SIP IPO 500 and Asterisk!

[johnnybrian]

Hi Guys!

Have a problem on my IPO 500 talking with a SIP line on an asterisk system. I can call locally to to a phone directly connected to the asterisk system, but not to external numbers. I get the following error all the time:
"SipDebugInfo: SipTrunks: Cannot free Txn Key 2015" and the phone just gives dialtone after some time, and displays "BUSY".

The avaya is directly on the internet with a public IP.

Here is my log:

81110183mS SIP Trunk: 7:Tx
ACK sip:8104576522002@Voip-gk.lvivfarlep.net SIP/2.0
Via: SIP/2.0/UDP 195.95.147.2:5060;rport;branch=z9hG4bKd4616c83d2a66d9df07cadb2260d0b97
From: 2448348 <sip:2448348@voip-gk.lvivfarlep.net>;tag=d9401813c926d35a
To: <sip:8104576522002@Voip-gk.lvivfarlep.net>;tag=9fc1003e600e0c417d959233a7e35b42-afbf
Call-ID: 58dffac78a08484c1a2bff918e1ed588@195.95.147.2
CSeq: 1643336771 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

81110188mS SipDebugInfo: *********************************************************
81110183mS SIP Tx: UDP 195.95.147.2:5060 -> 62.221.56.2:5060

ACK sip:8104576522002@Voip-gk.lvivfarlep.net SIP/2.0
Via: SIP/2.0/UDP 195.95.147.2:5060;rport;branch=z9hG4bKd4616c83d2a66d9df07cadb2260d0b97
From: 2448348 <sip:2448348@voip-gk.lvivfarlep.net>;tag=d9401813c926d35a
To: <sip:8104576522002@Voip-gk.lvivfarlep.net>;tag=9fc1003e600e0c417d959233a7e35b42-afbf
Call-ID: 58dffac78a08484c1a2bff918e1ed588@195.95.147.2
CSeq: 1643336771 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

81110188mS SipDebugInfo: State Transtion form Old State 17 to New state 40
81110188mS SipDebugInfo: *********************************************************
81110188mS SipDebugInfo: SIP Line (7): Cannot free Txn Key 2015
81115183mS SipDebugInfo: Timer 4 callback
81115183mS SipDebugInfo: SIPDialog destructor ... f58e9dd4
81115183mS SipDebugInfo: Completed ... removing Dialog of CallId: 58dffac78a08484c1a2bff918e1ed588@195.95.147.2 and State: 40
81115183mS SipDebugInfo: SIPDialog - Free SDPBody....
81115184mS SipDebugInfo: ~SipTrunkEndpoint
81140488mS PRN: DCP message rejected because the terminal is off-hook. port 8024 type 5


The SIP provder guy tells me that the problem is that my first package comes without the number, so the gateway doesnt know the number. How can i correct this?

Can you gurus help me?
Asterisk log can be provided!

 

[IPOfficeIreland]

dialtone after some time...

set your timers in System - Telephony

Dial Delay Timer -2 or 3 seconds (2-3 seconds after last digit dialled before completing call)
Dial Delay Count - 0 (number of digits to wait to be entered before the system looks for a match, if you have this set to 3 or 4 by default it will use the timer above before completing the call.)

With 0 in above and short code 9N you will get dial tone immediately.

 

[intrigrant]

After this line :
81110188mS SipDebugInfo: SIP Line (7): Cannot free Txn Key 2015
The Asterisk should respond with acknoledge messages and further call setup.
As this does not happen then the number you send to the Asterisk is not in it's "known number" list and now the Asterisk is waiting and it will do nothing else so after 5 seconds IPO terminates the call.

NIHIL NOVI SUB SOLE

 

[johnnybrian]

Intrigant; So the problem actually lies in the fact that the Asterisk does not respond with ACK on the request from the IPO?

But why does it work on a normal software sip client then?

 

[intrigrant]

It is always possible that there is a bug in IPO causing this to happen.
I have seen strange things with SIP like a SIP line which do work at the office and do not work at customer site, the IP 500 at customer site reboots when i try to make a call and in our office it is working as a charm.
You better contact Avaya and raise a case for this.

 

[johnnybrian]

Okay, now i found out this problem! Tested it with other SIP providers as well, and this might be pretty relevant info to some of you SIP users out there who are having trouble: The Sip calling WILL NOT WORK PROPERLY if you put in N"@ip_address_of_provider" in your ARS. It should be N"@fqdn_of_provider"

This actually makes no sense to me, as it shouldnt matter whether putting in the ip or the domain name. But apparently, it does. This solved all my problems anyway!

Cheers to you all!

 

[bareedon]

Hi

using HTC Cruise, we had to flash the original WM6, and replace it with WM6.1 version that comes with VoiP dialer. the device started connection with the SIP server, and make and recieve calls with no problem. the next step was to install the avaya one-x mobile client due to the need for dual voip/GSM functionality, and there started the problem.
the device can not connect with the sip server, and only works on GSM mode.
we have followed some suggesgtions to edit the reg files and with no luck.
What we are suspecting is the HTC built in dialer, which seems to be still enabled.
now the question for those who suceeded getting this thing running; do we have to remove the built in dialer, and if yes then how?

We certainly need your help and suggestions to solve this problem.

Thank you, in advance and have a good one.

 

[johnnybrian]

Hello bareedon!

I would suggest you start a new thread about this, as this thread only has to do with a IPO 500 and nothing mobile.

Cheers

 

[bareedon]

Dear cheers,

Thank you for your reply,
Which is the appropriate section or forum you recommend me to but my thread on?

Thankx