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SIP IPO 500 and Asterisk!

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johnnybrian

johnnybrian

IS-IT--Management
Sep 11, 2007
233
GB
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!
 
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i don't think that is correct. the trace shows that you are sending the number. If you can dial an extension on the Asterisk you are using the same method going via the same sip trunk on the ip office.

I would say there is something wrong with your config on the ip office.

What are your system timers, dial delay and count?
what are the short codes you use to put a call on the asterisk, both for extension and external numbers?
 
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Hi! Thanks for the replys, the problem still persists.

Yes, i have the SIP licenses.

Dial Delay is 5, count is 4.

I use 7N for dialing into this SIP line. 7N dials N on ARS 51.

ARS 51 dials n"@domain.name.com" out of line 7 which is the SIP line.

Please help, the config is available upon request!
 
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One observation:

Place a semicolon (;) at then end of the 7N (if not already).

Asterisk (along with any SIP provider that uses Asterisk) cannot take digits in succession. The entire dialing string must be submitted in one instance.

Hope this helps.
 
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So the code should be: 7N; dials N on ARS 51 ? Or should it be 7N; dials N; on ARS 51?

Should anything be changed in the ARS?

thanks for your help ;)
 
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JB:

Just the 7N;

I was using CallCentric for my SIP provider and couldn't understand why no matter what I dialed I'd get the "Number you have dialed..." error.

The semicolon says: wait until I'm done dialing (by timeout) before you (the IPO) send the string to the SIP provider (or another endpoint). This is not necessary when using SCN, or in my case, I have another IPO tied to mine via H.323 but DO NOT have SCN turned on. The IPO on the other end is capable of receiving and processing each digit individually and in succession.

Drew
 
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You do not get a dialtone after the 7 ?

What happens if you dial the complete number ?


ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!
 
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set dial count to 0

set the dial delay to something more usable in the ars, 2 or 3 seconds. 5 will pss people off, unless it's an old foks home and 5 is quick!

leave short code as 7N, N 51:ARS
In 51ARS set change to N;
 
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i have just tried this on a 4.1 system and I get dial tone. sip also works

check your main short code:

7N
Dial
TN: N
ARS:51


ARS51: (secondary dial tone)

N;
TN: N"@1234.com"
 
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Yessir, thats what i had for starts! It works for the dialtone, but still the problem remains:

I dial: 7; dialtone, the i dial the number, i get silence and after 10 seconds i get BUSY NUMBER in my display and a dialtone!!

:(
 
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sorry my mistake.

back to basics,
you have registration enabled but you don't have a password
you are using lan2.

do you have stun run on lan 2 and an ip route set for lan2.
you are using a dns name for registration, do you have a dns ip address assigned.

if all is correct and set, I would suggest that you turn off registration required and ask your provider to accept your ip address as a trusted source. still fill in the user name and password though.

 
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TheTaker is correct. You need the N; or the IPO will pass each digit as it's dialed onto Line 7 as is depicted in your JPG file. The ; is absolutely necessary.

Take care,
Drew
 
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does anyone have how the ipoffice and asterisk system should be configured?
 
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Are you using digital or IP phones on the IPO, if you use IP sets are you giving them direct media path? you might have to disable that

Joe W.

FHandw.
ACA
ACS
 
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@ Drew and TheTaker.

Configured. Still doesnt work! :/ Same issue; i get dialtone after some time. I tried again this morning to configure a SIP software client with the provider, and it works great! :(

Im really starting to get annoyed with this. Please tell me what information you need, and i will provide.

 
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all i need is the config for the ip office and for the asterisk box. all i want is extension to extension dialing
jdovejr"at"rticonnection"dot"com
please help

 
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@ westi: Direct media path is disabled. Adn when enabled, all it does is that i cant hear the other end. :(
 
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