AVAYA IP Office - SIP/2.0 503 Service Unavailable

[IPOMAN-1986]

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.
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[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

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IP information :

Avaya SIP - 10.10.10.20

Ameyo IP - 10.10.10.58

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Thanks.......

 

[mattKnight]

how many times do I hear this?

"It worked. Now it doesn't. Nothing was changed."

Let me give you a clue. Something has been changed. Figure out who changed what anda you'll know what to change to fix it.



Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.

 

[IPGuru]

"It worked. Now it doesn't. Nothing was changed."

Let me give you a clue. Something has been changed. Figure out who changed what anda you'll know what to change to fix it.

& just for clarity -

replacing a data switch is a change - even if you think it ha been programmed identically.​

Replacing a patch cable is a change - even if you think it has been correctly rematched check it again​

Adding a new server/ device is a change​

all the above should be obvious but I have had IT departments do all of the above & consider it to be "Nothing has changed"



Do things on the cheap & it will cost you dear