[OpenSIPS-Users] Users Digest, Vol 36, Issue 8

Mon Jul 4 15:17:59 CEST 2011

In english please :-)
Le 4 juil. 2011 15:00, <users-request at lists.opensips.org> a écrit :
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> Today's Topics:
>
> 1. 1.7 CRITICAL tm Error (Chris Martineau)
> 2. Re: Users Digest, Vol 36, Issue 6 (Jayesh Nambiar)
> 3. Re: 1.7 CRITICAL tm Error (Vlad Paiu)
> 4. passing auth credentials using opensips B2B modules
> (Jayesh Nambiar)
> 5. Re: SIP Registration Failure - 484 Address Incomplete
> (Bogdan-Andrei Iancu)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 4 Jul 2011 11:51:33 +0100
> From: "Chris Martineau" <chris at ghosttelecom.com>
> Subject: [OpenSIPS-Users] 1.7 CRITICAL tm Error
> To: <users at lists.opensips.org>
> Message-ID:
> <1D10AB188D6CCA46BB4369E3268E36EF2D1B87 at SVR01.ghosttelecom.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi,
>
>
>
> Getting the following error upon connecting a call. Didn't get it with
> the last head version but have just downloaded the 1.7 trunk and I am
> now getting this.
>
>
>
> CRITICAL:tm:run_trans_callbacks: callback type 128, id 1 entered
>
>
>
> Probably something to do with my config but getting no other errors so
> not sure where to look.
>
>
>
> Regards
>
>
>
> Chris
>
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> ------------------------------
>
> Message: 2
> Date: Mon, 4 Jul 2011 16:23:55 +0530
> From: Jayesh Nambiar <jayesh.voip at gmail.com>
> Subject: Re: [OpenSIPS-Users] Users Digest, Vol 36, Issue 6
> To: users at lists.opensips.org
> Message-ID:
> <CALvF6vAu343MMRKJJqhQ9APCAbOHXLgWt=_=KZhoGZ23b4Na3g at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thank you Ovidiu. I had assumed that the Custom-Headers parameters works
> only for the initial INVITE request. It works as expected. I have another
> concern here:
> I dont want the "100 Trying" response going out to the call initiator
> because I know that "407" is will obviously come from the "Proxy
Opensips".
> While using t_relay I could have done that using the flags inside the
> t_relay function. Is it possible to avoid the "100 Trying" somehow when
> relaying the call in B2B mode?
> To be honest, I have a web application which initiates the call and
expects
> a "407" response and then will send the INVITE again with digest
> credentials. When that application gets the "100 Trying" response, the
> application gets confused :)
> Either I will have to tweak the application to just ignore the 100
response
> which is little difficult or else find a way to avoid sending 100 response
> from opensips. Please Help !!
>
> Thanks in advance.
>
> --- Jayesh
>
>
>> Try adding your extra headers to the "cusotm_headers" parameter:
>> http://www.opensips.org/html/docs/modules/devel/b2b_logic.html#id250014
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Sat, Jul 2, 2011 at 5:46 AM, Jayesh Nambiar <jayesh.voip at gmail.com>
>> wrote:
>> > Hi All,
>> > I have a following situtation:
>> > Sip-Endpoint --> Opensips B2B (with REFER scenario) --> Opensips
>> Proxy(Using
>> > Proxy Authorize for Authentication) --> PSTN Gateway
>> > I need the "B2B Opensips" in between so that it can handle the REFER
>> > requests properly. The problem is that the "Proxy Opensips" responds
back
>> as
>> > 407 with Proxy-Authenticate header along with the realm and nonce
value;
>> but
>> > the "B2B Opensips" would only send the 407 response to the Endpoint
>> without
>> > any Proxy-Authenticate header. Thus the call fails. I cannot authorize
>> the
>> > Sip-Endpoint on the "B2B Opensips" because of the limitations of my
>> > architecture.
>> > What I need is, the 407 response gets back to the Sip-Endpoint
>> transparently
>> > and the response to the same also reaches the "Proxy Opensips"
properly,
>> so
>> > that the proxy authorization becomes successful and the call can go
>> through
>> > properly.
>> > Is there something that can be done on the b2b_response route? Any
>> pointers
>> > or directions in achieving this will be very helpful !!
>> > Thanks in advance.
>> > --- Jayesh
>> >
>>
>>
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>
> ------------------------------
>
> Message: 3
> Date: Mon, 04 Jul 2011 14:03:00 +0300
> From: Vlad Paiu <vladpaiu at opensips.org>
> Subject: Re: [OpenSIPS-Users] 1.7 CRITICAL tm Error
> To: users at lists.opensips.org
> Message-ID: <4E119DE4.9090701 at opensips.org>
> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
>
> Hello Chris,
>
> That CRITICAL message is there to help us track down a bug that
> sometimes causes OpenSIPS to crash. We are currently in the testing
> phase, before the release, and we are still working on that bug.
>
> Regards,
>
> --
> Vlad Paiu
> OpenSIPS Developer
>
>
>
> On 07/04/2011 01:51 PM, Chris Martineau wrote:
>>
>> Hi,
>>
>> Getting the following error upon connecting a call. Didn't get it with
>> the last head version but have just downloaded the 1.7 trunk and I am
>> now getting this.
>>
>> CRITICAL:tm:run_trans_callbacks: callback type 128, id 1 entered
>>
>> Probably something to do with my config but getting no other errors so
>> not sure where to look.
>>
>> Regards
>>
>> Chris
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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>
> ------------------------------
>
> Message: 4
> Date: Mon, 4 Jul 2011 16:38:20 +0530
> From: Jayesh Nambiar <jayesh.voip at gmail.com>
> Subject: [OpenSIPS-Users] passing auth credentials using opensips B2B
> modules
> To: users at lists.opensips.org
> Message-ID:
> <CALvF6vDNdTnrWq8ySk6G9ppZCurqxZp5DeFY+x0VkK0wFz5M=g at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thank you Ovidiu. I had assumed that the Custom-Headers parameters works
> only for the initial INVITE request. It works as expected. I have another
> concern here:
> I dont want the "100 Trying" response going out to the call initiator
> because I know that "407" is will obviously come from the "Proxy
Opensips".
> While using t_relay I could have done that using the flags inside the
> t_relay function. Is it possible to avoid the "100 Trying" somehow when
> relaying the call in B2B mode?
> To be honest, I have a web application which initiates the call and
expects
> a "407" response and then will send the INVITE again with digest
> credentials. When that application gets the "100 Trying" response, the
> application gets confused :)
> Either I will have to tweak the application to just ignore the 100
response
> which is little difficult or else find a way to avoid sending 100 response
> from opensips. Please Help !!
>
> Thanks in advance.
>
> --- Jayesh
>
>
>
>> Try adding your extra headers to the "cusotm_headers" parameter:
>> http://www.opensips.org/html/docs/modules/devel/b2b_logic.html#id250014
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Sat, Jul 2, 2011 at 5:46 AM, Jayesh Nambiar <jayesh.voip at gmail.com>
>> wrote:
>> > Hi All,
>> > I have a following situtation:
>> > Sip-Endpoint --> Opensips B2B (with REFER scenario) --> Opensips
>> Proxy(Using
>> > Proxy Authorize for Authentication) --> PSTN Gateway
>> > I need the "B2B Opensips" in between so that it can handle the REFER
>> > requests properly. The problem is that the "Proxy Opensips" responds
back
>> as
>> > 407 with Proxy-Authenticate header along with the realm and nonce
value;
>> but
>> > the "B2B Opensips" would only send the 407 response to the Endpoint
>> without
>> > any Proxy-Authenticate header. Thus the call fails. I cannot authorize
>> the
>> > Sip-Endpoint on the "B2B Opensips" because of the limitations of my
>> > architecture.
>> > What I need is, the 407 response gets back to the Sip-Endpoint
>> transparently
>> > and the response to the same also reaches the "Proxy Opensips"
properly,
>> so
>> > that the proxy authorization becomes successful and the call can go
>> through
>> > properly.
>> > Is there something that can be done on the b2b_response route? Any
>> pointers
>> > or directions in achieving this will be very helpful !!
>> > Thanks in advance.
>> > --- Jayesh
>> >
>>
>>
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>
> ------------------------------
>
> Message: 5
> Date: Mon, 04 Jul 2011 15:59:34 +0300
> From: Bogdan-Andrei Iancu <bogdan at opensips.org>
> Subject: Re: [OpenSIPS-Users] SIP Registration Failure - 484 Address
> Incomplete
> To: OpenSIPS users mailling list <users at lists.opensips.org>
> Message-ID: <4E11B936.10309 at opensips.org>
> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
>
> Hi Mark,
>
> Probably you miss an "exit" after the save(location) block:
>
> if (!save("location"))
> sl_reply_error();
> exit;
> }
> exit;
>
>
> Regards,
> Bogdan
>
> On 07/04/2011 06:23 AM, Mark Holloway wrote:
>> Odd, even though I now get 200 OK from OpenSIPS after registering I
>> still receive 484 Address Incomplete after the 200 OK even though the
>> endpoint is registered.
>>
>>
>> On Jul 3, 2011, at 11:35 AM, Mark Holloway wrote:
>>
>>> I resolved it. Forgot to uncomment one very important part of the
>>> cfg file.
>>>
>>> if (!save("location"))
>>> sl_reply_error();
>>>
>>> exit;
>>> }
>>>
>>>
>>> On Jul 3, 2011, at 11:31 AM, Mark Holloway wrote:
>>>
>>>> I have OpenSIPS 1.6.4 configured as a SIP Registrar with mysql. It
>>>> is configured to challenge SIP registrations with authentication.
>>>> Keeping it simple, I've used opensipsctl to add user 1000 with
>>>> password 1000 and set the IP/SIP Domain of OpenSIPS is 192.168.1.20.
>>>> I am attempting to register using Counterpath Bria 3.1 (Mac). Bria
>>>> sends the initial registration to OpenSIPS and it properly responds
>>>> with a 401 Unauthorized, but after Bria responds to the 401
>>>> challenge OpenSIPS responds with 484 Address Incomplete.
>>>>
>>>> Looking at the docs I noticed OpenSIPS will send 484 if the username
>>>> is not in the Request-Line. For example, OpenSIPS wants
>>>> 1000 at 192.168.1.20 but Bria sends 192.168.1.20. The way Bria
>>>> registers, the username portion (1000 in this case) is only in the
>>>> Contact, To, and From headers. The R-URI line does not include the
>>>> username portion. Has anyone else encountered this?
>>>>
>>>>
>>>> ==
>>>> Bria's response to 401 challenge
>>>> ==
>>>>
>>>> Session Initiation Protocol
>>>> * Request-Line: REGISTER sip:192.168.1.20 SIP/2.0*
>>>> Method: REGISTER
>>>> [Resent Packet: False]
>>>>
>>>>
>>>> Contact:
>>>> <sip:1000 at 192.168.1.116:62018;rinstance=3b31a8f560e5c3bc;transport=udp>
>>>> Contact Binding:
>>>> <sip:1000 at 192.168.1.116:62018;rinstance=3b31a8f560e5c3bc;transport=udp>
>>>> To: "Mark"<sip:1000 at 192.168.1.20>
>>>> SIP Display info: "Mark"
>>>> SIP to address: sip:1000 at 192.168.1.20
>>>> From: "Mark"<sip:1000 at 192.168.1.20>;tag=fbaf644d
>>>> SIP Display info: "Mark"
>>>> SIP from address: sip:1000 at 192.168.1.20
>>>> SIP tag: fbaf644d
>>>>
>>>>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS solutions and "know-how"
>
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>
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>
> End of Users Digest, Vol 36, Issue 8
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