Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE

[Uwe Kastens]

Hello Bogdan,

Now we changed the behaviour of the UAC. One of them will send a BYE and
this is relayed to the PSTN GW which drops the call, since opensips will
not handle the BYE locally. So loose_route is done and the BYE is
relayed to the PSTN GW.

The following is happening:

1) INVITE from PSTN GW
2) parallel forking to ast1 and ast2 (branches z9hG4bK51f6.9afa91c3.1
and z9hG4bK51f6.9afa91c3.0)
3) ast1 sends an 200 OK (branch z9hG4bK51f6.9afa91c3.0)
4) opensips sends an cancel to ast2 (branch z9hG4bK51f6.9afa91c3.1)
5) opensips receives the 200 OK from ast1 and sends an ACK (branch is
changing here to z9hG4bK51f6.9afa91c3.3)
6) opensips receives 200 OK from ast2 from the INVITE (branch
z9hG4bK51f6.9afa91c3.1)and sends an ACK (branch is changing to
z9hG4bK51f6.9afa91c3.3)
7) opensips reives 200 OK from ast2 for the cancel request ( branch
z9hG4bK51f6.9afa91c3.1)
8) opensips receives BYE from ast2 with branch z9hG4bK40d1af5d
9) opensips is doing loose_route and sends the BYE to the PSTN GW


The only thing I could see on the logs is:

 WARNING:dialog:dlg_onroute: tight matching failed for BYE with
callid='393105a419950c1f265f298914662...@10.20.30.100'/46,
ftag='as63949c6e'/10, ttag='as0d1597ca'/10 and direction=0
Oct 21 09:09:15 asne02 /usr/sbin/opensips[15615]:
WARNING:dialog:dlg_onroute: dialog identification elements are
callid='393105a419950c1f265f298914662...@10.20.30.100'/46, caller
tag='as0d1597ca'/10, callee tag='as79debd51'/10

Why is the opensips not handling the BYE locally and only closing one
branch?

BR

UWe


Bogdan-Andrei Iancu schrieb:
> Hi Uwe,
> 
> Uwe Kastens wrote:
>> Hi Bogdan,
>>
>>   
>>> So actually both legs do send 200 OK (but one faster than the 
>>> other)..so there is kind on race between the 200 OK from the slow 
>>> branch and the CANCEL from OpenSIPS...is this the case?
>>> 
>> Exactly
>>
>>   
>>> If so, the UAS will simply reply with negative reply to CANCEL (decline 
>>> it) and opensips (for INVITE transaction) will not close the second 
>>> branch as there is a 200 OK (and not a 487) received RFC3261 says 
>>> that a proxy must send all 200 OK (for a call), even if more than one, 
>>> to the UAC - the UAC is the one who will decide what branch to keep and 
>>> it will fire a BYE for the other branch.
>>>
>>> 
>> Could this explan, why only the 2nd Node will get the BYE, if the call
>> is released "behind" the opensips?
>>   
> yes, because the caller will hung up only one of the callee branch, so 
> the BYE will go to only one of them. The other branch will remain up and 
> will be the ongoing call.
> 
> Regards,
> Bogdan
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


-- 

kiste lat: 54.322684, lon: 10.13586

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] [Presence] Empty NOTIFY received after re-subscription

[Pascal Maugeri]

Hi

After re-subscribing to a user presence, we receive immediately a NOTIFY
with no content:

NOTIFY sip:us...@xx.xx.xx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6667;branch=z9hG4bK34ab.7700bc14.0
To: >;tag=14257163
From: 
>;tag=9886f112d485ff608a79084eaa27a247-48fb
CSeq: 5 NOTIFY
Call-ID: 87cbad243e9c1b48c8333cbbb99cb...@xx.xx.xx.xx
Route: 
Content-Length: 0
User-Agent: OpenSIPS (1.5.2-notls (x86_64/linux))
Max-Forwards: 70
Event: presence
Contact: 
Subscription-State: active;expires=100

 and then few seconds later I receive a NOTIFY with the proper presence
document.

Which are the possible reasons why the presence server sends an "empty"
NOTIFY ?

Cheers
Pascal
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] RLS Services: subscription expanded to 0 contacts.

[Sanjeev BA]

Hi,

I am facing the issue posted previously.

 

It is very clear what happens. Your rls-services document looks like this:

  http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
;pres-list=Default">

http://xcap.net1.test/xcap-root/resource-lists/users/sip%3aal
ice%40net1.test/index/~~/resource-lists/list...@name=%22default%22%5d



  presence

  
]
 
It has a reference to a resouce-list document where the list is actually 
defined. Unfortunately opensips does not have support for this. It only 
works with the list being defined inside the rls-services document. This 
is on the list of improvements and will be included until the next major 
release.
 

I am using OpenSIPS 1.5. Is it addressed in the release or do I need to
upgrade to 1.6?

 

Regards,

Sanjeev

 

 

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Undelivered Mail Returned to Sender

[JFRKRVFAXDHB]

This is the mail system at host posta.websolutions.it.

I'm sorry to have to inform you that your message could not
be delivered to one or more recipients. It's attached below.

For further assistance, please send mail to 

If you do so, please include this problem report. You can
delete your own text from the attached returned message.

   The mail system

: host merak.websolutions.it[192.168.0.12] said: 554
5.7.1 Message cannot be accepted, spam rejection (in reply to end of DATA
command)
Reporting-MTA: dns; posta.websolutions.it
X-Postfix-Queue-ID: A75EA14903D4
X-Postfix-Sender: rfc822; re.AC6KTKXW62X41@spammotel.com
Arrival-Date: Wed, 21 Oct 2009 09:35:13 +0200 (CEST)

Final-Recipient: rfc822; JFRKRVFAXDHB@spammotel.com
Original-Recipient: rfc822;JFRKRVFAXDHB@spammotel.com
Action: failed
Status: 5.7.1
Remote-MTA: dns; merak.websolutions.it
Diagnostic-Code: smtp; 554 5.7.1 Message cannot be accepted, spam rejection
--- Begin Message ---


Send Users mailing list submissions to
users@lists.opensips.org

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
or, via email, send a message with subject or body 'help' to
users-requ...@lists.opensips.org

You can reach the person managing the list at
users-ow...@lists.opensips.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of Users digest..."


Today's Topics:

   1. Re: parallel forking and CANCEL/BYE (Bogdan-Andrei Iancu)
   2. Re: NAT fixup question (Jeff Kronlage)
   3. Re: use case for settlement free peering (A G)
   4. Re: parallel forking and CANCEL/BYE (Uwe Kastens)
   5. Re: parallel forking and CANCEL/BYE (Bogdan-Andrei Iancu)
   6. Re: NAT fixup question (Bogdan-Andrei Iancu)


--

Message: 1
Date: Tue, 20 Oct 2009 19:33:13 +0300
From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
To: OpenSIPS users mailling list 
Message-ID: <4adde649.70...@voice-system.ro>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hi Uwe,

So actually both legs do send 200 OK (but one faster than the 
other)..so there is kind on race between the 200 OK from the slow 
branch and the CANCEL from OpenSIPS...is this the case?

If so, the UAS will simply reply with negative reply to CANCEL (decline 
it) and opensips (for INVITE transaction) will not close the second 
branch as there is a 200 OK (and not a 487) received RFC3261 says 
that a proxy must send all 200 OK (for a call), even if more than one, 
to the UAC - the UAC is the one who will decide what branch to keep and 
it will fire a BYE for the other branch.

Regards,
Bogdan

Uwe Kastens wrote:
> Hello again,
>
> I was wondering if there might be a bug with the correct handling of
> Cancel in case of receiving and final answer.
>
> I will fork one Call to 2 nodes. One node answers a little faster than
> the other and will get the call. Opensips will send a CANCEL for the
> other node which is sending a SIP/2.0 200 OK before receiving the
> CANCEL. So this node is not answering with a 487 but with a 200/OK.
>
> Opensips seems to drop the call leg and so the BYE from that node could
> not be handled.
>
> Is this behaviour RFC conform?
>
> I will attach one ngrep and one opensips logfile
>
> BR
>
> Uwe
>
>
>
>
> Uwe Kastens schrieb:
>   
>> Hi,
>>
>> I am using opensips to fork calls to UAs which are registrered from 
>> different IPs/Ports.
>>
>> If one UA accepts the INVITE the other UAs will get a CANCEL.
>>
>> Now I have one subscriber with 2 asterisk server which asked me to send 
>> a BYE after the CANCEL. Otherwise he wants me to send an BYE which could 
>> not be processed correctly on the opensips.
>>
>> I am pretty sure, that this kind of handling would not be RFC conform 
>> and so its not possible to handle this inside the routing script. Or did 
>> I missed something?
>>
>> BR
>>
>> Uwe
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>
>
>   
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   




--

Message: 2
Date: Tue, 20 Oct 2009 11:56:40 -0600
From: "Jeff Kronlage" 
Subject: Re: [OpenSIPS-Users] NAT fixup question
To: "OpenSIPS users mailling list" 
Message-ID:

Content-Type: text/plain;   charset="us-ascii"

Bogdan,

The pertinent part of my config file contains this snippet:

$var(ruriuser) = $rU;
if (has_totag()) {
xlog("DEBUG1 ru is $ru, du is $du");
if (loose_route()) {
xlog("DEBUG2 ru is $ru, du is $du");
$rU = $var(ruriuse

[OpenSIPS-Users] tel-uri

[Airton Kuada]

Hi all.Anybody know what is the schema tel-uri?  OpenSIPS works with the schema tel-uri?Thanks.Airton Kuada




___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RLS Services: subscription expanded to 0 contacts.

[Anca Vamanu]

Hi Sanjeev,

Unfortunately this feature is not implemented in 1.6 either.

Regards,
Anca

Sanjeev BA wrote:
>
> Hi,
>
> I am facing the issue posted previously.
>
>  
>
> It is very clear what happens. Your rls-services document looks like this:
>  xmlns="urn:ietf:params:xml:ns:rls-services">
>   http://lists.opensips.org/cgi-bin/mailman/listinfo/users>;pres-list=Default">
> 
> http://xcap.net1.test/xcap-root/resource-lists/users/sip%3aalice%40net1.test/index/~~/resource-lists/list...@name=%22default%22%5d  
> >
> 
>   presence
> 
>   
> ]
>  
> It has a reference to a resouce-list document where the list is actually 
> defined. Unfortunately opensips does not have support for this. It only 
> works with the list being defined inside the rls-services document. This 
> is on the list of improvements and will be included until the next major 
> release.
>  
>
> I am using OpenSIPS 1.5. Is it addressed in the release or do I need 
> to upgrade to 1.6?
>
>  
>
> Regards,
>
> Sanjeev
>
>  
>
>  
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RLS Services: subscription expanded to 0 contacts.

[Adrian Georgescu]

There are two major issues related to the implementing of such feature:

1. You expose your server to information provisioned in a remote system
2. OpenSIPS child may block waiting for the response from the remote  
XCAP server


So though this feature is well defined in the standards, it can  
introduce a lot of problems for the operator that implements it.


--
Adrian





On Oct 21, 2009, at 10:55 AM, Sanjeev BA wrote:


Hi,
I am facing the issue posted previously.

It is very clear what happens. Your rls-services document looks like  
this:


  

http://xcap.net1.test/xcap-root/resource-lists/users/sip%3aalice%40net1.test/index/ 
~~/resource-lists/list...@name=%22default%22%5d


  presence

  
]

It has a reference to a resouce-list document where the list is  
actually
defined. Unfortunately opensips does not have support for this. It  
only
works with the list being defined inside the rls-services document.  
This
is on the list of improvements and will be included until the next  
major

release.

I am using OpenSIPS 1.5. Is it addressed in the release or do I need  
to upgrade to 1.6?


Regards,
Sanjeev


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] append_branches in registrar module of 1.6

[Jayesh Nambiar]

Hello,I dont see the parameter append_branches in the registrar module of
1.6 version. I was using it so that Openser does not route calls to all the
AORs. My requirement was that the last registered client should get the
call.
So i used append_branches as 0 in the registrar module and desc_time_order
as 1 in the usrloc module which fulfilled my requirement properly.
How do i go ahead with 1.6 with similar requirement??

Any help or pointers will be greatly appreciated.

Thanks,

--- Jayesh
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [OpenSIPS-News] How modify Contact header

[Brett Nemeroff]

I'm not sure if it's a real good idea to rerwrite the contact header.
That may break loose routing? Do you have a good reason to do this?


On Tue, Oct 20, 2009 at 8:22 PM, Alex Balashov
 wrote:
> Try escaping the : as well.  They have special meaning in regex.
>
> Ariadne Ramos wrote:
>
>> Hello
>>
>>
>>
>> I’m trying to modify the contact header in onreply route
>>

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] send bye in case of parallel forked call

[Uwe Kastens]

Hi,

I have the following requirement:

If a from tm generated cancel is answered with a 200 OK I want to send a
BYE to the UAC.

Is this possible?

BR

Uwe

-- 

kiste lat: 54.322684, lon: 10.13586

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips 1.5.x nathelper/rtpproxy configuration

[Manivasagam Sivaraman]

Thanks Khan. I tried this before and it did not work, but your documentation
is the most comprehensive one, great job. May be i need some more tune up
for my setup. Basically I can make a call from UA behind NAT to PSTN through
opesips/rtpproxy outbound. I have installed opensips and rtpproxy on the
same Public IP address. I have a confusion over, whether I should run
rtpproxy in bridge mode with dual interface to be able to make rtp flow
between 2 UAs behind  2 different NATs ? Or is the single interface setup
sufficient ?

The first INVITE from UA1 to UA2 ( SDP c= IP address is modified correctly
as rtpproxy's IP address and sent to UA2)
The 200 OK response SDP from UA2 , reaches opensips .. and fwd as is, with
UA2's private IP address in SDP and UA1 cannot sent media. I made sure the
[onreply route1] method is called in script and the force_rtp_proxy() is
invoked by printing xlog. But rtpproxy is not modifying the 200 OK SDP. This
is my exact problem. This indicates that I might have to run rtpproxy in
bridge mode. Also the new opensips documentation I see new functions
rtpproxy_offerSDP and rtpproxy_answer() functions. But I need no example
using those. Every one is still using the deprecated force_rtp_proxy(),
which I think is ok as long as it works.

Thanks in advance. I'm in Chicago too.

On Tue, Oct 20, 2009 at 9:30 PM, Khan  wrote:

> Mani,
>
> There is a complete working configuration posted at the following blog
> link:
> http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html
>
> You might need to tuneup a little bit based on your needs but last time i
> have tried it and it was functional.
>
>
> --
> Khan
>
>
> VoIP Rookie
> Every beginning has an end regardless we believe it or not...
>
>   On Tue, Oct 20, 2009 at 8:20 PM, Manivasagam Sivaraman <
> smvasagam2...@gmail.com> wrote:
>
>>   Dear Pros
>>
>> The nathelper documentation is good as shown below
>> http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id228280
>>
>> Seems like from 1.5.x many nathelper functions are deprecated and new
>> functions are introduced like rtpproxy_offer and rtpproxy_answer.
>>
>> Actually NAT traversal is still a nightmare for many and beginers it would
>> be better if some professional post a full working opensips.cfg
>> nathelper/rtpproxy configuration script for the latest 1.5.3. THe above
>> documents shows bits and pieces, but a full working config example will
>> really help. Could any one please post a working example of opensips.cfg
>> nathelper configuration please. I searched and foind only old configuration
>> that does not work well.
>>
>> Thanks in Advance. I really appreaciate your understanding.
>> Mani
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] One Way Audio

[Ross Beer]

I have a server located on the internet running opensips and asterisk. When 
registering directly to asterisk I can perform echo tests and make calls.

 

If I register to Opensips and use the load_balance there is one way audio. I 
can hear sounds coming from the asterisk server but sound from the soft phone 
does not reach asterisk. I can confirm this when looking at a rtp debug on 
asterisk.

 

I can see that traffic is passing from the soft phone when performing a wire 
shark trace to the server and it also shows that some RTP packet are being 
passed out and back into my local address. This does not happen if I register 
directly to asterisk.

 

Any advice you can offer would be appreciated.

 

Opensips shouldn't effect the RTP if it only load balances?

 

Thanks,


Ross
  
_
Did you know you can get Messenger on your mobile?
http://clk.atdmt.com/UKM/go/174426567/direct/01/___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [OpenSIPS-News] How modify Contact header

[Bogdan-Andrei Iancu]

Ariadne,

do not abuse the "news" mailing list - that list only for announcing and 
discussion news (as the name says) and not for asking technical 
questions - please use the appropriate list (users) for your purpose.

Bogdan

Ariadne Ramos wrote:
>
> Hello
>
> I’m trying to modify the contact header in onreply route
>
> I’m using
>
> if(subst('/^Contact:(.*)@proxy.yy.com/Contact:\...@proxy.xx.com/')) {
>
> xlog("L_INFO", "SUBST OK $ct \n");
>
> }
>
> And the contact still being the same
>
> Logs
>
> [15224]: SUBST OK 
>
> Can you help me to understand what I’m missing?
>
> Thanks in advanced
>
> Ariadne
>
> P BE CARBON CONSCIOUS. PLEASE CONSIDER OUR ENVIRONMENT BEFORE PRINTING 
> THIS E-MAIL
>
> 
>
> ___
> News mailing list
> n...@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/news
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] One Way Audio

[Ross Beer]

NHi Duane,

 

There are is a firewall on the server end however all ports are open, no NAT at 
the server end however there is NATing on the end of the soft phone. Though 
when registering with asterisk directly there is no issue.

 

Regards,

 

Ross
 


Date: Wed, 21 Oct 2009 15:23:04 +
Subject: Re: [OpenSIPS-Users] One Way Audio
From: duane.lar...@gmail.com
To: ross_b...@hotmail.com

Are there any firewalls or NATing involved? 

On Oct 21, 2009 10:13am, Ross Beer  wrote: 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> I have a server located on the internet running opensips and asterisk. When 
> registering directly to asterisk I can perform echo tests and make calls. 
> 
> 
>   
> 
> 
> If I register to Opensips and use the load_balance there is one way audio. I 
> can hear sounds coming from the asterisk server but sound from the soft phone 
> does not reach asterisk. I can confirm this when looking at a rtp debug on 
> asterisk. 
> 
> 
>   
> 
> 
> I can see that traffic is passing from the soft phone when performing a wire 
> shark trace to the server and it also shows that some RTP packet are being 
> passed out and back into my local address. This does not happen if I register 
> directly to asterisk. 
> 
> 
>   
> 
> 
> Any advice you can offer would be appreciated. 
> 
> 
>   
> 
> 
> Opensips shouldn't effect the RTP if it only load balances? 
> 
> 
>   
> 
> 
> Thanks, 
> 
> 
> 
> Ross 
> 
> Did you know you can get Messenger on your mobile? Learn more. 
> 
> 
> 
_
Use Windows Live Messenger for free on selected mobiles
http://clk.atdmt.com/UKM/go/174426567/direct/01/___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] One Way Audio

[duane . larson]

So in the wireshark trace you see RTP traffic coming from the Asterisk  
servers IP address, but what about the traffic coming from the softphone?  
What IP address is that going towards?


On Oct 21, 2009 10:35am, Ross Beer  wrote:











NHi Duane,






There are is a firewall on the server end however all ports are open, no  
NAT at the server end however there is NATing on the end of the soft  
phone. Though when registering with asterisk directly there is no issue.







Regards,







Ross







Date: Wed, 21 Oct 2009 15:23:04 +
Subject: Re: [OpenSIPS-Users] One Way Audio
From: duane.lar...@gmail.com
To: ross_b...@hotmail.com



Are there any firewalls or NATing involved?



On Oct 21, 2009 10:13am, Ross Beer ross_b...@hotmail.com> wrote:
>
>
>
>
>
>
>
>
>
>
> I have a server located on the internet running opensips and asterisk.  
When registering directly to asterisk I can perform echo tests and make  
calls.

>
>
>
>
>
> If I register to Opensips and use the load_balance there is one way  
audio. I can hear sounds coming from the asterisk server but sound from  
the soft phone does not reach asterisk. I can confirm this when looking  
at a rtp debug on asterisk.

>
>
>
>
>
> I can see that traffic is passing from the soft phone when performing a  
wire shark trace to the server and it also shows that some RTP packet are  
being passed out and back into my local address. This does not happen if  
I register directly to asterisk.

>
>
>
>
>
> Any advice you can offer would be appreciated.
>
>
>
>
>
> Opensips shouldn't effect the RTP if it only load balances?
>
>
>
>
>
> Thanks,
>
>
>
> Ross
>
> Did you know you can get Messenger on your mobile? Learn more.
>
>
>
Use Windows Live Messenger for free on selected mobiles. Learn more.




___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RLS Services: subscription expanded to 0 contacts.

[Sanjeev BA]

Hi,

I am trying to get an RLS server setup working and find the following issue.

 

Client sends SUBSCRIBE with list URI. RLS expands the list URI to individual
URIs.

Sends individual subscribe to Presence Server (co-located)

Presence server responds with 202 or 200 response code.

 

I have an ngrep on port 5060 and I can see NOTIFY being sent by the presence
server.

Along with the log

Send_notify_request: NOTIFY  via  on
behalf of  for event presence.

 

However, when the contact presence status changes and is PUBLISHed, RLS does
not NOTIFY the client.

In the logs I can see the error 0 dialogs found.

 

Please let me know of any specific configuration required to make this work.

 

Regards

Sanjeev

 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Georgescu
Sent: Wednesday, October 21, 2009 8:59 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] RLS Services: subscription expanded to 0
contacts.

 

There are two major issues related to the implementing of such feature:

 

1. You expose your server to information provisioned in a remote system

2. OpenSIPS child may block waiting for the response from the remote XCAP
server

 

So though this feature is well defined in the standards, it can introduce a
lot of problems for the operator that implements it. 

 

--

Adrian

 

 





 

On Oct 21, 2009, at 10:55 AM, Sanjeev BA wrote:





Hi,

I am facing the issue posted previously.

 

It is very clear what happens. Your rls-services document looks like this:

  http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
;pres-list=Default">

http://xcap.net1.test/xcap-root/resource-lists/users/sip%3aal
ice%40net1.test/index/~~/resource-lists/list...@name=%22default%22%5d



  presence

  
]
 
It has a reference to a resouce-list document where the list is actually 
defined. Unfortunately opensips does not have support for this. It only 
works with the list being defined inside the rls-services document. This 
is on the list of improvements and will be included until the next major 
release.
 

I am using OpenSIPS 1.5. Is it addressed in the release or do I need to
upgrade to 1.6?

 

Regards,

Sanjeev

 

 

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

 

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] One Way Audio

[Ross Beer]

Yep, traffic comes from the asterisk server and can be heard on the softphone, 
but when the echo test starts no audo can be heard.
 
Therfore the flow goes like this:
 
Asterisk ---> Opensips > Softphone
 
But NOT:
 
Softphone ---> Opensips > Asterisk
 
Which is strange, if opensips is not in the path all works correctly. Also if I 
call out using a SIP provider I also get two way audio, but not when talking 
directly to asterisk.
 
Regards,
 
Ross
 


From: ross_b...@hotmail.com
To: duane.lar...@gmail.com; users@lists.opensips.org
Subject: RE: [OpenSIPS-Users] One Way Audio
Date: Wed, 21 Oct 2009 16:35:28 +0100



NHi Duane,
 
There are is a firewall on the server end however all ports are open, no NAT at 
the server end however there is NATing on the end of the soft phone. Though 
when registering with asterisk directly there is no issue.
 
Regards,
 
Ross
 


Date: Wed, 21 Oct 2009 15:23:04 +
Subject: Re: [OpenSIPS-Users] One Way Audio
From: duane.lar...@gmail.com
To: ross_b...@hotmail.com

Are there any firewalls or NATing involved? 

On Oct 21, 2009 10:13am, Ross Beer  wrote: 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> I have a server located on the internet running opensips and asterisk. When 
> registering directly to asterisk I can perform echo tests and make calls. 
> 
> 
>   
> 
> 
> If I register to Opensips and use the load_balance there is one way audio. I 
> can hear sounds coming from the asterisk server but sound from the soft phone 
> does not reach asterisk. I can confirm this when looking at a rtp debug on 
> asterisk. 
> 
> 
>   
> 
> 
> I can see that traffic is passing from the soft phone when performing a wire 
> shark trace to the server and it also shows that some RTP packet are being 
> passed out and back into my local address. This does not happen if I register 
> directly to asterisk. 
> 
> 
>   
> 
> 
> Any advice you can offer would be appreciated. 
> 
> 
>   
> 
> 
> Opensips shouldn't effect the RTP if it only load balances? 
> 
> 
>   
> 
> 
> Thanks, 
> 
> 
> 
> Ross 
> 
> Did you know you can get Messenger on your mobile? Learn more. 
> 
> 
> 


Use Windows Live Messenger for free on selected mobiles. Learn more.
  
_
Did you know you can get Messenger on your mobile?
http://clk.atdmt.com/UKM/go/174426567/direct/01/___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] install rtpproxy on same host or diff host - bridge mode or not ?

[Manivasagam Sivaraman]

I have installed rtpproxy on the same host as that of opensips, which as
only one external public IP. My rtpproxy is not running bridge
mode. Rtprpoxy NAT traversal is not working for users from 2 different
network. To make this case work, should I use bridge mode ? with 2 IP
interfaces ? or is this single host on the same host as that of opensips
work ? Please give me some inputs.

Thanks in advance.
Mani

On Wed, Oct 21, 2009 at 9:13 AM, Manivasagam Sivaraman <
smvasagam2...@gmail.com> wrote:

> Thanks Khan. I tried this before and it did not work, but your
> documentation is the most comprehensive one, great job. May be i need some
> more tune up for my setup. Basically I can make a call from UA behind NAT to
> PSTN through opesips/rtpproxy outbound. I have installed opensips and
> rtpproxy on the same Public IP address. I have a confusion over, whether I
> should run rtpproxy in bridge mode with dual interface to be able to make
> rtp flow between 2 UAs behind  2 different NATs ? Or is the single interface
> setup sufficient ?
>
> The first INVITE from UA1 to UA2 ( SDP c= IP address is modified correctly
> as rtpproxy's IP address and sent to UA2)
> The 200 OK response SDP from UA2 , reaches opensips .. and fwd as is, with
> UA2's private IP address in SDP and UA1 cannot sent media. I made sure the
> [onreply route1] method is called in script and the force_rtp_proxy() is
> invoked by printing xlog. But rtpproxy is not modifying the 200 OK SDP. This
> is my exact problem. This indicates that I might have to run rtpproxy in
> bridge mode. Also the new opensips documentation I see new functions
> rtpproxy_offerSDP and rtpproxy_answer() functions. But I need no example
> using those. Every one is still using the deprecated force_rtp_proxy(),
> which I think is ok as long as it works.
>
> Thanks in advance. I'm in Chicago too.
>
>   On Tue, Oct 20, 2009 at 9:30 PM, Khan  wrote:
>
>> Mani,
>>
>> There is a complete working configuration posted at the following blog
>> link:
>> http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html
>>
>> You might need to tuneup a little bit based on your needs but last time i
>> have tried it and it was functional.
>>
>>
>> --
>> Khan
>>
>>
>> VoIP Rookie
>> Every beginning has an end regardless we believe it or not...
>>
>>   On Tue, Oct 20, 2009 at 8:20 PM, Manivasagam Sivaraman <
>> smvasagam2...@gmail.com> wrote:
>>
>>>   Dear Pros
>>>
>>> The nathelper documentation is good as shown below
>>> http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id228280
>>>
>>> Seems like from 1.5.x many nathelper functions are deprecated and new
>>> functions are introduced like rtpproxy_offer and rtpproxy_answer.
>>>
>>> Actually NAT traversal is still a nightmare for many and beginers it
>>> would be better if some professional post a full working opensips.cfg
>>> nathelper/rtpproxy configuration script for the latest 1.5.3. THe above
>>> documents shows bits and pieces, but a full working config example will
>>> really help. Could any one please post a working example of opensips.cfg
>>> nathelper configuration please. I searched and foind only old configuration
>>> that does not work well.
>>>
>>> Thanks in Advance. I really appreaciate your understanding.
>>> Mani
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] send bye in case of parallel forked call

[Alex Balashov]

No.

--
Sent from mobile device

On Oct 21, 2009, at 9:34 AM, Uwe Kastens  wrote:

> Hi,
>
> I have the following requirement:
>
> If a from tm generated cancel is answered with a 200 OK I want to  
> send a
> BYE to the UAC.
>
> Is this possible?
>
> BR
>
> Uwe
>
> -- 
>
> kiste lat: 54.322684, lon: 10.13586
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] send bye in case of parallel forked call

[Uwe Kastens]

Hi Alex,

Any other option to solve this 200 OK for INVITE relayed after CANCEL
issue with opensips and asterisk?

http://lists.kamailio.org/pipermail/devel/2008-August/015209.html

BR

Uwe


Alex Balashov schrieb:
> No.
> 
> --
> Sent from mobile device
> 
> On Oct 21, 2009, at 9:34 AM, Uwe Kastens  wrote:
> 
>> Hi,
>>
>> I have the following requirement:
>>
>> If a from tm generated cancel is answered with a 200 OK I want to  
>> send a
>> BYE to the UAC.
>>
>> Is this possible?
>>
>> BR
>>
>> Uwe
>>
>> -- 
>>
>> kiste lat: 54.322684, lon: 10.13586
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


-- 

kiste lat: 54.322684, lon: 10.13586

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] User inbound rules

[osiris123d]

I am also setting up FindMe/FollowMe type conditional based routing on a test
box and have got it working with avp_db_query.  I am inserting the mobile
and home numbers for the called user into the location database table with
specific Q values depending on how the user wants the phones to ring and
then serializing and next_branching to get serial/parrallel calls.  I was
wondering if this is what you guys are doing to accomplish this feature and
also how you are handling the location database cleanup after the call?

Do you delete the mobile and home location entries after the call via the
avp_db_query or do you insert an "expires" time that is set to something
like 10 seconds???  I haven't moved on to the memcache stuff yet since I
first want to get the basics working.  Any info is appreciated.




Brett Nemeroff wrote:
> 
> Hi Gustavo,I'm doing something similar, being that I need a lookup per
> INVITE. Bogdan's recommendation for me was to use the memcaching functions
> in 1.5:
> http://www.opensips.org/index.php?n=Resources.DocsTutMemcache
> 
> 
> 
> On Wed, Feb 18, 2009 at 8:45 AM, Gustavo Mistrinelli
> > wrote:
> 
>> Hi all, I'm trying to figure out if we can use or add functionalities to
>> dynamic routing module to do "incoming routing" based on callee
>> destination
>> ($ru) per user
>> The idea is to have inbound rules per user (username/domain or user
>> aliases
>> i.e. numbers )
>> Each user will set their incoming rules, i.e. First rule ring my numbers
>> and username (lookup registered devices) for 15 seconds, then call my
>> cellphone for 10 second, then call
>> home number for 20 seconds and then call my voicemail, each step may ring
>> on more than one devices at the same time, it's a mix of serial and
>> parallel
>> forking. Condition to do next step is if get 4XX errors (not found, busy,
>> etc)
>> We can add also time conditions and black/white list.
>>
>> I did it using custom avp_db_query but will be nice have rules on memory
>> without querying tables every time.
>>
>> I'll be waiting for your suggestions
>>
>> Best,
>>
>> --
>> Gustavo Mistrinelli
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

-- 
View this message in context: 
http://n2.nabble.com/User-inbound-rules-tp2347188p3867601.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] One Way Audio

[Ross Beer]

It looks like it is sending in to the server's IP address and back to it's self 
which is strange.

 

I think this has something to do with the SDP and possibly my router. I am 
doing an echo test so audio should come back, however Asterisk should stay in 
the media path as it does when directly using asterisk.

 

If I use a different network there isn't a problem however directly using 
asterisk on the problem network has no issues.

 

Ideally I would like to resolve this issue so all networks can use OpenSips.

 

I am currently testing MediaProxy however it does not appear to receive the RTP 
stream from the soft phone either.

 

Thank you for you help,

 

Ross
 


Date: Wed, 21 Oct 2009 17:55:10 +
Subject: Re: RE: [OpenSIPS-Users] One Way Audio
From: duane.lar...@gmail.com
To: ross_b...@hotmail.com

In the wireshark trace what IP is the softphone sending the RTP packets to? 
Whats the destination? Is it actually sending the RTP to the Asterisk box? 

On Oct 21, 2009 11:15am, Ross Beer  wrote: 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Yep, traffic comes from the asterisk server and can be heard on the 
> softphone, but when the echo test starts no audo can be heard. 
> 
> 
>   
> 
> 
> Therfore the flow goes like this: 
> 
> 
>   
> 
> 
> Asterisk ---> Opensips > Softphone 
> 
> 
>   
> 
> 
> But NOT: 
> 
> 
>   
> 
> 
> Softphone ---> Opensips > Asterisk 
> 
> 
>   
> 
> 
> Which is strange, if opensips is not in the path all works correctly. Also if 
> I call out using a SIP provider I also get two way audio, but not when 
> talking directly to asterisk. 
> 
> 
>   
> 
> 
> Regards, 
> 
> 
>   
> 
> 
> Ross 
>   
> 
> 
> 
> 
> Date: Wed, 21 Oct 2009 15:40:38 + 
> Subject: Re: RE: [OpenSIPS-Users] One Way Audio 
> From: duane.lar...@gmail.com 
> To: ross_b...@hotmail.com; duane.lar...@gmail.com 
> CC: users@lists.opensips.org 
> 
> So in the wireshark trace you see RTP traffic coming from the Asterisk 
> servers IP address, but what about the traffic coming from the softphone? 
> What IP address is that going towards? 
> 
> On Oct 21, 2009 10:35am, Ross Beer ross_b...@hotmail.com> wrote: 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > NHi Duane, 
> > 
> > 
> >   
> > 
> > 
> > There are is a firewall on the server end however all ports are open, no 
> > NAT at the server end however there is NATing on the end of the soft phone. 
> > Though when registering with asterisk directly there is no issue. 
> > 
> > 
> >   
> > 
> > 
> > Regards, 
> > 
> > 
> >   
> > 
> > 
> > Ross 
> >   
> > 
> > 
> > 
> > 
> > Date: Wed, 21 Oct 2009 15:23:04 + 
> > Subject: Re: [OpenSIPS-Users] One Way Audio 
> > From: duane.lar...@gmail.com 
> > To: ross_b...@hotmail.com 
> > 
> > Are there any firewalls or NATing involved? 
> > 
> > On Oct 21, 2009 10:13am, Ross Beer ross_b...@hotmail.com> wrote: 
> > > 
> > > 
> > > 
> > > 
> > > 
> > > 
> > > 
> > > 
> > > 
> > > 
> > > I have a server located on the internet running opensips and asterisk. 
> > > When registering directly to asterisk I can perform echo tests and make 
> > > calls. 
> > > 
> > > 
> > >   
> > > 
> > > 
> > > If I register to Opensips and use the load_balance there is one way 
> > > audio. I can hear sounds coming from the asterisk server but sound from 
> > > the soft phone does not reach asterisk. I can confirm this when looking 
> > > at a rtp debug on asterisk. 
> > > 
> > > 
> > >   
> > > 
> > > 
> > > I can see that traffic is passing from the soft phone when performing a 
> > > wire shark trace to the server and it also shows that some RTP packet are 
> > > being passed out and back into my local address. This does not happen if 
> > > I register directly to asterisk. 
> > > 
> > > 
> > >   
> > > 
> > > 
> > > Any advice you can offer would be appreciated. 
> > > 
> > > 
> > >   
> > > 
> > > 
> > > Opensips shouldn't effect the RTP if it only load balances? 
> > > 
> > > 
> > >   
> > > 
> > > 
> > > Thanks, 
> > > 
> > > 
> > > 
> > > Ross 
> > > 
> > > Did you know you can get Messenger on your mobile? Learn more. 
> > > 
> > > 
> > > 
> > Use Windows Live Messenger for free on selected mobiles. Learn more. 
> > 
> > 
> > 
> Stay in touch with your friends through Messenger on your mobile. Learn more. 
> 
> 
> 
_
Stay in touch with your friends through Messenger on your mobile
http://clk.atdmt.com/UKM/go/174426567/direct/01/___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] One Way Audio

[duane . larson]

Now that you are testing with Mediaproxy are you sure that you are using  
the use_media_proxy() correctly? The more info you provide the more we can  
help. SIP traces are good.


On Oct 21, 2009 2:42pm, Ross Beer  wrote:










It looks like it is sending in to the server's IP address and back to  
it's self which is strange.






I think this has something to do with the SDP and possibly my router. I  
am doing an echo test so audio should come back, however Asterisk should  
stay in the media path as it does when directly using asterisk.






If I use a different network there isn'ta problem however directly using  
asterisk on the problem network has no issues.






Ideally I would like to resolve this issue so all networks can use  
OpenSips.






I am currently testing MediaProxy however it does not appear to receive  
the RTP stream from the soft phone either.







Thank you for you help,







Ross







Date: Wed, 21 Oct 2009 17:55:10 +
Subject: Re: RE: [OpenSIPS-Users] One Way Audio
From: duane.lar...@gmail.com
To: ross_b...@hotmail.com


In the wireshark trace what IP is the softphone sending the RTP packets  
to? Whats the destination? Is it actually sending the RTP to the Asterisk  
box?



On Oct 21, 2009 11:15am, Ross Beer ross_b...@hotmail.com> wrote:
>
>
>
>
>
>
>
>
>
>
> Yep, traffic comes from the asterisk server and can be heard on the  
softphone, but when the echo test starts no audo can be heard.

>
>
>
>
>
> Therfore the flow goes like this:
>
>
>
>
>
> Asterisk ---> Opensips > Softphone
>
>
>
>
>
> But NOT:
>
>
>
>
>
> Softphone ---> Opensips > Asterisk
>
>
>
>
>
> Which is strange, if opensips is not in the path all works correctly.  
Also if I call out using a SIP provider I also get two way audio, but not  
when talking directly to asterisk.

>
>
>
>
>
> Regards,
>
>
>
>
>
> Ross
>
>
>
>
>
> Date: Wed, 21 Oct 2009 15:40:38 +
> Subject: Re: RE: [OpenSIPS-Users] One Way Audio
> From: duane.lar...@gmail.com
> To: ross_b...@hotmail.com; duane.lar...@gmail.com
> CC: users@lists.opensips.org
>
> So in the wireshark trace you see RTP traffic coming from the Asterisk  
servers IP address, but what about the traffic coming from the softphone?  
What IP address is that going towards?

>
> On Oct 21, 2009 10:35am, Ross Beer ross_b...@hotmail.com> wrote:
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > NHi Duane,
> >
> >
> >
> >
> >
> > There are is a firewall on the server end however all ports are open,  
no NAT at the server end however there is NATing on the end of the soft  
phone. Though when registering with asterisk directly there is no issue.

> >
> >
> >
> >
> >
> > Regards,
> >
> >
> >
> >
> >
> > Ross
> >
> >
> >
> >
> >
> > Date: Wed, 21 Oct 2009 15:23:04 +
> > Subject: Re: [OpenSIPS-Users] One Way Audio
> > From: duane.lar...@gmail.com
> > To: ross_b...@hotmail.com
> >
> > Are there any firewalls or NATing involved?
> >
> > On Oct 21, 2009 10:13am, Ross Beer ross_b...@hotmail.com> wrote:
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > I have a server located on the internet running opensips and  
asterisk. When registering directly to asterisk I can perform echo tests  
and make calls.

> > >
> > >
> > >
> > >
> > >
> > > If I register to Opensips and use the load_balance there is one way  
audio. I can hear sounds coming from the asterisk server but sound from  
the soft phone does not reach asterisk. I can confirm this when looking  
at a rtp debug on asterisk.

> > >
> > >
> > >
> > >
> > >
> > > I can see that traffic is passing from the soft phone when  
performing a wire shark trace to the server and it also shows that some  
RTP packet are being passed out and back into my local address. This does  
not happen if I register directly to asterisk.

> > >
> > >
> > >
> > >
> > >
> > > Any advice you can offer would be appreciated.
> > >
> > >
> > >
> > >
> > >
> > > Opensips shouldn't effect the RTP if it only load balances?
> > >
> > >
> > >
> > >
> > >
> > > Thanks,
> > >
> > >
> > >
> > > Ross
> > >
> > > Did you know you can get Messenger on your mobile? Learn more.
> > >
> > >
> > >
> > Use Windows Live Messenger for free on selected mobiles. Learn more.
> >
> >
> >
> Stay in touch with your friends through Messenger on your mobile. Learn  
more.

>
>
>
Stay in touch with your friends through Messenger on your mobile. Learn  
more.




___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] nat problems with one-way audio and forwarding

[prescott]

I need some advice:
I have a test case that looks like this:
outside customer calls a phone number, number is busy.
opensips looks up the customer preference and forwards the busy call to another 
phone.

the first (busy) number is behind  a nat.
the second is not.
I am using rtpproxy for my media relaying on the nat side.
The problem is that when the 200 OK response ggets sent from the second phone 
picking up the call, opensips does not fix the sdp in the message.

This results in one-way audio on the call.
I am using opensips-1.5.3
I am attaching the sip trace for reference.
The way I do call forwarding is: look for the 486 busy response and then 
append_branch to the forwarded destination.
as you can see in the invite, the sdp information is retained, but the system 
doesn't seem to recognise the ok response as part of that sip transaction.

Any help or suggestions of where to look would be appreciated.
Thanks.

-- Kelly Prescott65.17.128.151 is the outside telco gateway

65.17.129.121 is the phone1 which is busy
( it is natted)

65.17.129.124 is the phone which is forwarded to on busy.
( it is not natted.)

#
U 2009/10/21 16:02:00.468484 65.17.128.151:5060 -> 65.17.128.34:5060
INVITE sip:14195164...@65.17.128.34 SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP 65.17.128.151;branch=z9hG4bK4803.52e16032.0.
Via: SIP/2.0/UDP 
64.156.174.74:5060;rport=5060;branch=z9hG4bK02B43e45559190ff6fd.
From: "LIMA OH" 
;tag=gK020297fd.
To: .
Call-ID: 1073924919_32347...@64.156.174.74.
CSeq: 20889 INVITE.
Max-Forwards: 69.
Allow: 
INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Accept: application/sdp, application/isup, application/dtmf, 
application/dtmf-relay,  multipart/mixed.
Contact: "LIMA OH" .
Remote-Party-ID: "LIMA OH" 
;privacy=off.
Content-Length:  337.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 22204 2866 IN IP4 64.156.174.74.
s=SIP Media Capabilities.
c=IN IP4 64.156.174.71.
t=0 0.
m=audio 14514 RTP/AVP 0 18 4 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000.
a=fmtp:4 annexa=no;bitrate=6.3.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=ptime:30.

#
U 2009/10/21 16:02:00.470995 65.17.128.34:5060 -> 65.17.128.151:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 65.17.128.151;branch=z9hG4bK4803.52e16032.0;rport=5060.
Via: SIP/2.0/UDP 
64.156.174.74:5060;rport=5060;branch=z9hG4bK02B43e45559190ff6fd.
From: "LIMA OH" 
;tag=gK020297fd.
To: .
Call-ID: 1073924919_32347...@64.156.174.74.
CSeq: 20889 INVITE.
Server: OpenSIPS (1.5.3-notls (i386/linux)).
Content-Length: 0.
.

#
U 2009/10/21 16:02:00.488688 65.17.128.34:5060 -> 65.17.129.121:5060
INVITE sip:14195164...@192.168.1.100:5060 SIP/2.0.
Record-Route: .
Record-Route: .
Via: SIP/2.0/UDP 65.17.128.34;branch=z9hG4bK4803.5e2f8764.0.
Via: SIP/2.0/UDP 
65.17.128.151;rport=5060;received=65.17.128.151;branch=z9hG4bK4803.52e16032.0.
Via: SIP/2.0/UDP 
64.156.174.74:5060;rport=5060;branch=z9hG4bK02B43e45559190ff6fd.
From: "LIMA OH" 
;tag=gK020297fd.
To: .
Call-ID: 1073924919_32347...@64.156.174.74.
CSeq: 20889 INVITE.
Max-Forwards: 68.
Allow: 
INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Accept: application/sdp, application/isup, application/dtmf, 
application/dtmf-relay,  multipart/mixed.
Contact: "LIMA OH" .
Remote-Party-ID: "LIMA OH" 
;privacy=off.
Content-Length: 353.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
P-Charge-Info: .
.
v=0.
o=Sonus_UAC 22204 2866 IN IP4 65.17.128.34.
s=SIP Media Capabilities.
c=IN IP4 65.17.128.34.
t=0 0.
m=audio 62242 RTP/AVP 0 18 4 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000.
a=fmtp:4 annexa=no;bitrate=6.3.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=ptime:30.
a=nortpproxy:yes.

#
U 2009/10/21 16:02:00.521804 65.17.129.121:5060 -> 65.17.128.34:5060
SIP/2.0 100 Trying.
To: .
From: "LIMA OH" 
;tag=gK020297fd.
Call-ID: 1073924919_32347...@64.156.174.74.
CSeq: 20889 INVITE.
Via: SIP/2.0/UDP 65.17.128.34;branch=z9hG4bK4803.5e2f8764.0.
Via: SIP/2.0/UDP 
65.17.128.151;rport=5060;received=65.17.128.151;branch=z9hG4bK4803.52e16032.0.
Via: SIP/2.0/UDP 
64.156.174.74:5060;rport=5060;branch=z9hG4bK02B43e45559190ff6fd.
Record-Route: .
Record-Route: .
Server: Linksys/SPA1001-3.1.19(SE).
Content-Length: 0.
.

#
U 2009/10/21 16:02:00.532062 65.17.129.121:5060 -> 65.17.128.34:5060
SIP/2.0 486 Busy Here.
To: ;tag=889cb26a984d13c1i0.
From: "LIMA OH" 
;tag=gK020297fd.
Call-ID: 1073924919_32347...@64.156.174.74.
CSeq: 20889 INVITE.
Via: SIP/2.0/UDP 65.17.128.34;branch=z9hG4bK4803.5e2f8764.0.
Via: SIP/2.0/UDP 
65.17.128.151;rport=5060;received=65.17.128.151;branch=z9hG4bK4803.52e16032.0.
Via: SIP/2.0/UDP 
64.156.174.74:5060;rport=5060;branch=z9hG4bK02B43e45559190ff6fd.
Record-Route: .
Record-Route: .
Server: Linksy

[OpenSIPS-Users] Does Opensips nat function handle Subscribe/Notify ?

[Manivasagam Sivaraman]

Do any one know if there is a function like fixed_nated_subscribe() , just
like fix_nated_register() ? I'm facing problem where opensips is not fixing
the natted subscribes ? I'm using rtpproxy for INVITE/Rtp and it works fine.

Please help
Mani

On Wed, Oct 21, 2009 at 3:33 PM,  wrote:

> I need some advice:
> I have a test case that looks like this:
> outside customer calls a phone number, number is busy.
> opensips looks up the customer preference and forwards the busy call to
> another phone.
> the first (busy) number is behind  a nat.
> the second is not.
> I am using rtpproxy for my media relaying on the nat side.
> The problem is that when the 200 OK response ggets sent from the second
> phone picking up the call, opensips does not fix the sdp in the message.
> This results in one-way audio on the call.
> I am using opensips-1.5.3
> I am attaching the sip trace for reference.
> The way I do call forwarding is: look for the 486 busy response and then
> append_branch to the forwarded destination.
> as you can see in the invite, the sdp information is retained, but the
> system doesn't seem to recognise the ok response as part of that sip
> transaction.
> Any help or suggestions of where to look would be appreciated.
> Thanks.
>
> -- Kelly Prescott
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] tel-uri

[Iñaki Baz Castillo]

El Miércoles, 21 de Octubre de 2009, Airton Kuada escribió:
>  Hi all.
> 
> Anybody know what is the schema tel-uri?

http://www.google.com/search?q=tel+URI+RFC&ie=UTF-8&oe=UTF-8


>  OpenSIPS works with the schema tel-uri?

Not very well but it accepts it and parses it as follows:

  - username = TEL NUMBER
  - domain = null

Example:

  INVITE tel:+123456789 SIP/2.0
=>
  - $rU = +12345678
  - $rd = null

Be careful because if you modify $rU the generated Request Line would look 
like:

  INVITE sip:NEW_rU@ SIP/2.0  (malformed).
  

-- 
Iñaki Baz Castillo 

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Does Opensips nat function handle Subscribe/Notify?

[Iñaki Baz Castillo]

El Miércoles, 21 de Octubre de 2009, Manivasagam Sivaraman escribió:
> Do any one know if there is a function like fixed_nated_subscribe() , just
> like fix_nated_register() ? I'm facing problem where opensips is not fixing
> the natted subscribes ? I'm using rtpproxy for INVITE/Rtp and it works
>  fine.

Subscriptions is just signalling so fix_nat_contact() is enough. Just it.




PS: Why the following copy&paste?:

> On Wed, Oct 21, 2009 at 3:33 PM,  wrote:
> > I need some advice:
> > I have a test case that looks like this:
> > outside customer calls a phone number, number is busy.
> > opensips looks up the customer preference and forwards the busy call to
> > another phone.
> > the first (busy) number is behind  a nat.
> > the second is not.
> > I am using rtpproxy for my media relaying on the nat side.
> > The problem is that when the 200 OK response ggets sent from the second
> > phone picking up the call, opensips does not fix the sdp in the message.
> > This results in one-way audio on the call.
> > I am using opensips-1.5.3
> > I am attaching the sip trace for reference.
> > The way I do call forwarding is: look for the 486 busy response and then
> > append_branch to the forwarded destination.
> > as you can see in the invite, the sdp information is retained, but the
> > system doesn't seem to recognise the ok response as part of that sip
> > transaction.
> > Any help or suggestions of where to look would be appreciated.
> > Thanks.
> >
> > -- Kelly Prescott
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 


-- 
Iñaki Baz Castillo 

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Audio problem

[Justin L]

Hi,

I have a question related to my load balancing configuration of opensips.

I have an X-Lite softphone that connects to Opensips server, which transfers
the INVITE request to one of the asterisk boxes.
All of them are behind firewall on the same network. Then asterisk calls to
my cell phone through the voip provider.

The SIP balancing works fine and I get the call, but there is no audio. The
firewall should be configured correctly to transfer the SIP and RTP ports.

Since I just started to use opensips it sounds to me like a very basic
problem, that many people probably have faced.
Could you please recommend me a  way to troubleshoot this issue?

Thanks a lot,

Justin.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Audio problem

[Raúl Alexis Betancor Santana]

On Wednesday 21 October 2009 23:13:36 Justin L wrote:
> Hi,
>
> I have a question related to my load balancing configuration of opensips.
>
> I have an X-Lite softphone that connects to Opensips server, which
> transfers the INVITE request to one of the asterisk boxes.
> All of them are behind firewall on the same network. Then asterisk calls to
> my cell phone through the voip provider.
>
> The SIP balancing works fine and I get the call, but there is no audio. The
> firewall should be configured correctly to transfer the SIP and RTP ports.
>
> Since I just started to use opensips it sounds to me like a very basic
> problem, that many people probably have faced.
> Could you please recommend me a  way to troubleshoot this issue?
>
> Thanks a lot,
>
> Justin.

Some SIP trace would be nice to begin ...

-- 
Raúl Alexis Betancor Santana
Dimensión Virtual

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Audio problem

[Justin L]

Here is the INVITE:

INVITE sip:13101234...@ask00-rvn SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 10.1.3.130;branch=z9hG4bK88c5.4ae45bf5.0
Via: SIP/2.0/UDP 172.16.100.159:21874
;received=172.16.100.159;branch=z9hG4bK-d87543-2376e4785757b07b-1--d87543-;rport=21874
Max-Forwards: 69
Contact: 
To: "13101234567"
From: "2";tag=c020195b
Call-ID: NDg4Y2Y0ZWU5MGM4NjhiNWVlZGNiZTc1ZGQxMjlhYzc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 529

v=0
o=- 8 2 IN IP4 172.16.100.159
s=CounterPath X-Lite 3.0
c=IN IP4 172.16.100.159
t=0 0
m=audio 39148 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 3 : 5AkMoAfO yRnFlRIn 172.16.100.159 39148
a=alt:2 2 : 7PbWVKqn VccqHBD1 192.168.2.59 39148
a=alt:3 1 : TXSbExav /8BXXCL+ 192.168.176.152 39148
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv


2009/10/21 Raúl Alexis Betancor Santana 

> On Wednesday 21 October 2009 23:13:36 Justin L wrote:
> > Hi,
> >
> > I have a question related to my load balancing configuration of opensips.
> >
> > I have an X-Lite softphone that connects to Opensips server, which
> > transfers the INVITE request to one of the asterisk boxes.
> > All of them are behind firewall on the same network. Then asterisk calls
> to
> > my cell phone through the voip provider.
> >
> > The SIP balancing works fine and I get the call, but there is no audio.
> The
> > firewall should be configured correctly to transfer the SIP and RTP
> ports.
> >
> > Since I just started to use opensips it sounds to me like a very basic
> > problem, that many people probably have faced.
> > Could you please recommend me a  way to troubleshoot this issue?
> >
> > Thanks a lot,
> >
> > Justin.
>
> Some SIP trace would be nice to begin ...
>
> --
> Raúl Alexis Betancor Santana
> Dimensión Virtual
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] User inbound rules

[Bogdan-Andrei Iancu]

Hi,

you can do the Follow me / call hunting by placing the additional 
destinations in the usrloc (and use serialize), or you can let usrloc to 
work in the normal way (no permanent contacts) and in failure route, 
when you got the negative reply from the registered phone, you can load 
the new contacts to try from DB and push them for usage.

Regards,
Bogdan

osiris123d wrote:
> I am also setting up FindMe/FollowMe type conditional based routing on a test
> box and have got it working with avp_db_query.  I am inserting the mobile
> and home numbers for the called user into the location database table with
> specific Q values depending on how the user wants the phones to ring and
> then serializing and next_branching to get serial/parrallel calls.  I was
> wondering if this is what you guys are doing to accomplish this feature and
> also how you are handling the location database cleanup after the call?
>
> Do you delete the mobile and home location entries after the call via the
> avp_db_query or do you insert an "expires" time that is set to something
> like 10 seconds???  I haven't moved on to the memcache stuff yet since I
> first want to get the basics working.  Any info is appreciated.
>
>
>
>
> Brett Nemeroff wrote:
>   
>> Hi Gustavo,I'm doing something similar, being that I need a lookup per
>> INVITE. Bogdan's recommendation for me was to use the memcaching functions
>> in 1.5:
>> http://www.opensips.org/index.php?n=Resources.DocsTutMemcache
>>
>>
>>
>> On Wed, Feb 18, 2009 at 8:45 AM, Gustavo Mistrinelli
>> > 
>>> wrote:
>>>   
>>> Hi all, I'm trying to figure out if we can use or add functionalities to
>>> dynamic routing module to do "incoming routing" based on callee
>>> destination
>>> ($ru) per user
>>> The idea is to have inbound rules per user (username/domain or user
>>> aliases
>>> i.e. numbers )
>>> Each user will set their incoming rules, i.e. First rule ring my numbers
>>> and username (lookup registered devices) for 15 seconds, then call my
>>> cellphone for 10 second, then call
>>> home number for 20 seconds and then call my voicemail, each step may ring
>>> on more than one devices at the same time, it's a mix of serial and
>>> parallel
>>> forking. Condition to do next step is if get 4XX errors (not found, busy,
>>> etc)
>>> We can add also time conditions and black/white list.
>>>
>>> I did it using custom avp_db_query but will be nice have rules on memory
>>> without querying tables every time.
>>>
>>> I'll be waiting for your suggestions
>>>
>>> Best,
>>>
>>> --
>>> Gustavo Mistrinelli
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>   
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> 
>
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE

[Bogdan-Andrei Iancu]

Hi Uwe,

as I understand from you, from end devices (GW, as1 and as2) everything 
work ok, but the dialog state on opensips is not properly kept??

Regards,
Bogdan

Uwe Kastens wrote:
> Hello Bogdan,
>
> Now we changed the behaviour of the UAC. One of them will send a BYE and
> this is relayed to the PSTN GW which drops the call, since opensips will
> not handle the BYE locally. So loose_route is done and the BYE is
> relayed to the PSTN GW.
>
> The following is happening:
>
> 1) INVITE from PSTN GW
> 2) parallel forking to ast1 and ast2 (branches z9hG4bK51f6.9afa91c3.1
> and z9hG4bK51f6.9afa91c3.0)
> 3) ast1 sends an 200 OK (branch z9hG4bK51f6.9afa91c3.0)
> 4) opensips sends an cancel to ast2 (branch z9hG4bK51f6.9afa91c3.1)
> 5) opensips receives the 200 OK from ast1 and sends an ACK (branch is
> changing here to z9hG4bK51f6.9afa91c3.3)
> 6) opensips receives 200 OK from ast2 from the INVITE (branch
> z9hG4bK51f6.9afa91c3.1)and sends an ACK (branch is changing to
> z9hG4bK51f6.9afa91c3.3)
> 7) opensips reives 200 OK from ast2 for the cancel request ( branch
> z9hG4bK51f6.9afa91c3.1)
> 8) opensips receives BYE from ast2 with branch z9hG4bK40d1af5d
> 9) opensips is doing loose_route and sends the BYE to the PSTN GW
>
>
> The only thing I could see on the logs is:
>
>  WARNING:dialog:dlg_onroute: tight matching failed for BYE with
> callid='393105a419950c1f265f298914662...@10.20.30.100'/46,
> ftag='as63949c6e'/10, ttag='as0d1597ca'/10 and direction=0
> Oct 21 09:09:15 asne02 /usr/sbin/opensips[15615]:
> WARNING:dialog:dlg_onroute: dialog identification elements are
> callid='393105a419950c1f265f298914662...@10.20.30.100'/46, caller
> tag='as0d1597ca'/10, callee tag='as79debd51'/10
>
> Why is the opensips not handling the BYE locally and only closing one
> branch?
>
> BR
>
> UWe
>
>
> Bogdan-Andrei Iancu schrieb:
>   
>> Hi Uwe,
>>
>> Uwe Kastens wrote:
>> 
>>> Hi Bogdan,
>>>
>>>   
>>>   
 So actually both legs do send 200 OK (but one faster than the 
 other)..so there is kind on race between the 200 OK from the slow 
 branch and the CANCEL from OpenSIPS...is this the case?
 
 
>>> Exactly
>>>
>>>   
>>>   
 If so, the UAS will simply reply with negative reply to CANCEL (decline 
 it) and opensips (for INVITE transaction) will not close the second 
 branch as there is a 200 OK (and not a 487) received RFC3261 says 
 that a proxy must send all 200 OK (for a call), even if more than one, 
 to the UAC - the UAC is the one who will decide what branch to keep and 
 it will fire a BYE for the other branch.

 
 
>>> Could this explan, why only the 2nd Node will get the BYE, if the call
>>> is released "behind" the opensips?
>>>   
>>>   
>> yes, because the caller will hung up only one of the callee branch, so 
>> the BYE will go to only one of them. The other branch will remain up and 
>> will be the ongoing call.
>>
>> Regards,
>> Bogdan
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>
>
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] append_branches in registrar module of 1.6

[Bogdan-Andrei Iancu]

Hi Jayesh,

The append_branch() module param was converted to a function param for 
the lookup(). See:
   
http://www.opensips.org/html/docs/modules/devel/registrar.html#id271025

the "b" flag.

Regards,
Bogdan

Jayesh Nambiar wrote:
> Hello,
> I dont see the parameter append_branches in the registrar module of 
> 1.6 version. I was using it so that Openser does not route calls to 
> all the AORs. My requirement was that the last registered client 
> should get the call.
> So i used append_branches as 0 in the registrar module and 
> desc_time_order as 1 in the usrloc module which fulfilled my 
> requirement properly.
> How do i go ahead with 1.6 with similar requirement??
>
> Any help or pointers will be greatly appreciated.
>
> Thanks,
>
> --- Jayesh
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] 1.6 core dump uac_replace_to and uac_replace_from

[Brad Bendy]

Hi,

Running 1.6 non SVN we are getting random crashes, it appears this is 
from the uac_replace_from and uac_replace_to, we did not have this 
problem until we started using these function. Below is the bt from gdb.

The only errors we get in the logs are from memcache, which I think are 
unrelated from this, has anyone else seen this issue? I can try the 
latest SVN tonight and see how if that fixes it.

Thanks for any help or suggestions.

#0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
branch=1, request=0x801647020) at t_fwd.c:132
132 memcpy( p, request->dst_uri.s, 
request->dst_uri.len);



#0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
branch=1, request=0x801647020) at t_fwd.c:132
#1  0x000801518092 in add_uac (t=0x802a18bf8, request=0x801647020, 
uri=0x7fffd0d0,
next_hop=0x7fffd0e0, path=Variable "path" is not available.
) at t_fwd.c:400
#2  0x000801519bda in t_forward_nonack (t=0x802a18bf8, 
p_msg=0x801647020, proxy=0x0) at t_fwd.c:625
#3  0x000801531d1e in w_t_relay (p_msg=0x801647020, proxy=0x0, 
flags=Variable "flags" is not available.
) at tm.c:1101
#4  0x0040d9d1 in do_action (a=0x6c7a78, msg=0x801647020) at 
action.c:962
#5  0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#6  0x00465b69 in eval_expr (e=0x6c7b48, msg=0x801647020, 
val=0x0) at route.c:1240
#7  0x0046584e in eval_expr (e=0x6c7b90, msg=0x801647020, 
val=0x0) at route.c:1553
#8  0x00465869 in eval_expr (e=0x6c7bd8, msg=0x801647020, 
val=0x0) at route.c:1558
#9  0x0040e0c7 in do_action (a=0x6c7d80, msg=0x801647020) at 
action.c:689
#10 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#11 0x0040eee5 in do_action (a=0x6f2310, msg=0x801647020) at 
action.c:119
#12 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#13 0x0040fc6c in do_action (a=0x6f24b0, msg=0x801647020) at 
action.c:706
#14 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#15 0x00410ce8 in run_top_route (a=0x6f1d28, msg=0x801647020) at 
action.c:119
#16 0x000801529a4d in t_should_relay_response (Trans=0x802a18bf8, 
new_code=Variable "new_code" is not available.
) at t_reply.c:612
#17 0x00080152b5f6 in relay_reply (t=0x802a18bf8, p_msg=0x71f3b0, 
branch=0, msg_status=503, cancel_bitmap=dwarf2_read_address: Corrupted 
DWARF expression.
)
at t_reply.c:1124
#18 0x00080152cc76 in reply_received (p_msg=0x71f3b0) at t_reply.c:1493
#19 0x00420d77 in forward_reply (msg=0x71f3b0) at forward.c:559
#20 0x004551d5 in receive_msg (
buf=0x659e40 "SIP/2.0 503 Service Unavailable\r\nVia: SIP/2.0/UDP 
72.44.195.164:5060;branch=z9hG4bK8ac3.4b441ca2.0\r\nVia: SIP/2.0/UDP 
68.165.121.51:5060;branch=z9hG4bK8ac3.9b5220d1.1\r\nVia: SIP/2.0/UDP 
xx.xx.xx.xx:50"...,
len=665, rcv_info=0x7fffeac0) at receive.c:200
#21 0x00499297 in udp_rcv_loop () at udp_server.c:492
#22 0x0042865a in main (argc=3, argv=Variable "argv" is not 
available.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] send bye in case of parallel forked call

[Bogdan-Andrei Iancu]

Hi Uwe,

I'm trying to follow why you actually need this ? sorting out multiple 
200 OKs of a call is not typically a just for a proxy, but rather 
something that needs to handled between end points.

There are ways to do the BYEing on OpeSIPS (for the second 200 OK), but 
I have the feeling that maybe we should look for a more conventional 
problem.

So, more or less what is the exact scenario that fails for you and why 
do you need the help from the proxy side ?

Regards,
Bogdan

Uwe Kastens wrote:
> Hi,
>
> I have the following requirement:
>
> If a from tm generated cancel is answered with a 200 OK I want to send a
> BYE to the UAC.
>
> Is this possible?
>
> BR
>
> Uwe
>
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] User inbound rules

[osiris123d]

Yeah.  I did some more testing with manually placing temporary records in
usrloc and I am thinking thats not a good method because if one of the
internal users gets called by multiple people at the same time then my
current config would place duplicate Home and Mobile records in usrloc and
that seems to cause a major issue.

I will play with it some more and possibly do like you say and just make the
changes in the Failure Route.  The other thing I was thinking about was
using the new feature in OpenSIPS 1.6 that allows you to edit branches like
so
$(branch(q)[0]) = $avp(s:office_q_value);
$(branch(q)[1]) = $avp(s:mobile_q_value);
$(branch(q)[2]) = $avp(s:home_q_value);
I am thinking that there might be a way to use this to accomplish what I
want, but will have to play with it first.

Thanks


On Wed, Oct 21, 2009 at 7:27 PM, Bogdan-Andrei Iancu [via OpenSIPS (Open SIP
Server)] 

> wrote:

> Hi,
>
> you can do the Follow me / call hunting by placing the additional
> destinations in the usrloc (and use serialize), or you can let usrloc to
> work in the normal way (no permanent contacts) and in failure route,
> when you got the negative reply from the registered phone, you can load
> the new contacts to try from DB and push them for usage.
>
> Regards,
> Bogdan
>
> osiris123d wrote:
> > I am also setting up FindMe/FollowMe type conditional based routing on a
> test
> > box and have got it working with avp_db_query.  I am inserting the mobile
>
> > and home numbers for the called user into the location database table
> with
> > specific Q values depending on how the user wants the phones to ring and
> > then serializing and next_branching to get serial/parrallel calls.  I was
>
> > wondering if this is what you guys are doing to accomplish this feature
> and
> > also how you are handling the location database cleanup after the call?
> >
> > Do you delete the mobile and home location entries after the call via the
>
> > avp_db_query or do you insert an "expires" time that is set to something
> > like 10 seconds???  I haven't moved on to the memcache stuff yet since I
> > first want to get the basics working.  Any info is appreciated.
> >
> >
> >
> >
> > Brett Nemeroff wrote:
> >
> >> Hi Gustavo,I'm doing something similar, being that I need a lookup per
> >> INVITE. Bogdan's recommendation for me was to use the memcaching
> functions
> >> in 1.5:
> >> http://www.opensips.org/index.php?n=Resources.DocsTutMemcache
> >>
> >>
> >>
> >> On Wed, Feb 18, 2009 at 8:45 AM, Gustavo Mistrinelli
> >> <[hidden 
> >> email]
> >>
> >>> wrote:
> >>>
> >>> Hi all, I'm trying to figure out if we can use or add functionalities
> to
> >>> dynamic routing module to do "incoming routing" based on callee
> >>> destination
> >>> ($ru) per user
> >>> The idea is to have inbound rules per user (username/domain or user
> >>> aliases
> >>> i.e. numbers )
> >>> Each user will set their incoming rules, i.e. First rule ring my
> numbers
> >>> and username (lookup registered devices) for 15 seconds, then call my
> >>> cellphone for 10 second, then call
> >>> home number for 20 seconds and then call my voicemail, each step may
> ring
> >>> on more than one devices at the same time, it's a mix of serial and
> >>> parallel
> >>> forking. Condition to do next step is if get 4XX errors (not found,
> busy,
> >>> etc)
> >>> We can add also time conditions and black/white list.
> >>>
> >>> I did it using custom avp_db_query but will be nice have rules on
> memory
> >>> without querying tables every time.
> >>>
> >>> I'll be waiting for your suggestions
> >>>
> >>> Best,
> >>>
> >>> --
> >>> Gustavo Mistrinelli
> >>>
> >>> ___
> >>> Users mailing list
> >>> [hidden 
> >>> email]
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>>
> >>>
> >>>
> >> ___
> >> Users mailing list
> >> [hidden 
> >> email]
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >>
> >>
> >
> >
>
>
> ___
> Users mailing list
> [hidden 
> email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
> View message @
> http://n2.nabble.com/User-inbound-rules-tp2347188p3869416.html
> To unsubscribe from Re: User inbound rules, click here< (link removed) =>.
>
>
>


-- 
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
--

-- 
View this message in context: 
http://n2.nabble.com/User-inbound-rules-tp2347188p3870025.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/li

Re: [OpenSIPS-Users] One Way Audio

[Bogdan-Andrei Iancu]

Hi Ross,

Actually you do not need any media relay (mediaproxy or rtpproxy) here. 
As time as Asterisk is on the public side, it should directly work even 
with a natted client.

What you have to check is the SDP received by the nated client in the 
200 OK - check what IP it is instructed to send traffic to. Normally it 
should be the IP of Asterisk.

Regards,
Bogdan

Ross Beer wrote:
> It looks like it is sending in to the server's IP address and back to 
> it's self which is strange.
>  
> I think this has something to do with the SDP and possibly my router. 
> I am doing an echo test so audio should come back, however Asterisk 
> should stay in the media path as it does when directly using asterisk.
>  
> If I use a different network there isn't a problem however directly 
> using asterisk on the problem network has no issues.
>  
> Ideally I would like to resolve this issue so all networks can use 
> OpenSips.
>  
> I am currently testing MediaProxy however it does not appear to 
> receive the RTP stream from the soft phone either.
>  
> Thank you for you help,
>  
> Ross
>  
> 
> Date: Wed, 21 Oct 2009 17:55:10 +
> Subject: Re: RE: [OpenSIPS-Users] One Way Audio
> From: duane.lar...@gmail.com
> To: ross_b...@hotmail.com
>
> In the wireshark trace what IP is the softphone sending the RTP 
> packets to? Whats the destination? Is it actually sending the RTP to 
> the Asterisk box?
>
> On Oct 21, 2009 11:15am, Ross Beer  wrote:
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Yep, traffic comes from the asterisk server and can be heard on the 
> softphone, but when the echo test starts no audo can be heard.
> >
> >
> >  
> >
> >
> > Therfore the flow goes like this:
> >
> >
> >  
> >
> >
> > Asterisk ---> Opensips > Softphone
> >
> >
> >  
> >
> >
> > But NOT:
> >
> >
> >  
> >
> >
> > Softphone ---> Opensips > Asterisk
> >
> >
> >  
> >
> >
> > Which is strange, if opensips is not in the path all works 
> correctly. Also if I call out using a SIP provider I also get two way 
> audio, but not when talking directly to asterisk.
> >
> >
> >  
> >
> >
> > Regards,
> >
> >
> >  
> >
> >
> > Ross
> >  
> >
> >
> >
> >
> > Date: Wed, 21 Oct 2009 15:40:38 +
> > Subject: Re: RE: [OpenSIPS-Users] One Way Audio
> > From: duane.lar...@gmail.com
> > To: ross_b...@hotmail.com; duane.lar...@gmail.com
> > CC: users@lists.opensips.org
> >
> > So in the wireshark trace you see RTP traffic coming from the 
> Asterisk servers IP address, but what about the traffic coming from 
> the softphone? What IP address is that going towards?
> >
> > On Oct 21, 2009 10:35am, Ross Beer ross_b...@hotmail.com> wrote:
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > NHi Duane,
> > >
> > >
> > >  
> > >
> > >
> > > There are is a firewall on the server end however all ports are 
> open, no NAT at the server end however there is NATing on the end of 
> the soft phone. Though when registering with asterisk directly there 
> is no issue.
> > >
> > >
> > >  
> > >
> > >
> > > Regards,
> > >
> > >
> > >  
> > >
> > >
> > > Ross
> > >  
> > >
> > >
> > >
> > >
> > > Date: Wed, 21 Oct 2009 15:23:04 +
> > > Subject: Re: [OpenSIPS-Users] One Way Audio
> > > From: duane.lar...@gmail.com
> > > To: ross_b...@hotmail.com
> > >
> > > Are there any firewalls or NATing involved?
> > >
> > > On Oct 21, 2009 10:13am, Ross Beer ross_b...@hotmail.com> wrote:
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > I have a server located on the internet running opensips and 
> asterisk. When registering directly to asterisk I can perform echo 
> tests and make calls.
> > > >
> > > >
> > > >  
> > > >
> > > >
> > > > If I register to Opensips and use the load_balance there is one 
> way audio. I can hear sounds coming from the asterisk server but sound 
> from the soft phone does not reach asterisk. I can confirm this when 
> looking at a rtp debug on asterisk.
> > > >
> > > >
> > > >  
> > > >
> > > >
> > > > I can see that traffic is passing from the soft phone when 
> performing a wire shark trace to the server and it also shows that 
> some RTP packet are being passed out and back into my local address. 
> This does not happen if I register directly to asterisk.
> > > >
> > > >
> > > >  
> > > >
> > > >
> > > > Any advice you can offer would be appreciated.
> > > >
> > > >
> > > >  
> > > >
> > > >
> > > > Opensips shouldn't effect the RTP if it only load balances?
> > > >
> > > >
> > > >  
> > > >
> > > >
> > > > Thanks,
> > > >
> > > >
> > > >
> > > > Ross
> > > >
> > > > Did you know you can get Messenger on your mobile? Learn more.
> > > >
> > > >
> > > >
> > > Use Windows Live Messenger for free on selected mobiles. Learn more.
> > >
> > >
> > >
> > Stay in touch with your friends through Messenger on your mobile. 
> Learn more.
> >
> >
> >
> 
> St

Re: [OpenSIPS-Users] install rtpproxy on same host or diff host - bridge mode or not ?

[Bogdan-Andrei Iancu]

Hi Mani,

so opensips and rtpproxy are situated in the public internet and you 
have clients behind different NATs ? is this correct? If so, you do not 
bridging mode for RTPproxy.

Regards,
Bogdan

Manivasagam Sivaraman wrote:
> I have installed rtpproxy on the same host as that of opensips, which 
> as only one external public IP. My rtpproxy is not running bridge 
> mode. Rtprpoxy NAT traversal is not working for users from 2 different 
> network. To make this case work, should I use bridge mode ? with 2 IP 
> interfaces ? or is this single host on the same host as that of 
> opensips work ? Please give me some inputs.
>  
> Thanks in advance.
> Mani
>
> On Wed, Oct 21, 2009 at 9:13 AM, Manivasagam Sivaraman 
> mailto:smvasagam2...@gmail.com>> wrote:
>
> Thanks Khan. I tried this before and it did not work, but your
> documentation is the most comprehensive one, great job. May be i
> need some more tune up for my setup. Basically I can make a call
> from UA behind NAT to PSTN through opesips/rtpproxy outbound. I
> have installed opensips and rtpproxy on the same Public IP
> address. I have a confusion over, whether I should run rtpproxy in
> bridge mode with dual interface to be able to make rtp flow
> between 2 UAs behind  2 different NATs ? Or is the single
> interface setup sufficient ?
>  
> The first INVITE from UA1 to UA2 ( SDP c= IP address is modified
> correctly as rtpproxy's IP address and sent to UA2)
> The 200 OK response SDP from UA2 , reaches opensips .. and fwd as
> is, with UA2's private IP address in SDP and UA1 cannot sent
> media. I made sure the [onreply route1] method is called in script
> and the force_rtp_proxy() is invoked by printing xlog. But
> rtpproxy is not modifying the 200 OK SDP. This is my exact
> problem. This indicates that I might have to run rtpproxy in
> bridge mode. Also the new opensips documentation I see new
> functions rtpproxy_offerSDP and rtpproxy_answer() functions. But I
> need no example using those. Every one is still using the
> deprecated force_rtp_proxy(), which I think is ok as long as it works.
>  
> Thanks in advance. I'm in Chicago too.
>
> On Tue, Oct 20, 2009 at 9:30 PM, Khan  > wrote:
>
> Mani,
>
> There is a complete working configuration posted at the
> following blog link:
> http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html
>
> You might need to tuneup a little bit based on your needs but
> last time i have tried it and it was functional.
>
>
> -- 
> Khan
>
>
> VoIP Rookie
> Every beginning has an end regardless we believe it or not...
>
> On Tue, Oct 20, 2009 at 8:20 PM, Manivasagam Sivaraman
> mailto:smvasagam2...@gmail.com>> wrote:
>
> Dear Pros
>  
> The nathelper documentation is good as shown below
> 
> http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id228280
>  
> Seems like from 1.5.x many nathelper functions are
> deprecated and new functions are introduced like
> rtpproxy_offer and rtpproxy_answer.
>  
> Actually NAT traversal is still a nightmare for many and
> beginers it would be better if some professional post a
> full working opensips.cfg nathelper/rtpproxy configuration
> script for the latest 1.5.3. THe above documents shows
> bits and pieces, but a full working config example will
> really help. Could any one please post a working example
> of opensips.cfg nathelper configuration please. I searched
> and foind only old configuration  that does not work well.
>  
> Thanks in Advance. I really appreaciate your understanding.
> Mani
>
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] nat problems with one-way audio and forwarding

[Bogdan-Andrei Iancu]

Hi Kelly,

There are 2 approaches:

1) if you enabled rtpproxy in request route for the INVITE, then, 
whatever branches you keep forking, take care and force rtpproxy in all 
200 OK (whatever branch - nated or not).

2) enable rtproxy individually, per branch - instead of using the 
request route for forcing RTPP in the INVITE, do use branch_route[] as 
changes in branch route do apply only for that branch and not to all 
branches (so for the second not nated branch, you can simply avoid the 
usage of RTPP).

Regards,
Bogdan

presc...@wcoil.com wrote:
> I need some advice:
> I have a test case that looks like this:
> outside customer calls a phone number, number is busy.
> opensips looks up the customer preference and forwards the busy call 
> to another phone.
> the first (busy) number is behind  a nat.
> the second is not.
> I am using rtpproxy for my media relaying on the nat side.
> The problem is that when the 200 OK response ggets sent from the 
> second phone picking up the call, opensips does not fix the sdp in the 
> message.
> This results in one-way audio on the call.
> I am using opensips-1.5.3
> I am attaching the sip trace for reference.
> The way I do call forwarding is: look for the 486 busy response and 
> then append_branch to the forwarded destination.
> as you can see in the invite, the sdp information is retained, but the 
> system doesn't seem to recognise the ok response as part of that sip 
> transaction.
> Any help or suggestions of where to look would be appreciated.
> Thanks.
>
> -- Kelly Prescott
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Audio problem

[Bogdan-Andrei Iancu]

Hi Justin,

a trace means all SIP messages from that call (not only the INVITE) :).

Also, "audio problem" means there is not audio at all or means you have 
one way audio ?

Regards,
Bogdan

Justin L wrote:
> Here is the INVITE:
>
> INVITE sip:13101234...@ask00-rvn SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 10.1.3.130;branch=z9hG4bK88c5.4ae45bf5.0
> Via: SIP/2.0/UDP 
> 172.16.100.159:21874;received=172.16.100.159;branch=z9hG4bK-d87543-2376e4785757b07b-1--d87543-;rport=21874
> Max-Forwards: 69
> Contact:  >
> To: "13101234567"
> From: "2";tag=c020195b
> Call-ID: NDg4Y2Y0ZWU5MGM4NjhiNWVlZGNiZTc1ZGQxMjlhYzc.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
> SUBSCRIBE, INFO
> Content-Type: application/sdp
> User-Agent: X-Lite release 1011s stamp 41150
> Content-Length: 529
>
> v=0
> o=- 8 2 IN IP4 172.16.100.159
> s=CounterPath X-Lite 3.0
> c=IN IP4 172.16.100.159
> t=0 0
> m=audio 39148 RTP/AVP 107 119 100 106 0 105 98 8 101
> a=alt:1 3 : 5AkMoAfO yRnFlRIn 172.16.100.159 39148
> a=alt:2 2 : 7PbWVKqn VccqHBD1 192.168.2.59 39148
> a=alt:3 1 : TXSbExav /8BXXCL+ 192.168.176.152 39148
> a=fmtp:101 0-15
> a=rtpmap:107 BV32/16000
> a=rtpmap:119 BV32-FEC/16000
> a=rtpmap:100 SPEEX/16000
> a=rtpmap:106 SPEEX-FEC/16000
> a=rtpmap:105 SPEEX-FEC/8000
> a=rtpmap:98 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
>
>
> 2009/10/21 Raúl Alexis Betancor Santana  >
>
> On Wednesday 21 October 2009 23:13:36 Justin L wrote:
> > Hi,
> >
> > I have a question related to my load balancing configuration of
> opensips.
> >
> > I have an X-Lite softphone that connects to Opensips server, which
> > transfers the INVITE request to one of the asterisk boxes.
> > All of them are behind firewall on the same network. Then
> asterisk calls to
> > my cell phone through the voip provider.
> >
> > The SIP balancing works fine and I get the call, but there is no
> audio. The
> > firewall should be configured correctly to transfer the SIP and
> RTP ports.
> >
> > Since I just started to use opensips it sounds to me like a very
> basic
> > problem, that many people probably have faced.
> > Could you please recommend me a  way to troubleshoot this issue?
> >
> > Thanks a lot,
> >
> > Justin.
>
> Some SIP trace would be nice to begin ...
>
> --
> Raúl Alexis Betancor Santana
> Dimensión Virtual
>
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] 1.6 core dump uac_replace_to and uac_replace_from

[Bogdan-Andrei Iancu]

Hi Brad,

Do you use both replace_to and replace_from in the same time ?

Do you still have the core file ? could you check in frame 0 for 
"request"  and  "request->dst_uri" ?

Thanks and regards,
Bogdan

Brad Bendy wrote:
> Hi,
>
> Running 1.6 non SVN we are getting random crashes, it appears this is 
> from the uac_replace_from and uac_replace_to, we did not have this 
> problem until we started using these function. Below is the bt from gdb.
>
> The only errors we get in the logs are from memcache, which I think are 
> unrelated from this, has anyone else seen this issue? I can try the 
> latest SVN tonight and see how if that fixes it.
>
> Thanks for any help or suggestions.
>
> #0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
> branch=1, request=0x801647020) at t_fwd.c:132
> 132 memcpy( p, request->dst_uri.s, 
> request->dst_uri.len);
>
>
>
> #0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
> branch=1, request=0x801647020) at t_fwd.c:132
> #1  0x000801518092 in add_uac (t=0x802a18bf8, request=0x801647020, 
> uri=0x7fffd0d0,
> next_hop=0x7fffd0e0, path=Variable "path" is not available.
> ) at t_fwd.c:400
> #2  0x000801519bda in t_forward_nonack (t=0x802a18bf8, 
> p_msg=0x801647020, proxy=0x0) at t_fwd.c:625
> #3  0x000801531d1e in w_t_relay (p_msg=0x801647020, proxy=0x0, 
> flags=Variable "flags" is not available.
> ) at tm.c:1101
> #4  0x0040d9d1 in do_action (a=0x6c7a78, msg=0x801647020) at 
> action.c:962
> #5  0x0040c707 in run_action_list (a=Variable "a" is not available.
> ) at action.c:139
> #6  0x00465b69 in eval_expr (e=0x6c7b48, msg=0x801647020, 
> val=0x0) at route.c:1240
> #7  0x0046584e in eval_expr (e=0x6c7b90, msg=0x801647020, 
> val=0x0) at route.c:1553
> #8  0x00465869 in eval_expr (e=0x6c7bd8, msg=0x801647020, 
> val=0x0) at route.c:1558
> #9  0x0040e0c7 in do_action (a=0x6c7d80, msg=0x801647020) at 
> action.c:689
> #10 0x0040c707 in run_action_list (a=Variable "a" is not available.
> ) at action.c:139
> #11 0x0040eee5 in do_action (a=0x6f2310, msg=0x801647020) at 
> action.c:119
> #12 0x0040c707 in run_action_list (a=Variable "a" is not available.
> ) at action.c:139
> #13 0x0040fc6c in do_action (a=0x6f24b0, msg=0x801647020) at 
> action.c:706
> #14 0x0040c707 in run_action_list (a=Variable "a" is not available.
> ) at action.c:139
> #15 0x00410ce8 in run_top_route (a=0x6f1d28, msg=0x801647020) at 
> action.c:119
> #16 0x000801529a4d in t_should_relay_response (Trans=0x802a18bf8, 
> new_code=Variable "new_code" is not available.
> ) at t_reply.c:612
> #17 0x00080152b5f6 in relay_reply (t=0x802a18bf8, p_msg=0x71f3b0, 
> branch=0, msg_status=503, cancel_bitmap=dwarf2_read_address: Corrupted 
> DWARF expression.
> )
> at t_reply.c:1124
> #18 0x00080152cc76 in reply_received (p_msg=0x71f3b0) at t_reply.c:1493
> #19 0x00420d77 in forward_reply (msg=0x71f3b0) at forward.c:559
> #20 0x004551d5 in receive_msg (
> buf=0x659e40 "SIP/2.0 503 Service Unavailable\r\nVia: SIP/2.0/UDP 
> 72.44.195.164:5060;branch=z9hG4bK8ac3.4b441ca2.0\r\nVia: SIP/2.0/UDP 
> 68.165.121.51:5060;branch=z9hG4bK8ac3.9b5220d1.1\r\nVia: SIP/2.0/UDP 
> xx.xx.xx.xx:50"...,
> len=665, rcv_info=0x7fffeac0) at receive.c:200
> #21 0x00499297 in udp_rcv_loop () at udp_server.c:492
> #22 0x0042865a in main (argc=3, argv=Variable "argv" is not 
> available.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] 1.6 core dump uac_replace_to and uac_replace_from

[Brad Bendy]

Hi Bogdan,

Well, sometimes we do, sometimes we only call or the other, 90% of the
time we use both on the same call.

I do still have the core, inside of gdb do I need to run a command?

When I run "bt", frame 0 shows: #0  0x000801517b39 in
pre_print_uac_request (t=0x802976678, branch=1, request=0x801647020) at
t_fwd.c:132

Thanks

Bogdan-Andrei Iancu wrote:

  Hi Brad,

Do you use both replace_to and replace_from in the same time ?

Do you still have the core file ? could you check in frame 0 for 
"request"  and  "request->dst_uri" ?

Thanks and regards,
Bogdan

Brad Bendy wrote:
  
  
Hi,

Running 1.6 non SVN we are getting random crashes, it appears this is 
from the uac_replace_from and uac_replace_to, we did not have this 
problem until we started using these function. Below is the bt from gdb.

The only errors we get in the logs are from memcache, which I think are 
unrelated from this, has anyone else seen this issue? I can try the 
latest SVN tonight and see how if that fixes it.

Thanks for any help or suggestions.

#0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
branch=1, request=0x801647020) at t_fwd.c:132
132 memcpy( p, request->dst_uri.s, 
request->dst_uri.len);



#0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
branch=1, request=0x801647020) at t_fwd.c:132
#1  0x000801518092 in add_uac (t=0x802a18bf8, request=0x801647020, 
uri=0x7fffd0d0,
next_hop=0x7fffd0e0, path=Variable "path" is not available.
) at t_fwd.c:400
#2  0x000801519bda in t_forward_nonack (t=0x802a18bf8, 
p_msg=0x801647020, proxy=0x0) at t_fwd.c:625
#3  0x000801531d1e in w_t_relay (p_msg=0x801647020, proxy=0x0, 
flags=Variable "flags" is not available.
) at tm.c:1101
#4  0x0040d9d1 in do_action (a=0x6c7a78, msg=0x801647020) at 
action.c:962
#5  0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#6  0x00465b69 in eval_expr (e=0x6c7b48, msg=0x801647020, 
val=0x0) at route.c:1240
#7  0x0046584e in eval_expr (e=0x6c7b90, msg=0x801647020, 
val=0x0) at route.c:1553
#8  0x00465869 in eval_expr (e=0x6c7bd8, msg=0x801647020, 
val=0x0) at route.c:1558
#9  0x0040e0c7 in do_action (a=0x6c7d80, msg=0x801647020) at 
action.c:689
#10 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#11 0x0040eee5 in do_action (a=0x6f2310, msg=0x801647020) at 
action.c:119
#12 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#13 0x0040fc6c in do_action (a=0x6f24b0, msg=0x801647020) at 
action.c:706
#14 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#15 0x00410ce8 in run_top_route (a=0x6f1d28, msg=0x801647020) at 
action.c:119
#16 0x000801529a4d in t_should_relay_response (Trans=0x802a18bf8, 
new_code=Variable "new_code" is not available.
) at t_reply.c:612
#17 0x00080152b5f6 in relay_reply (t=0x802a18bf8, p_msg=0x71f3b0, 
branch=0, msg_status=503, cancel_bitmap=dwarf2_read_address: Corrupted 
DWARF _expression_.
)
at t_reply.c:1124
#18 0x00080152cc76 in reply_received (p_msg=0x71f3b0) at t_reply.c:1493
#19 0x00420d77 in forward_reply (msg=0x71f3b0) at forward.c:559
#20 0x004551d5 in receive_msg (
buf=0x659e40 "SIP/2.0 503 Service Unavailable\r\nVia: SIP/2.0/UDP 
72.44.195.164:5060;branch=z9hG4bK8ac3.4b441ca2.0\r\nVia: SIP/2.0/UDP 
68.165.121.51:5060;branch=z9hG4bK8ac3.9b5220d1.1\r\nVia: SIP/2.0/UDP 
xx.xx.xx.xx:50"...,
len=665, rcv_info=0x7fffeac0) at receive.c:200
#21 0x00499297 in udp_rcv_loop () at udp_server.c:492
#22 0x0042865a in main (argc=3, argv=Variable "argv" is not 
available.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

  

  
  

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
  





___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] 1.6 core dump uac_replace_to and uac_replace_from

[Bogdan-Andrei Iancu]

Brad,

in gdb, just do:

f 0
print request
print request->dst_uri

Regards,
Bogdan

Brad Bendy wrote:
> Hi Bogdan,
>
> Well, sometimes we do, sometimes we only call or the other, 90% of the 
> time we use both on the same call.
>
> I do still have the core, inside of gdb do I need to run a command?
>
> When I run "bt", frame 0 shows: #0  0x000801517b39 in 
> pre_print_uac_request (t=0x802976678, branch=1, request=0x801647020) 
> at t_fwd.c:132
>
> Thanks
>
> Bogdan-Andrei Iancu wrote:
>> Hi Brad,
>>
>> Do you use both replace_to and replace_from in the same time ?
>>
>> Do you still have the core file ? could you check in frame 0 for 
>> "request"  and  "request->dst_uri" ?
>>
>> Thanks and regards,
>> Bogdan
>>
>> Brad Bendy wrote:
>>   
>>> Hi,
>>>
>>> Running 1.6 non SVN we are getting random crashes, it appears this is 
>>> from the uac_replace_from and uac_replace_to, we did not have this 
>>> problem until we started using these function. Below is the bt from gdb.
>>>
>>> The only errors we get in the logs are from memcache, which I think are 
>>> unrelated from this, has anyone else seen this issue? I can try the 
>>> latest SVN tonight and see how if that fixes it.
>>>
>>> Thanks for any help or suggestions.
>>>
>>> #0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
>>> branch=1, request=0x801647020) at t_fwd.c:132
>>> 132 memcpy( p, request->dst_uri.s, 
>>> request->dst_uri.len);
>>>
>>>
>>>
>>> #0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
>>> branch=1, request=0x801647020) at t_fwd.c:132
>>> #1  0x000801518092 in add_uac (t=0x802a18bf8, request=0x801647020, 
>>> uri=0x7fffd0d0,
>>> next_hop=0x7fffd0e0, path=Variable "path" is not available.
>>> ) at t_fwd.c:400
>>> #2  0x000801519bda in t_forward_nonack (t=0x802a18bf8, 
>>> p_msg=0x801647020, proxy=0x0) at t_fwd.c:625
>>> #3  0x000801531d1e in w_t_relay (p_msg=0x801647020, proxy=0x0, 
>>> flags=Variable "flags" is not available.
>>> ) at tm.c:1101
>>> #4  0x0040d9d1 in do_action (a=0x6c7a78, msg=0x801647020) at 
>>> action.c:962
>>> #5  0x0040c707 in run_action_list (a=Variable "a" is not available.
>>> ) at action.c:139
>>> #6  0x00465b69 in eval_expr (e=0x6c7b48, msg=0x801647020, 
>>> val=0x0) at route.c:1240
>>> #7  0x0046584e in eval_expr (e=0x6c7b90, msg=0x801647020, 
>>> val=0x0) at route.c:1553
>>> #8  0x00465869 in eval_expr (e=0x6c7bd8, msg=0x801647020, 
>>> val=0x0) at route.c:1558
>>> #9  0x0040e0c7 in do_action (a=0x6c7d80, msg=0x801647020) at 
>>> action.c:689
>>> #10 0x0040c707 in run_action_list (a=Variable "a" is not available.
>>> ) at action.c:139
>>> #11 0x0040eee5 in do_action (a=0x6f2310, msg=0x801647020) at 
>>> action.c:119
>>> #12 0x0040c707 in run_action_list (a=Variable "a" is not available.
>>> ) at action.c:139
>>> #13 0x0040fc6c in do_action (a=0x6f24b0, msg=0x801647020) at 
>>> action.c:706
>>> #14 0x0040c707 in run_action_list (a=Variable "a" is not available.
>>> ) at action.c:139
>>> #15 0x00410ce8 in run_top_route (a=0x6f1d28, msg=0x801647020) at 
>>> action.c:119
>>> #16 0x000801529a4d in t_should_relay_response (Trans=0x802a18bf8, 
>>> new_code=Variable "new_code" is not available.
>>> ) at t_reply.c:612
>>> #17 0x00080152b5f6 in relay_reply (t=0x802a18bf8, p_msg=0x71f3b0, 
>>> branch=0, msg_status=503, cancel_bitmap=dwarf2_read_address: Corrupted 
>>> DWARF expression.
>>> )
>>> at t_reply.c:1124
>>> #18 0x00080152cc76 in reply_received (p_msg=0x71f3b0) at t_reply.c:1493
>>> #19 0x00420d77 in forward_reply (msg=0x71f3b0) at forward.c:559
>>> #20 0x004551d5 in receive_msg (
>>> buf=0x659e40 "SIP/2.0 503 Service Unavailable\r\nVia: SIP/2.0/UDP 
>>> 72.44.195.164:5060;branch=z9hG4bK8ac3.4b441ca2.0\r\nVia: SIP/2.0/UDP 
>>> 68.165.121.51:5060;branch=z9hG4bK8ac3.9b5220d1.1\r\nVia: SIP/2.0/UDP 
>>> xx.xx.xx.xx:50"...,
>>> len=665, rcv_info=0x7fffeac0) at receive.c:200
>>> #21 0x00499297 in udp_rcv_loop () at udp_server.c:492
>>> #22 0x0042865a in main (argc=3, argv=Variable "argv" is not 
>>> available.
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>   
>>> 
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>   
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] 1.6 core dump uac_replace_to and uac_replace_from

[Brad Bendy]

#0  0x000801517b39 in pre_print_uac_request (t=0x802976678,
branch=1, request=0x801647020) at t_fwd.c:132
132 memcpy( p, request->dst_uri.s,
request->dst_uri.len);


$2 = (struct sip_msg *) 0x801647020
(gdb) print request->dst_uri
$3 = {s = 0x0, len = -1}

I see len = -1, so basically a value must be null when im using one of
these functions? From the logging ive had setup I have seen no empty or
weird values, but I can check again as well.

Thanks

That's what I got back.

Thanks

Bogdan-Andrei Iancu wrote:

  Brad,

in gdb, just do:

f 0
print request
print request->dst_uri

Regards,
Bogdan

Brad Bendy wrote:
  
  
Hi Bogdan,

Well, sometimes we do, sometimes we only call or the other, 90% of the 
time we use both on the same call.

I do still have the core, inside of gdb do I need to run a command?

When I run "bt", frame 0 shows: #0  0x000801517b39 in 
pre_print_uac_request (t=0x802976678, branch=1, request=0x801647020) 
at t_fwd.c:132

Thanks

Bogdan-Andrei Iancu wrote:


  Hi Brad,

Do you use both replace_to and replace_from in the same time ?

Do you still have the core file ? could you check in frame 0 for 
"request"  and  "request->dst_uri" ?

Thanks and regards,
Bogdan

Brad Bendy wrote:
  
  
  
Hi,

Running 1.6 non SVN we are getting random crashes, it appears this is 
from the uac_replace_from and uac_replace_to, we did not have this 
problem until we started using these function. Below is the bt from gdb.

The only errors we get in the logs are from memcache, which I think are 
unrelated from this, has anyone else seen this issue? I can try the 
latest SVN tonight and see how if that fixes it.

Thanks for any help or suggestions.

#0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
branch=1, request=0x801647020) at t_fwd.c:132
132 memcpy( p, request->dst_uri.s, 
request->dst_uri.len);



#0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
branch=1, request=0x801647020) at t_fwd.c:132
#1  0x000801518092 in add_uac (t=0x802a18bf8, request=0x801647020, 
uri=0x7fffd0d0,
next_hop=0x7fffd0e0, path=Variable "path" is not available.
) at t_fwd.c:400
#2  0x000801519bda in t_forward_nonack (t=0x802a18bf8, 
p_msg=0x801647020, proxy=0x0) at t_fwd.c:625
#3  0x000801531d1e in w_t_relay (p_msg=0x801647020, proxy=0x0, 
flags=Variable "flags" is not available.
) at tm.c:1101
#4  0x0040d9d1 in do_action (a=0x6c7a78, msg=0x801647020) at 
action.c:962
#5  0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#6  0x00465b69 in eval_expr (e=0x6c7b48, msg=0x801647020, 
val=0x0) at route.c:1240
#7  0x0046584e in eval_expr (e=0x6c7b90, msg=0x801647020, 
val=0x0) at route.c:1553
#8  0x00465869 in eval_expr (e=0x6c7bd8, msg=0x801647020, 
val=0x0) at route.c:1558
#9  0x0040e0c7 in do_action (a=0x6c7d80, msg=0x801647020) at 
action.c:689
#10 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#11 0x0040eee5 in do_action (a=0x6f2310, msg=0x801647020) at 
action.c:119
#12 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#13 0x0040fc6c in do_action (a=0x6f24b0, msg=0x801647020) at 
action.c:706
#14 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#15 0x00410ce8 in run_top_route (a=0x6f1d28, msg=0x801647020) at 
action.c:119
#16 0x000801529a4d in t_should_relay_response (Trans=0x802a18bf8, 
new_code=Variable "new_code" is not available.
) at t_reply.c:612
#17 0x00080152b5f6 in relay_reply (t=0x802a18bf8, p_msg=0x71f3b0, 
branch=0, msg_status=503, cancel_bitmap=dwarf2_read_address: Corrupted 
DWARF _expression_.
)
at t_reply.c:1124
#18 0x00080152cc76 in reply_received (p_msg=0x71f3b0) at t_reply.c:1493
#19 0x00420d77 in forward_reply (msg=0x71f3b0) at forward.c:559
#20 0x004551d5 in receive_msg (
buf=0x659e40 "SIP/2.0 503 Service Unavailable\r\nVia: SIP/2.0/UDP 
72.44.195.164:5060;branch=z9hG4bK8ac3.4b441ca2.0\r\nVia: SIP/2.0/UDP 
68.165.121.51:5060;branch=z9hG4bK8ac3.9b5220d1.1\r\nVia: SIP/2.0/UDP 
xx.xx.xx.xx:50"...,
len=665, rcv_info=0x7fffeac0) at receive.c:200
#21 0x00499297 in udp_rcv_loop () at udp_server.c:492
#22 0x0042865a in main (argc=3, argv=Variable "argv" is not 
available.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

  


  
  
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
  
  



___
Users mailing list
Users@lists.opensips.org
http://l

Re: [OpenSIPS-Users] 1.6 core dump uac_replace_to and uac_replace_from

[Bogdan-Andrei Iancu]

Brad,

Please apply the attached patch and see if it's fixing the problem for you.

Regards,
Bogdan

Brad Bendy wrote:
#0  0x000801517b39 in pre_print_uac_request (t=0x802976678, 
branch=1, request=0x801647020) at t_fwd.c:132
132 memcpy( p, request->dst_uri.s, 
request->dst_uri.len);



$2 = (struct sip_msg *) 0x801647020
(gdb) print request->dst_uri
$3 = {s = 0x0, len = -1}

I see len = -1, so basically a value must be null when im using one of 
these functions? From the logging ive had setup I have seen no empty 
or weird values, but I can check again as well.


Thanks

That's what I got back.

Thanks

Bogdan-Andrei Iancu wrote:

Brad,

in gdb, just do:

f 0
print request
print request->dst_uri

Regards,
Bogdan

Brad Bendy wrote:
  

Hi Bogdan,

Well, sometimes we do, sometimes we only call or the other, 90% of the 
time we use both on the same call.


I do still have the core, inside of gdb do I need to run a command?

When I run "bt", frame 0 shows: #0  0x000801517b39 in 
pre_print_uac_request (t=0x802976678, branch=1, request=0x801647020) 
at t_fwd.c:132


Thanks

Bogdan-Andrei Iancu wrote:


Hi Brad,

Do you use both replace_to and replace_from in the same time ?

Do you still have the core file ? could you check in frame 0 for 
"request"  and  "request->dst_uri" ?


Thanks and regards,
Bogdan

Brad Bendy wrote:
  
  

Hi,

Running 1.6 non SVN we are getting random crashes, it appears this is 
from the uac_replace_from and uac_replace_to, we did not have this 
problem until we started using these function. Below is the bt from gdb.


The only errors we get in the logs are from memcache, which I think are 
unrelated from this, has anyone else seen this issue? I can try the 
latest SVN tonight and see how if that fixes it.


Thanks for any help or suggestions.

#0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
branch=1, request=0x801647020) at t_fwd.c:132
132 memcpy( p, request->dst_uri.s, 
request->dst_uri.len);




#0  0x000801517b39 in pre_print_uac_request (t=0x802a18bf8, 
branch=1, request=0x801647020) at t_fwd.c:132
#1  0x000801518092 in add_uac (t=0x802a18bf8, request=0x801647020, 
uri=0x7fffd0d0,

next_hop=0x7fffd0e0, path=Variable "path" is not available.
) at t_fwd.c:400
#2  0x000801519bda in t_forward_nonack (t=0x802a18bf8, 
p_msg=0x801647020, proxy=0x0) at t_fwd.c:625
#3  0x000801531d1e in w_t_relay (p_msg=0x801647020, proxy=0x0, 
flags=Variable "flags" is not available.

) at tm.c:1101
#4  0x0040d9d1 in do_action (a=0x6c7a78, msg=0x801647020) at 
action.c:962

#5  0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#6  0x00465b69 in eval_expr (e=0x6c7b48, msg=0x801647020, 
val=0x0) at route.c:1240
#7  0x0046584e in eval_expr (e=0x6c7b90, msg=0x801647020, 
val=0x0) at route.c:1553
#8  0x00465869 in eval_expr (e=0x6c7bd8, msg=0x801647020, 
val=0x0) at route.c:1558
#9  0x0040e0c7 in do_action (a=0x6c7d80, msg=0x801647020) at 
action.c:689

#10 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#11 0x0040eee5 in do_action (a=0x6f2310, msg=0x801647020) at 
action.c:119

#12 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#13 0x0040fc6c in do_action (a=0x6f24b0, msg=0x801647020) at 
action.c:706

#14 0x0040c707 in run_action_list (a=Variable "a" is not available.
) at action.c:139
#15 0x00410ce8 in run_top_route (a=0x6f1d28, msg=0x801647020) at 
action.c:119
#16 0x000801529a4d in t_should_relay_response (Trans=0x802a18bf8, 
new_code=Variable "new_code" is not available.

) at t_reply.c:612
#17 0x00080152b5f6 in relay_reply (t=0x802a18bf8, p_msg=0x71f3b0, 
branch=0, msg_status=503, cancel_bitmap=dwarf2_read_address: Corrupted 
DWARF expression.

)
at t_reply.c:1124
#18 0x00080152cc76 in reply_received (p_msg=0x71f3b0) at t_reply.c:1493
#19 0x00420d77 in forward_reply (msg=0x71f3b0) at forward.c:559
#20 0x004551d5 in receive_msg (
buf=0x659e40 "SIP/2.0 503 Service Unavailable\r\nVia: SIP/2.0/UDP 
72.44.195.164:5060;branch=z9hG4bK8ac3.4b441ca2.0\r\nVia: SIP/2.0/UDP 
68.165.121.51:5060;branch=z9hG4bK8ac3.9b5220d1.1\r\nVia: SIP/2.0/UDP 
xx.xx.xx.xx:50"...,

len=665, rcv_info=0x7fffeac0) at receive.c:200
#21 0x00499297 in udp_rcv_loop () at udp_server.c:492
#22 0x0042865a in main (argc=3, argv=Variable "argv" is not 
available.
  


Index: modules/tm/t_fwd.c
===
--- modules/tm/t_fwd.c	(revision 6272)
+++ modules/tm/t_fwd.c	(working copy)
@@ -123,7 +123,7 @@
 	 * to allow to be changed --bogdan */
 	if (t->on_branch) {
 		/* need to pkg_malloc the dst_uri */
-		if ( request->dst_uri.len ) {
+		if ( request->dst_uri.s && request->dst_uri.len>0 ) {
 			if ( (p=pkg_malloc(re

Re: [OpenSIPS-Users] Audio problem

[Justin L]

Thanks for reply.

Actually I already solved the problem. It was an asterisk configuration
issue.
Thanks anyway.

Justin.

On Wed, Oct 21, 2009 at 9:52 PM, Bogdan-Andrei Iancu  wrote:

> Hi Justin,
>
> a trace means all SIP messages from that call (not only the INVITE) :).
>
> Also, "audio problem" means there is not audio at all or means you have
> one way audio ?
>
> Regards,
> Bogdan
>
> Justin L wrote:
> > Here is the INVITE:
> >
> > INVITE sip:13101234...@ask00-rvn SIP/2.0
> > Record-Route: 
> > Via: SIP/2.0/UDP 10.1.3.130;branch=z9hG4bK88c5.4ae45bf5.0
> > Via: SIP/2.0/UDP
> > 172.16.100.159:21874
> ;received=172.16.100.159;branch=z9hG4bK-d87543-2376e4785757b07b-1--d87543-;rport=21874
> > Max-Forwards: 69
> > Contact:  > >
> > To: "13101234567"
> > From: "2";tag=c020195b
> > Call-ID: NDg4Y2Y0ZWU5MGM4NjhiNWVlZGNiZTc1ZGQxMjlhYzc.
> > CSeq: 1 INVITE
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> > SUBSCRIBE, INFO
> > Content-Type: application/sdp
> > User-Agent: X-Lite release 1011s stamp 41150
> > Content-Length: 529
> >
> > v=0
> > o=- 8 2 IN IP4 172.16.100.159
> > s=CounterPath X-Lite 3.0
> > c=IN IP4 172.16.100.159
> > t=0 0
> > m=audio 39148 RTP/AVP 107 119 100 106 0 105 98 8 101
> > a=alt:1 3 : 5AkMoAfO yRnFlRIn 172.16.100.159 39148
> > a=alt:2 2 : 7PbWVKqn VccqHBD1 192.168.2.59 39148
> > a=alt:3 1 : TXSbExav /8BXXCL+ 192.168.176.152 39148
> > a=fmtp:101 0-15
> > a=rtpmap:107 BV32/16000
> > a=rtpmap:119 BV32-FEC/16000
> > a=rtpmap:100 SPEEX/16000
> > a=rtpmap:106 SPEEX-FEC/16000
> > a=rtpmap:105 SPEEX-FEC/8000
> > a=rtpmap:98 iLBC/8000
> > a=rtpmap:101 telephone-event/8000
> > a=sendrecv
> >
> >
> > 2009/10/21 Raúl Alexis Betancor Santana  > >
> >
> > On Wednesday 21 October 2009 23:13:36 Justin L wrote:
> > > Hi,
> > >
> > > I have a question related to my load balancing configuration of
> > opensips.
> > >
> > > I have an X-Lite softphone that connects to Opensips server, which
> > > transfers the INVITE request to one of the asterisk boxes.
> > > All of them are behind firewall on the same network. Then
> > asterisk calls to
> > > my cell phone through the voip provider.
> > >
> > > The SIP balancing works fine and I get the call, but there is no
> > audio. The
> > > firewall should be configured correctly to transfer the SIP and
> > RTP ports.
> > >
> > > Since I just started to use opensips it sounds to me like a very
> > basic
> > > problem, that many people probably have faced.
> > > Could you please recommend me a  way to troubleshoot this issue?
> > >
> > > Thanks a lot,
> > >
> > > Justin.
> >
> > Some SIP trace would be nice to begin ...
> >
> > --
> > Raúl Alexis Betancor Santana
> > Dimensión Virtual
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE

[Uwe Kastens]

Hi Borgan,

Sorry by trying to debug the problem I understood the hole picture. I
think it might be a bug or a feature request for the tm module.

The setup is:

PSTN-GW <-> opensips as statefull proxy <-> AST1 + AST2

If I make a call from pstn over the opensips to a specific SIP-URI, the
call will be forked parallel to AST1 and AST2. This is done statefull
via tm (relay). AST1 will send a 200 OK with SDP, tm will generate a
CANCEL message for the 2nd branch to AST2. AST2 has already sent a 200
OK with SDP and will therefore send 200 OK for the CANCEL request.

The problem is, that both 200 OK with SDP are sent back to the PSTN GW,
 which has ACKed one call already and will get a 2nd 200 OK with the
same branch but different call-id. This is ignored because the PSTN-GW
is not aware about branches/call-id.

So there are 2 possible solutions:

- The PSTN GW needs to send a BYE to the branch which comes later on
with the same branch but different call-id.
- opensips TM should send a bye if the CANCEL if the call is forked
parallel and the CANCEL Message is answered with a 200 OK.

I know there is a lot of discussion about this issue, but I need a
solution

BR

Uwe



Bogdan-Andrei Iancu schrieb:
> Hi Uwe,
> 
> as I understand from you, from end devices (GW, as1 and as2) everything 
> work ok, but the dialog state on opensips is not properly kept??
> 
> Regards,
> Bogdan
> 
> Uwe Kastens wrote:
>> Hello Bogdan,
>>
>> Now we changed the behaviour of the UAC. One of them will send a BYE and
>> this is relayed to the PSTN GW which drops the call, since opensips will
>> not handle the BYE locally. So loose_route is done and the BYE is
>> relayed to the PSTN GW.
>>
>> The following is happening:
>>
>> 1) INVITE from PSTN GW
>> 2) parallel forking to ast1 and ast2 (branches z9hG4bK51f6.9afa91c3.1
>> and z9hG4bK51f6.9afa91c3.0)
>> 3) ast1 sends an 200 OK (branch z9hG4bK51f6.9afa91c3.0)
>> 4) opensips sends an cancel to ast2 (branch z9hG4bK51f6.9afa91c3.1)
>> 5) opensips receives the 200 OK from ast1 and sends an ACK (branch is
>> changing here to z9hG4bK51f6.9afa91c3.3)
>> 6) opensips receives 200 OK from ast2 from the INVITE (branch
>> z9hG4bK51f6.9afa91c3.1)and sends an ACK (branch is changing to
>> z9hG4bK51f6.9afa91c3.3)
>> 7) opensips reives 200 OK from ast2 for the cancel request ( branch
>> z9hG4bK51f6.9afa91c3.1)
>> 8) opensips receives BYE from ast2 with branch z9hG4bK40d1af5d
>> 9) opensips is doing loose_route and sends the BYE to the PSTN GW
>>
>>
>> The only thing I could see on the logs is:
>>
>>  WARNING:dialog:dlg_onroute: tight matching failed for BYE with
>> callid='393105a419950c1f265f298914662...@10.20.30.100'/46,
>> ftag='as63949c6e'/10, ttag='as0d1597ca'/10 and direction=0
>> Oct 21 09:09:15 asne02 /usr/sbin/opensips[15615]:
>> WARNING:dialog:dlg_onroute: dialog identification elements are
>> callid='393105a419950c1f265f298914662...@10.20.30.100'/46, caller
>> tag='as0d1597ca'/10, callee tag='as79debd51'/10
>>
>> Why is the opensips not handling the BYE locally and only closing one
>> branch?
>>
>> BR
>>
>> UWe
>>
>>
>> Bogdan-Andrei Iancu schrieb:
>>   
>>> Hi Uwe,
>>>
>>> Uwe Kastens wrote:
>>> 
 Hi Bogdan,

   
   
> So actually both legs do send 200 OK (but one faster than the 
> other)..so there is kind on race between the 200 OK from the slow 
> branch and the CANCEL from OpenSIPS...is this the case?
> 
> 
 Exactly

   
   
> If so, the UAS will simply reply with negative reply to CANCEL (decline 
> it) and opensips (for INVITE transaction) will not close the second 
> branch as there is a 200 OK (and not a 487) received RFC3261 says 
> that a proxy must send all 200 OK (for a call), even if more than one, 
> to the UAC - the UAC is the one who will decide what branch to keep and 
> it will fire a BYE for the other branch.
>
> 
> 
 Could this explan, why only the 2nd Node will get the BYE, if the call
 is released "behind" the opensips?
   
   
>>> yes, because the caller will hung up only one of the callee branch, so 
>>> the BYE will go to only one of them. The other branch will remain up and 
>>> will be the ongoing call.
>>>
>>> Regards,
>>> Bogdan
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> 
>>
>>   
> 
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


-- 

kiste lat: 54.322684, lon: 10.13586

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users