2 Calls from 1 SIP account strange problem 1

[RyanEOD]

Greetings all, I'm having a strange problem that is baffling me. I have a 3rd Party SIP account, if I call outbound to a number, let it ring on the distant end and try calling again to a different number (or the same, doesn't matter) the second call gets a 503 Service Unavailable back. Now, if that first call was ANSWERED before I try the second call, the second one works fine. Also, I have tested this on 4 different IPO switches, it works fine on 2 and has the same problem on 2 others. So, I think there is a setting but for the life I me I can't find it. Any suggestions? I've checked the extension and the user and stared and compared settings and everything seems the same on the different IPO switches.

In doing a trace, here is the error from the IPO:


16:58:05 1631205216mS SIP Tx: UDP 10.50.1.35:5060 -> 10.50.4.89:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.50.4.89:5060;branch=z9hG4bK3c9092b6509c6118;rport
From: "88051" <sip:88051@XX.XX.X.XX:5060>;tag=7c365c0926
Call-ID: d8ad371eccb11754-49411409
CSeq: 9933 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
Supported: timer,100rel
Server: IP Office 9.1.7.0 build 163
Reason: Q.850;cause=47;text="Resource unavailable, unspecified"
To: <sip:XXXXXXXXXXX@XX.XX.X.XX>;tag=5fd4c3a0c6065e38
Content-Length: 0


Any thoughts?


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[amriddle01]

That's the way it should work, the second call works once the first is answered as it could then be initiating a transfer, you can't transfer an alerting call, even on IP/digital sets.

SIP endpoints work like analogue devices on IPO (not multiline) so treat them as such

 

[RyanEOD]

I have two IPO switches that DON'T work that way, so it makes me think there is a setting for one way or the other. These are 3rd Party SIP endpoints, just to clarify. I have a lab system that I can call out from the same line twice without the distant end answering either call and it is treated as two separate calls.

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[amriddle01]

Oddly, the Call waiting setting of the user effects what SIP endpoints can do, try messing with that, and the Re-invite setting on the extn part....

 

[RyanEOD]

I thought those settings might play in as well . . I've tested it with them on and off and every combination, and on my working lab system I can't break it. On the Non-working system changing them can't get it working.

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[amriddle01]

That's the issue with SIP endpoints, different results every time.
Some devices can transfer some can't, some can supervised transfer, some only blind transfer, it's always a game of suck it and see

 

[RyanEOD]

Yeah, I was afraid of that. This is for a ACD integration, and I'm trying to figure out the difference for the customer.

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[janni78]

Do a monitor trace when it works and when it doesn't and compare.

"Trying is the first step to failure..." - Homer

 

[RyanEOD]

@Janni78 - I have traces from both and just can't seem to figure out the problem spot myself. Any suggestions on the best places to look or what to look for?

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[intrigrant]

Just default all and start from scratch, maybe you missed something when programming the user/extension/phone.
One other thing to keep in mind that with each major release AVAYA tends to change/add features slightly which are poorly documented(if at all).
So it may work on 9.0 but it may fail on 9.0 and work on 10.0 again.

 

[janni78]

If you do a default trace it should say why it can't dial out.

Don't know exactly what you 3rd party application does but if it's doing several outgoing calls maybe it should use a SIP trunk instead, or several SIP users.

"Trying is the first step to failure..." - Homer