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AVAYA IP Office - SIP/2.0 503 Service Unavailable

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IPOMAN-1986

IPOMAN-1986

Systems Engineer
Dec 16, 2017
13
OM
We have AVAYA IP Office Call center, where agents are using Ameyo application on their PC's to answer or transfer calls.
Ameyo server is connected through SIP trunk which is configured on IPO LAN2

When Agents are transferring the calls from Ameyo App to IPO LAN1 users, few call transfer's are working but out of 10,
5 to 6 calls will drop randomly.

We have logs which says SIP/2.0 503 Service Unavailable from IP Office end.

Can anyone please help to get this sorted.
-----------------------------------------------------------------------------------------------------------------------

[Nov 23 09:42:25] NOTICE[10986]: rtp.c:1174 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.10.10.20
== Using SIP RTP CoS mark 5
Audio is at 10.10.10.58 port 10468
Adding codec 0x40 (slin) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.10.10.20:5060:
INVITE sip:593111191@10.10.10.20:5060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.58:5060;branch=z9hG4bK69fe70b7;rport
Max-Forwards: 70
From: "ameyo" ;tag=as5015d595
To:
Contact:
Call-ID: 763cfc901595981356e33b815ae7895b@10.10.10.20
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2
Date: Thu, 23 Nov 2017 05:42:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 487

v=0
o=root 421836496 421836496 IN IP4 10.10.10.58
s=Asterisk PBX 1.6.2
c=IN IP4 10.10.10.58
t=0 0
m=audio 10468 RTP/AVP 10 3 0 8 112 5 7 18 111 9 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from TCP:10.10.10.20:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 10.10.10.58:5060;branch=z9hG4bK69fe70b7;rport
From: "ameyo" ;tag=as5015d595
To: ;tag=1e4202c13423bab7
Call-ID: 763cfc901595981356e33b815ae7895b@10.10.10.20
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
-- Got SIP response 503 "Service Unavailable" back from 10.10.10.20
Transmitting (NAT) to 10.10.10.20:5060:
ACK sip:593111191@10.10.10.20:5060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.58:5060;branch=z9hG4bK69fe70b7;rport
Max-Forwards: 70
From: "ameyo" ;tag=as5015d595
To: ;tag=1e4202c13423bab7
Contact:
Call-ID: 763cfc901595981356e33b815ae7895b@10.10.10.20
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2
Content-Length: 0


---
> Channel SIP/avaya-00008d79 was never answered.
Really destroying SIP dialog '763cfc901595981356e33b815ae7895b@10.10.10.20' Method: INVITE

------------------------------------------------------------------------------------------------------------------------
IP information :

Avaya SIP - 10.10.10.20

Ameyo IP - 10.10.10.58

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

Thanks.......
 
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This part is clear enough for me:

[Nov 23 09:42:25] NOTICE[10986]: rtp.c:1174 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.10.10.20

BAZINGA!

I'm not insane, my mother had me tested!
 
Upvote 0 Downvote
Everything was fine from last 1 year....
Suddenly the issue started from last week, shared logs are from Ameyo Server, and they are pretty sure that the issue is from IP Office end..
your help will be much appreciated...

Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.10.10.20

(Client is IP Office LAN1 port which IP Address is 10.10.10.20)
 
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Then turn it off, in Avaya it is called "VoIP Silence Suppression", it is off by default.
 
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VoIP Silence Suppression is already off by default, but still the issue is there
 
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If it was working fine for one year and nothing has changed on the Avaya side i ask you : where is the problem to be found?
 
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intrigrant
Exactly, but other team has shared trace logs which indicate SIP Service unavailable from IPO End
 
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Would be easier for us to see the traces of the IPO side.
 
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Sure, I will send it by tomorrow... Thank you guys for all your support.
 
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Did you delete the To and Contact fields?

From: "ameyo" ;tag=as5015d595
To:
Contact:

"Trying is the first step to failure..." - Homer
 
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janni78
From where to delete To and Contact fields? Ameyo Server?
 
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IPOMAN-1986 said:
Exactly, but other team has shared trace logs which indicate SIP Service unavailable from IPO End
Click to expand...

That doesn't proof there is a problem with the Avaya.
They may changed something and now the Avaya disapprove the change and generates an error.

The trace shows no "503 Service Unavailable" message.
Beside that the trace is nearly unreadable due to a overflood of info which is not needed.
You could start with disabling the H.323 messages and a description of a call flow.

Like "external call arriving via PRI trunk to DDI targeted to xxx answerd by Ameyo and then transferred by Ameyo to agent yyy which generated a 503 SIP error".
Then we have something to go on, now it is searching for a needle in a haystack in the dark with your hands bound to your back.
 
Upvote 0 Downvote
In your first post the From, To and Contact field contain no numbers.
It should contain who is calling and what they are calling

INVITE sip:593111191@10.10.10.20:5060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.58:5060;branch=z9hG4bK69fe70b7;rport
Max-Forwards: 70
From: "ameyo" ;tag=as5015d595
To:
Contact:
Call-ID: 763cfc901595981356e33b815ae7895b@10.10.10.20
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2
Date: Thu, 23 Nov 2017 05:42:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 487


"Trying is the first step to failure..." - Homer
 
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intrigrant
I will try to recreate the issue in monitor trace... I will update you soon..
 
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janni78

all call traces are showing without to and contact,
in this case all calls should drop, but here few call transfers are working

I have enabled sysmon trace again, i will try to capture the issue....
 
Upvote 0 Downvote
intrigrant
No....We have forwarded call through shortcode
 
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