:36:03 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:36:03 519
14:36:03 SIP>SIP/2.0 487 Request Terminated
14:36:03 Call-ID: 103_6045efef-1a435164l4y5i4y331s65311l29726_I8
14:36:03 519
14:36:03 idle station 8519 cid 0xaaf
14:36:22 TRACE COMPLETE station 8519 cid 0x0 UAE_Test_ext

Any help will be greatly appreciated.

 

[G van Hamburg]

Try the trace on the TAC. List trace TAC. I do not think the station trace will help.

There are no H323 stations on the system? I still wonder if SIP is the issue or the trunk.

You could try and set up remote access to the system and see if that gives some more info.

Sorry but for now, this is all I can offer.

Freelance Certified Avaya Aura Engineer

 

[MBTC]

Ok, i will trace TAC and post back here.

 

[smokinjoe2938]

I know on an ISDN trunk you cannot dial out by the tac code so maybe as well with SIP