On Tue, Jan 26, 2010 at 9:16 PM, Tony Graziano
<[email protected]> wrote:
> I don't know what "206 no peers available" means from sipxbridge.

Well, there are two problems at hand here. The 481 error from
sipxbridge on the re-INVITE -- I believe this to be a sipxbridge bug
that I have fixed already in 4.1. You can give that a try.
However, I am curious about why verizon  issending a re-INVITE as soon
as the call is. Perhaps it does not like the fact that you are using
10.87.20.5 address in your call setup.



Regards,


Ranga.



>
> Perhaps someone else can help?
> ============================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.984.8431
>
> Email: [email protected]
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> ----- Original Message -----
> From: [email protected] <[email protected]>
> To: Tony Graziano <[email protected]>;
> [email protected] <[email protected]>
> Sent: Tue Jan 26 21:12:55 2010
> Subject: Re: Can't forward from external number to external number
>
> Just 1. 95% of my users have 1 line assigned to 1 phone, and everyone is
> definitely having the issue.
>
> On 1/26/2010 8:01 PM, Tony Graziano wrote:
>> How many phones is your line registered on?
>>
>> On Tue, Jan 26, 2010 at 8:40 PM, [email protected]
>> <mailto:[email protected]> <[email protected]
>> <mailto:[email protected]>> wrote:
>>
>>     Ok. Lets try this again. I think I have some better data now
>>     thanks to some help from Tony. I waited until nobody was on my
>>     system, moved all the logs out, immediately made a test call, and
>>     then immediately moved those calls to a temp directory. I ran
>>     merge-logs in that temp directory, opened it in sipviewer, and
>>     attached the merged.xml that was created.
>>     In this test, I was calling from 6155008073 (my cell phone) to
>>     6159253043 (my Sipx/Desk phone) which was set in the Sipx GUI to
>>     forward to 6155914780 (my home phone). My home phone rung, and the
>>     call was dropped as soon as I answered it. I didn't try to mask
>>     any of my phone numbers in the logs this time.
>>
>>     I have to guess the 2 parts from merged.xml below indicate a
>>     problem. Googling '481 Peer dialog is null' doesn't get too many hits.
>>     Sorry for not sending correct or helpful information earlier.
>>     Hopefully what I'm sending now is a little more helpful. I didn't
>>     think I needed to send a screenshot from sipviewer, but I will be
>>     glad to if that would help.
>>
>>     Thank you all for your help,
>>     Matthew
>>
>>     Time: 2010-01-27T01:16:12.311000Z
>>     Frame: 29 sipxbridge.xml:378
>>
>>     Source: nshpbx1.sipx.voip-sipXbridge
>>     Dest: 10.87.20.5:5060 <http://10.87.20.5:5060>
>>
>>     SIP/2.0 481 Peer dialog is null
>>     Via: SIP/2.0/UDP
>>     10.87.20.5;branch=z9hG4bK-sipXecs-19e8a5983eb41c5558b4327c16988037cc75
>>     Via: SIP/2.0/UDP
>>     10.87.20.5:5090;branch=z9hG4bKeca98329a090bdd96d9d633f294be82f333631
>>     CSeq: 917280447 INVITE
>>     Call-ID: [email protected]
>>     From: "WIRELESS CALLER" <sip:[email protected]
>>
>> <mailto:sip%[email protected]>;user=phone>;tag=2099232641-1264554945912-
>>     To: "DSI HOLDING COMPANY 251 DSI Corp"
>>     <sip:[email protected]>;tag=9df1f5b6
>>
>>     Server: sipXecs/4.0.4 sipXecs/sipxbridge (Linux)
>>     Contact: <sip:[email protected]:5090
>>     <http://[email protected]:5090>>
>>     Supported: replaces,100rel
>>     Content-Length: 0
>>
>>     Time: 2010-01-27T01:16:12.321000Z
>>     Frame: 33 sipxbridge.xml:383
>>
>>     Source: nshpbx1.sipx.voip-sipXbridge
>>     Dest: 172.30.209.62:5070 <http://172.30.209.62:5070>
>>
>>     SIP/2.0 481 Call leg/Transaction does not exist
>>     Via: SIP/2.0/UDP
>>     172.30.209.62:5070;branch=z9hG4bK548gtc308ghhhoccp4g1cbm1bd9v2.1
>>     From: "WIRELESS CALLER" <sip:[email protected]
>>
>> <mailto:sip%[email protected]>;user=phone>;tag=2099232641-1264554945912-
>>     To: "DSI HOLDING COMPANY 251 DSI Corp" <sip:[email protected]
>>     <mailto:sip%[email protected]>>;tag=5102113
>>     Call-ID: [email protected]
>>     <mailto:[email protected]>
>>     CSeq: 917280446 INVITE
>>
>>     Server: sipXecs/4.0.4 sipXecs/sipxbridge (Linux)
>>     Supported: replaces
>>     Contact: <sip:[email protected]:5080;transport=udp>
>>     Reason: ~~id~bridge;cause=213;text="Relayed Error Response"
>>     Content-Length: 0
>>
>>
>>     On 1/26/2010 3:26 PM, [email protected]
>>     <mailto:[email protected]> wrote:
>>
>>         This is similar to something I posted a week or so ago about
>>         trying to forward at the handset level, but I'm assuming it is
>>         a a completely different issue.
>>         If a user sets a forward through the web gui and specifies an
>>         external number, they have an issue if the inbound call to be
>>         forwarded is also from an external number. The call rings on
>>         the destination phone, but is disconnected with a click as
>>         soon as it is answered. If the call is forwarded to an
>>         internal extension, everything is fine. If the call is
>>         forwarded to an external number and the caller is on an
>>         internal phone, everything is fine. This sounds like a
>>         permission issue, but if so, I don't understand why it makes
>>         it as far as calling the destination phone , but then
>>         disconnects when it is answered.
>>
>>         The text below is from sipxbridge.log. I didn't want to post
>>         the phone numbers in question for a automated routine of some
>>         sort to grab at least, so I changed the 615 area code to 222
>>         in the logs. All area codes involved in this log are 615. In
>>         this case, the polycom phone is at 4670142. I set it to
>>         forward to 5008073. The inbound call came from 2439019.
>>         10.87.20.5 is my sipx server. pcelbcn0001.dsi.globalipcom.com
>>         <http://pcelbcn0001.dsi.globalipcom.com> [172.30.209.62] is my
>>         Verizon gateway. I would be more than happy to provide any
>>         more information, but I'm not sure where I should be looking.
>>
>>         Sipx 4.0.4, sixbridge, Verizon VOIP, No firewall (not needed,
>>         private connection), Polycom 450s and 550s - bootrom 4.2.1,
>>         firmware 3.1.3C split.
>>
>>         Thanks as always,
>>         Matthew
>>
>>
>>
>>
>>
>> --
>> ======================
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> Fax: 434.984.8431
>>
>> Email: [email protected] <mailto:[email protected]>
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>> Telephone: 434.984.8426
>> Fax: 434.984.8427
>>
>> Helpdesk Contract Customers:
>> http://www.myitdepartment.net/gethelp/
>>
>> Why do mathematicians always confuse Halloween and Christmas?
>> Because 31 Oct = 25 Dec.
>>
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>



-- 
M. Ranganathan
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