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Cisco ATA 186 to Session Manager

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JuanAvaya

JuanAvaya

Technical User
Jul 27, 2015
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Hi everyone,
I am trying to make a Cisco ATA 186 through SIP work with Session Manager.
I have managed to register to SM and make successful calls but ONLY from the ATA to SM. In the opposite direction, the ATA rings but there is no audio on any way and after 15 seconds Avaya drops the call.

Any idea on what could be wrong?
 
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Have you looked at this ?


Have you run a traceSM?

Have you got a hub that you can sit in parralel with the ata?

Without trying to sound harsh , you have given hardly any info with what you have already tried with regards to troubleshooting , run the trace and compare the good call to the bad .. got to be the first step


ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.
 
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Hi montyzummer
I have followed the documentation step by step and performed some traces. However, even though I see a clear difference in the good and bad traces I cannot tell why it is behaving that way. I am pasting both traces below...


GOOD CALL (from ATA to CM)
-----------------------------------------------------------------------------------
10.120.11.8 Communication Manager . SM100
-----------------------------------------------------------------------------------
16:16:39.501 |--INVITE-->| | | (5) T:6512 U:6512
16:16:39.502 |<--Trying--| | | (5) 100 Trying
16:16:39.503 |<--Proxy A-| | | (5) 407 Proxy Authentication Required
16:16:39.569 |----ACK--->| | | (5) sip:6512@10.100.90.26
16:16:39.587 |--INVITE-->| | | (5) T:6512 U:6512
16:16:39.588 |<--Trying--| | | (5) 100 Trying
16:16:39.590 | |--INVITE-->| | (5) T:6512 F:2868 U:6512 P:imsorig
16:16:39.591 | |<--Trying--| | (5) 100 Trying
16:16:39.595 | |<--Ringing-| | (5) 180 Ringing
16:16:39.597 |<--Ringing-| | | (5) 180 Ringing
16:16:39.684 |---PRACK-->| | | (5) sip:6512@10.100.90.18
16:16:39.686 | |---PRACK-->| | (5) sip:6512@10.100.90.18
16:16:39.690 | |<--200 OK--| | (5) 200 OK (PRACK)
16:16:39.691 |<--200 OK--| | | (5) 200 OK (PRACK)
16:16:42.537 | |<--200 OK--| | (5) 200 OK (INVITE)
16:16:42.540 |<--200 OK--| | | (5) 200 OK (INVITE)
16:16:42.613 |----ACK--->| | | (5) sip:6512@10.100.90.18
16:16:42.614 | |----ACK--->| | (5) sip:6512@10.100.90.18
16:16:47.416 | |<----BYE---| | (5) sip:2868@10.120.11.8
16:16:47.418 |<----BYE---| | | (5) sip:2868@10.120.11.8
16:16:47.471 |--200 OK-->| | | (5) 200 OK (BYE)
16:16:47.472 | |--200 OK-->| | (5) 200 OK (BYE)



BAD CALL (from CM to ATA)
-----------------------------------------------------------------------------------------
Communication Manager 10.120.11.8
SM100 MG
-----------------------------------------------------------------------------------------
16:25:00.802 |--INVITE-->| | | (3) T:2868 F:6512 U:2868 P:terminating
16:25:00.803 |<--Trying--| | | (3) 100 Trying
16:25:00.805 |<--INVITE--| | | (3) T:2868 F:6512 U:2868 P:imsterm
16:25:00.806 |--INVITE-->| | | (3) T:2868 F:6512 U:2868 P:termdone
16:25:00.807 |<--Trying--| | | (3) 100 Trying
16:25:00.808 |--Trying-->| | | (3) 100 Trying
16:25:00.809 | |--INVITE-->| | (3) T:2868 F:6512 U:2868
16:25:00.898 | |<--Trying--| | (3) 100 Trying
16:25:00.919 | |<--Ringing-| | (3) 180 Ringing
16:25:00.922 |<--Ringing-| | | (3) 180 Ringing
16:25:00.923 |--Ringing->| | | (3) 180 Ringing
16:25:00.924 |<--Ringing-| | | (3) 180 Ringing
16:25:00.925 |---PRACK-->| | | (3) sip:2868@10.120.11.8
16:25:00.966 | |---PRACK-->| | (3) sip:2868@10.120.11.8
16:25:01.421 | |<--Ringing-| | (3) 180 Ringing
16:25:01.423 |<--Ringing-| | | (3) 180 Ringing
16:25:01.424 |--Ringing->| | | (3) 180 Ringing
16:25:01.426 |<--Ringing-| | | (3) 180 Ringing
16:25:01.466 | |---PRACK-->| | (3) sip:2868@10.120.11.8
16:25:02.421 | |<--Ringing-| | (3) 180 Ringing
16:25:02.423 |<--Ringing-| | | (3) 180 Ringing
16:25:02.424 |--Ringing->| | | (3) 180 Ringing
16:25:02.425 |<--Ringing-| | | (3) 180 Ringing
16:25:02.465 | |---PRACK-->| | (3) sip:2868@10.120.11.8
16:25:04.422 | |<--Ringing-| | (3) 180 Ringing
16:25:04.424 |<--Ringing-| | | (3) 180 Ringing
16:25:04.425 |--Ringing->| | | (3) 180 Ringing
16:25:04.427 |<--Ringing-| | | (3) 180 Ringing
16:25:04.465 | |---PRACK-->| | (3) sip:2868@10.120.11.8
16:25:04.518 | |<--200 OK--| | (3) 200 OK (INVITE)
16:25:04.518 | | |<===G711u=>| (3) RTP 10.100.90.22:2136 <-G711u-> 10.120.11.8:16384
16:25:04.521 |<--200 OK--| | | (3) 200 OK (INVITE)
16:25:04.522 |--200 OK-->| | | (3) 200 OK (INVITE)
16:25:04.525 |<--200 OK--| | | (3) 200 OK (INVITE)
16:25:04.526 |----ACK--->| | | (3) sip:2868@10.120.11.8
16:25:04.567 | |----ACK--->| | (3) sip:2868@10.120.11.8
16:25:05.018 | |<--200 OK--| | (3) 200 OK (INVITE)
16:25:05.020 |<--200 OK--| | | (3) 200 OK (INVITE)
16:25:05.021 |--200 OK-->| | | (3) 200 OK (INVITE)
16:25:05.024 |<--200 OK--| | | (3) 200 OK (INVITE)
16:25:05.024 |----ACK--->| | | (3) sip:2868@10.120.11.8
16:25:05.064 | |----ACK--->| | (3) sip:2868@10.120.11.8
16:25:06.021 | |<--200 OK--| | (3) 200 OK (INVITE)
16:25:08.020 | |<--200 OK--| | (3) 200 OK (INVITE)
16:25:08.466 | |---PRACK-->| | (3) sip:2868@10.120.11.8
16:25:10.020 | |<--200 OK--| | (3) 200 OK (INVITE)
16:25:12.014 | |<--200 OK--| | (3) 200 OK (INVITE)
16:25:12.466 | |---PRACK-->| | (3) sip:2868@10.120.11.8
16:25:14.014 | |<--200 OK--| | (3) 200 OK (INVITE)
16:25:16.019 | |<--200 OK--| | (3) 200 OK (INVITE)
16:25:16.467 | |---PRACK-->| | (3) sip:2868@10.120.11.8
16:25:18.014 | |<----BYE---| | (3) sip:6512@10.100.90.18
16:25:18.015 | |--Proxy A->| | (3) 407 Proxy Authentication Required
16:25:18.079 | |<----BYE---| | (3) sip:6512@10.100.90.18
16:25:18.081 |<----BYE---| | | (3) sip:6512@10.100.90.18
16:25:18.082 |--200 OK-->| | | (3) 200 OK (BYE)
16:25:18.084 | |--200 OK-->| | (3) 200 OK (BYE)
16:25:20.466 | |---PRACK-->| | (3) sip:2868@10.120.11.8
16:25:24.467 | |---PRACK-->| | (3) sip:2868@10.120.11.8


It seems CM is expecting some sort of response from the other side once the communication is stablished so when nothing comes, it drops the call.
 
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So this could be and looks like full/half call model in CM , how are your sig groups defined in regards to ims (what is your CM acting as FS /ES what version is it on). also how does CM route the call to Session manager ?


ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.
 
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IMS enabled? is set to NO on every sig group and CM is configured as an Evolution Server (6.3 SP2.1)
Regarding the routing question what is exactly what you need to know?

I must say that had to read something about IMS as was a new term for me. That led to something else and ended up realizing that SIP to ATA works just fine, the problem is only H.323 to ATA.

Thank for the help
 
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So with the routing i mean how does the CM route calls to the ata , have you got a sip station built in CM (matches the ata extn number)that uses AAR to route the call down the sip trunk to session manager?

Or are you just trying to pass calls direct down the sip trunk?
I

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.
 
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This is set up:
- SIP ext in CM matching ATA
- UDP entry pointing to AAR
- AAR routing the call to the SIP trunk

A user was also created in SM for the ATA of course.
 
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have you got off pbx station mapping set up for the station also?

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.
 
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Yes, I forgot to mentioned it. It is pointing to the SIP trunk
 
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Juan,

Could you please check the codecs on the trunk group as well? Just thinking loud

Thanks
Ganti
 
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