date:20130619

Re: [OpenSIPS-Users] [Bulk] Re: Setting up opensips

2013-06-19 Thread Bogdan-Andrei Iancu

Hello Stuart,

The USRLOC module is not used and loaded by the LB scenarios, so simply
ignore that.

Regards

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/19/2013 05:10 PM, Stuart Mills wrote:
> Hi Bogdan,
>
> I figured it out and it's now working thanks.
>
> I am however seeing UsrLoc Stats: 404 statistics not found. 
>
> Would you have any idea why that might be? I am using a generated load
> balance script.
>
> Cheers,
>
> Stuart
>
>
> Sent from my iPhone
>
> On 19 Jun 2013, at 14:28, Bogdan-Andrei Iancu  > wrote:
>
>> Hello Stuart,
>>
>> Maybe your calls are negatively replied by OpenSIPS ? are you doing
>> any t_relay() after the LB part ?
>>
>> It seems a script error to me.
>>
>> Regards,
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>> On 06/18/2013 11:56 AM, Stuart Mills wrote:
>>> Hi,
>>>  
>>> I wonder if anyone can help me.
>>>  
>>> I’ve setup OpenSIPS 1.9.1-tls stable on Debian 7, I have installed
>>> the mysql module, create the database schema and gone through make
>>> menuselect to create a load balance script, saved and generated the
>>> script, all seemed to go according to plan, after that I created
>>> some load balance nodes using some examples as a guide.
>>>  
>>> The problem I seem to be having is when the calls arrive they show
>>> up on the UAS_transactions count but the call doesn’t seem to be
>>> passed onto any of the load balance nodes.
>>>  
>>> Is there a way i can see what OpenSIPS is trying to do with the call?
>>>  
>>> Regards,
>>>  
>>> Stuart Mills
>>>  
>>>  
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread M.Khaled W Chehab

Thanks man fixed J 

 

Please can you send me the link that I can do my queries on J as I am always 
using google ,or it’s a google search 

 

 

Thanks for help , Appreciated 

 

Regards



 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Laszlo
Sent: Wednesday, June 19, 2013 5:07 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] uac_replace_to problem

 

Cool!

If you checked the link what I have sent, then you can easily see the logic how 
it can be implemented.

-Laszlo

 

2013/6/19 M.Khaled W Chehab 

Hi Laszlo 

 

I am searching since yesterday,as I inform you that I use uac_restore_to() and 
I set iuac_replace_to   in different places  and in different shapes

, 

Today afternoon I ask this question in the users list since most of google seah 
result links marked as read J

At all thanks 

Sure I will keep searching ,,,

regards

 

 

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Laszlo
Sent: Wednesday, June 19, 2013 4:50 PM


To: OpenSIPS users mailling list

Subject: Re: [OpenSIPS-Users] uac_replace_to problem

 

Yes, but why don't you try to search for the answer first? This problem was 
discussed several times before.

By doing searching, everybody can see that you make some steps to resolve the 
problem by yourself, not just asking for the "solution".

A quick search for the problem will show this result too:

http://lists.opensips.org/pipermail/users/2010-June/013200.html

Took 40 seconds to find.

-Laszlo

 

 

2013/6/19 M.Khaled W Chehab 

Please can you show me what do you mean by coding it ,what do you mean branch 
route  for example is it route (6) I am using , you can find my config below 

 

while i am using uac_replace_to in failover route branch i can find that TO 
header is not changed(sip user part ) but appended an new raw  as I want it to 
be To: "971552448304" 

 

SIP to address: 
sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55

SIP to address User Part: 835822971552448304

SIP to address Host Part: xx.xx.xx.55sip

SIP to address Host Port: 808971552448...@xx.xx.xx.55

 

 

Thanks in advance 

 

regards

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Flavio Goncalves
Sent: Wednesday, June 19, 2013 3:13 PM
To: OpenSIPS users mailling list
Cc: users-boun...@lists.opensips.org
Subject: Re: [OpenSIPS-Users] uac_replace_to problem

 

Hi Khaleb, 

 

Set uac_replace_to and uac_replace_from in a branch_route. 

 

Flavio E. Goncalves

 

 

2013/6/19 M.Khaled W Chehab 

Hi,

 

I am running opensips 1.8.3 with  do_routing module 

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

 

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using uac_replace_to( and 
the call go to gw1 with correct TO tag as I set it ,

 

but when calls fails on gw1 ,then  I set the $rU in route[6] to go to second in 
route and it goes with bad TO header, since it goes with the same To header in 
the 1st invite 

That target gw1 

1-how to fix the To header in the second invite to gw2

 

 

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

 

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");

   t_on_failure("1");

 

on failure_route[1] {

.

if (!t_check_status("487")) {

 #xlog("route6---\n");

 $avp(failure_count) = $avp(failure_count) + 1;  
#480|486|603  

   route(6);

 

} 

 

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td");

 

}

 

Regards

 


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Re: [OpenSIPS-Users] [Bulk] Re: Setting up opensips

2013-06-19 Thread Stuart Mills

Hi Bogdan,

I figured it out and it's now working thanks.

I am however seeing UsrLoc Stats: 404 statistics not found. 

Would you have any idea why that might be? I am using a generated load balance 
script.

Cheers,

Stuart


Sent from my iPhone

On 19 Jun 2013, at 14:28, Bogdan-Andrei Iancu  wrote:

> Hello Stuart,
> 
> Maybe your calls are negatively replied by OpenSIPS ? are you doing any 
> t_relay() after the LB part ?
> 
> It seems a script error to me.
> 
> Regards,
>  Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
> On 06/18/2013 11:56 AM, Stuart Mills wrote:
>> 
>> Hi,
>>  
>> I wonder if anyone can help me.
>>  
>> I’ve setup OpenSIPS 1.9.1-tls stable on Debian 7, I have installed the mysql 
>> module, create the database schema and gone through make menuselect to 
>> create a load balance script, saved and generated the script, all seemed to 
>> go according to plan, after that I created some load balance nodes using 
>> some examples as a guide.
>>  
>> The problem I seem to be having is when the calls arrive they show up on the 
>> UAS_transactions count but the call doesn’t seem to be passed onto any of 
>> the load balance nodes.
>>  
>> Is there a way i can see what OpenSIPS is trying to do with the call?
>>  
>> Regards,
>>  
>> Stuart Mills
>>  
>>  
>> 
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread Laszlo

Cool!
If you checked the link what I have sent, then you can easily see the logic
how it can be implemented.

-Laszlo



2013/6/19 M.Khaled W Chehab 

> Hi Laszlo 
>
> ** **
>
> I am searching since yesterday,as I inform you that I use uac_restore_to()
> and I set iuac_replace_to   in different places  and in different shapes**
> **
>
> , 
>
> Today afternoon I ask this question in the users list since most of google
> seah result links marked as read J
>
> 
>
> At all thanks 
>
> Sure I will keep searching ,,,
>
> regards
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Laszlo
> *Sent:* Wednesday, June 19, 2013 4:50 PM
>
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] uac_replace_to problem
>
> ** **
>
> Yes, but why don't you try to search for the answer first? This problem
> was discussed several times before.
>
> By doing searching, everybody can see that you make some steps to resolve
> the problem by yourself, not just asking for the "solution".
>
> A quick search for the problem will show this result too:
>
> http://lists.opensips.org/pipermail/users/2010-June/013200.html
>
> Took 40 seconds to find.
>
> -Laszlo
>
> ** **
>
> ** **
>
> 2013/6/19 M.Khaled W Chehab 
>
> Please can you show me what do you mean by coding it ,what do you mean
> branch route  for example is it route (6) I am using , you can find my
> config below 
>
>  
>
> while i am using uac_replace_to in failover route branch i can find that
> TO header is not changed(sip user part ) but appended an new raw  as I want
> it to be To: "971552448304" 
>
>  
>
> SIP to address:
> sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55
>
> SIP to address User Part: 835822971552448304
>
> SIP to address Host Part: xx.xx.xx.55sip
>
> SIP to address Host Port: 808971552448...@xx.xx.xx.55
>
>  
>
>  
>
> Thanks in advance 
>
>  
>
> regards
>
>  
>
>  
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Flavio Goncalves
> *Sent:* Wednesday, June 19, 2013 3:13 PM
> *To:* OpenSIPS users mailling list
> *Cc:* users-boun...@lists.opensips.org
> *Subject:* Re: [OpenSIPS-Users] uac_replace_to problem
>
>  
>
> Hi Khaleb, 
>
>  
>
> Set uac_replace_to and uac_replace_from in a branch_route. 
>
>  
>
> Flavio E. Goncalves
>
>  
>
>  
>
> 2013/6/19 M.Khaled W Chehab 
>
> Hi,
>
>  
>
> I am running opensips 1.8.3 with  do_routing module 
>
> A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)
>
>  
>
> After do_routing(,)
>
> I am setting the $rU and fixing  the To Tag header using uac_replace_to(
> and the call go to gw1 with correct TO tag as I set it ,
>
>  
>
> but when calls fails on gw1 ,then  I set the $rU in route[6] to go to
> second in route and it goes with bad TO header, since it goes with the same
> To header in the 1st invite 
>
> That target gw1 
>
> 1-how to fix the To header in the second invite to gw2
>
>  
>
>  
>
> loadmodule "uac.so"
>
> modparam("uac","restore_mode","auto")
>
>  
>
> uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");
>
>t_on_failure("1");
>
>  
>
> on failure_route[1] {
>
> .
>
> if (!t_check_status("487")) {
>
>  #xlog("route6---\n");
>
>  $avp(failure_count) = $avp(failure_count) +
> 1;  #480|486|603  
>
>route(6);
>
>  
>
> } 
>
>  
>
> Route[6]{
>
> .
>
> .
>
> .
>
> $rU = $var(prefix) + $avp(dst);
>
> uac_replace_to("sip:$var(prefix)$avp(dst)@$td");
>
>  
>
> }
>
>  
>
> Regards
>
>  
>
>
> ___
> 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
>
>
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread Laszlo

This error message can be totally unrelated to the problem you have with
the uac_replace_to thing.
It means that the proxy tried to close a session which doesn't exist
anymore in mediaproxy.

I guess you not using engage_media_proxy() function, or you trying to end
the media session more than once with end_media_session() upon receiving
the BYE.

-Laszlo


2013/6/19 M.Khaled W Chehab 

> Hi Laszlo
>
> ** **
>
> I can see this error in the syslog 
>
> error: Got `remove' command from OpenSIPS for unknown session with call-id
> `ca8ccd95-d81511e2-bebfd582-e4ab...@xx.xx.xx.xx-b2b_
>
> please can you show me how I can do the uac_replace  per branch ,in other
> words from where and how I can remove the uac_replace_to from global since
> second time I am doing it in a branch 
>
> ** **
>
> Regard
>
> ** **
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Laszlo
> *Sent:* Wednesday, June 19, 2013 4:24 PM
>
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] uac_replace_to problem
>
> ** **
>
> BTW, looks like you doing uac_replace_to twice. once in global, and once
> in a per-branch route.
>
> You can do it only once, I'm sure you see something about this in the
> syslog.
>
> try what Flavio suggested, so do these changes only in the branch route.**
> **
>
> -Laszlo
>
> ** **
>
> 2013/6/19 M.Khaled W Chehab 
>
> while i am using uac_replace_to in failover route branch i can find that
> TO header is not changed(sip user part ) but appended an new raw  as I want
> it to be To: "971552448304" 
>
>  
>
> SIP to address:
> sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55
>
> SIP to address User Part: 835822971552448304
>
> SIP to address Host Part: xx.xx.xx.55sip
>
> SIP to address Host Port: 808971552448...@xx.xx.xx.55
>
>  
>
>  
>
> please advice 
>
>  
>
> regards
>
>  
>
>  
>
> *From:* M.Khaled W Chehab [mailto:kche...@icucall.com]
> *Sent:* Wednesday, June 19, 2013 2:41 PM
> *To:* users@lists.opensips.org
> *Cc:* users-boun...@lists.opensips.org
> *Subject:* uac_replace_to problem 
>
>  
>
> Hi,
>
>  
>
> I am running opensips 1.8.3 with  do_routing module 
>
> A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)
>
>  
>
> After do_routing(,)
>
> I am setting the $rU and fixing  the To Tag header using uac_replace_to(
> and the call go to gw1 with correct TO tag as I set it ,
>
>  
>
> but when calls fails on gw1 ,then  I set the $rU in route[6] to go to
> second in route and it goes with bad TO header, since it goes with the same
> To header in the 1st invite 
>
> That target gw1 
>
> 1-how to fix the To header in the second invite to gw2
>
>  
>
>  
>
> loadmodule "uac.so"
>
> modparam("uac","restore_mode","auto")
>
>  
>
> uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");
>
>t_on_failure("1");
>
>  
>
> on failure_route[1] {
>
> .
>
> if (!t_check_status("487")) {
>
>  #xlog("route6---\n");
>
>  $avp(failure_count) = $avp(failure_count) +
> 1;  #480|486|603  
>
>route(6);
>
>  
>
> } 
>
>  
>
> Route[6]{
>
> .
>
> .
>
> .
>
> $rU = $var(prefix) + $avp(dst);
>
> uac_replace_to("sip:$var(prefix)$avp(dst)@$td");
>
>  
>
> }
>
>  
>
> Regards
>
>  
>
>
> ___
> 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
>
>
___
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread M.Khaled W Chehab

Hi Laszlo 

 

I am searching since yesterday,as I inform you that I use uac_restore_to() and 
I set iuac_replace_to   in different places  and in different shapes

, 

Today afternoon I ask this question in the users list since most of google seah 
result links marked as read J

At all thanks 

Sure I will keep searching ,,,

regards

 

 

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Laszlo
Sent: Wednesday, June 19, 2013 4:50 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] uac_replace_to problem

 

Yes, but why don't you try to search for the answer first? This problem was 
discussed several times before.

By doing searching, everybody can see that you make some steps to resolve the 
problem by yourself, not just asking for the "solution".

A quick search for the problem will show this result too:

http://lists.opensips.org/pipermail/users/2010-June/013200.html

Took 40 seconds to find.

-Laszlo

 

 

2013/6/19 M.Khaled W Chehab 

Please can you show me what do you mean by coding it ,what do you mean branch 
route  for example is it route (6) I am using , you can find my config below 

 

while i am using uac_replace_to in failover route branch i can find that TO 
header is not changed(sip user part ) but appended an new raw  as I want it to 
be To: "971552448304" 

 

SIP to address: 
sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55

SIP to address User Part: 835822971552448304

SIP to address Host Part: xx.xx.xx.55sip

SIP to address Host Port: 808971552448...@xx.xx.xx.55

 

 

Thanks in advance 

 

regards

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Flavio Goncalves
Sent: Wednesday, June 19, 2013 3:13 PM
To: OpenSIPS users mailling list
Cc: users-boun...@lists.opensips.org
Subject: Re: [OpenSIPS-Users] uac_replace_to problem

 

Hi Khaleb, 

 

Set uac_replace_to and uac_replace_from in a branch_route. 

 

Flavio E. Goncalves

 

 

2013/6/19 M.Khaled W Chehab 

Hi,

 

I am running opensips 1.8.3 with  do_routing module 

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

 

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using uac_replace_to( and 
the call go to gw1 with correct TO tag as I set it ,

 

but when calls fails on gw1 ,then  I set the $rU in route[6] to go to second in 
route and it goes with bad TO header, since it goes with the same To header in 
the 1st invite 

That target gw1 

1-how to fix the To header in the second invite to gw2

 

 

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

 

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");

   t_on_failure("1");

 

on failure_route[1] {

.

if (!t_check_status("487")) {

 #xlog("route6---\n");

 $avp(failure_count) = $avp(failure_count) + 1;  
#480|486|603  

   route(6);

 

} 

 

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td");

 

}

 

Regards

 


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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread Laszlo

Yes, but why don't you try to search for the answer first? This problem was
discussed several times before.
By doing searching, everybody can see that you make some steps to resolve
the problem by yourself, not just asking for the "solution".

A quick search for the problem will show this result too:

http://lists.opensips.org/pipermail/users/2010-June/013200.html

Took 40 seconds to find.

-Laszlo



2013/6/19 M.Khaled W Chehab 

> Please can you show me what do you mean by coding it ,what do you mean
> branch route  for example is it route (6) I am using , you can find my
> config below 
>
> ** **
>
> while i am using uac_replace_to in failover route branch i can find that
> TO header is not changed(sip user part ) but appended an new raw  as I want
> it to be To: "971552448304" 
>
> ** **
>
> SIP to address:
> sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55
>
> SIP to address User Part: 835822971552448304
>
> SIP to address Host Part: xx.xx.xx.55sip
>
> SIP to address Host Port: 808971552448...@xx.xx.xx.55
>
> ** **
>
> ** **
>
> Thanks in advance 
>
> ** **
>
> regards
>
> ** **
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Flavio Goncalves
> *Sent:* Wednesday, June 19, 2013 3:13 PM
> *To:* OpenSIPS users mailling list
> *Cc:* users-boun...@lists.opensips.org
> *Subject:* Re: [OpenSIPS-Users] uac_replace_to problem
>
> ** **
>
> Hi Khaleb, 
>
> ** **
>
> Set uac_replace_to and uac_replace_from in a branch_route. 
>
> ** **
>
> Flavio E. Goncalves
>
> ** **
>
> ** **
>
> 2013/6/19 M.Khaled W Chehab 
>
> Hi,
>
>  
>
> I am running opensips 1.8.3 with  do_routing module 
>
> A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)
>
>  
>
> After do_routing(,)
>
> I am setting the $rU and fixing  the To Tag header using uac_replace_to(
> and the call go to gw1 with correct TO tag as I set it ,
>
>  
>
> but when calls fails on gw1 ,then  I set the $rU in route[6] to go to
> second in route and it goes with bad TO header, since it goes with the same
> To header in the 1st invite 
>
> That target gw1 
>
> 1-how to fix the To header in the second invite to gw2
>
>  
>
>  
>
> loadmodule "uac.so"
>
> modparam("uac","restore_mode","auto")
>
>  
>
> uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");
>
>t_on_failure("1");
>
>  
>
> on failure_route[1] {
>
> .
>
> if (!t_check_status("487")) {
>
>  #xlog("route6---\n");
>
>  $avp(failure_count) = $avp(failure_count) +
> 1;  #480|486|603  
>
>route(6);
>
>  
>
> } 
>
>  
>
> Route[6]{
>
> .
>
> .
>
> .
>
> $rU = $var(prefix) + $avp(dst);
>
> uac_replace_to("sip:$var(prefix)$avp(dst)@$td");
>
>  
>
> }
>
>  
>
> Regards
>
>  
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ** **
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread M.Khaled W Chehab

Hi Laszlo

 

I can see this error in the syslog 

error: Got `remove' command from OpenSIPS for unknown session with call-id 
`ca8ccd95-d81511e2-bebfd582-e4ab...@xx.xx.xx.xx-b2b_

please can you show me how I can do the uac_replace  per branch ,in other words 
from where and how I can remove the uac_replace_to from global since second 
time I am doing it in a branch 

 

Regard

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Laszlo
Sent: Wednesday, June 19, 2013 4:24 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] uac_replace_to problem

 

BTW, looks like you doing uac_replace_to twice. once in global, and once in a 
per-branch route.

You can do it only once, I'm sure you see something about this in the syslog.

try what Flavio suggested, so do these changes only in the branch route.

-Laszlo

 

2013/6/19 M.Khaled W Chehab 

while i am using uac_replace_to in failover route branch i can find that TO 
header is not changed(sip user part ) but appended an new raw  as I want it to 
be To: "971552448304" 

 

SIP to address: 
sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55

SIP to address User Part: 835822971552448304

SIP to address Host Part: xx.xx.xx.55sip

SIP to address Host Port: 808971552448...@xx.xx.xx.55

 

 

please advice 

 

regards

 

 

From: M.Khaled W Chehab [mailto:kche...@icucall.com] 
Sent: Wednesday, June 19, 2013 2:41 PM
To: users@lists.opensips.org
Cc: users-boun...@lists.opensips.org
Subject: uac_replace_to problem 

 

Hi,

 

I am running opensips 1.8.3 with  do_routing module 

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

 

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using uac_replace_to( and 
the call go to gw1 with correct TO tag as I set it ,

 

but when calls fails on gw1 ,then  I set the $rU in route[6] to go to second in 
route and it goes with bad TO header, since it goes with the same To header in 
the 1st invite 

That target gw1 

1-how to fix the To header in the second invite to gw2

 

 

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

 

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");

   t_on_failure("1");

 

on failure_route[1] {

.

if (!t_check_status("487")) {

 #xlog("route6---\n");

 $avp(failure_count) = $avp(failure_count) + 1;  
#480|486|603  

   route(6);

 

} 

 

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td");

 

}

 

Regards

 


___
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread Flavio Goncalves

Hi Khaled,

I can try to show you the way, but to walk is up to you ;-).

Flavio E. Goncalves


2013/6/19 M.Khaled W Chehab 

> Yes,
>
> ** **
>
> I am in need to change it every time I send the call to different trunk ,*
> ***
>
> Is there a way to restore the original header before changing it second
> time as I try  
>
> Uac_restore_to() before uac_replace_to but it didn’t work  too 
>
> ** **
>
> Regards
>
> ** **
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Stefano Pisani
> *Sent:* Wednesday, June 19, 2013 4:33 PM
>
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] uac_replace_to problem
>
> ** **
>
> Hi ,
>
> have you changed to twice?
> This happens if you try to use uac_replace_to (or uac_replace_from) twice.
>
> s
>
> Il 19/06/2013 15.18, M.Khaled W Chehab ha scritto:
>
> while i am using uac_replace_to in failover route branch i can find that
> TO header is not changed(sip user part ) but appended an new raw  as I want
> it to be To: "971552448304" 
>
>  
>
> SIP to address:
> sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55
>
> SIP to address User Part: 835822971552448304
>
> SIP to address Host Part: xx.xx.xx.55sip
>
> SIP to address Host Port: 808971552448...@xx.xx.xx.55
>
>  
>
>  
>
> please advice 
>
>  
>
> regards
>
>  
>
>  
>
> *From:* M.Khaled W Chehab [mailto:kche...@icucall.com]
>
> *Sent:* Wednesday, June 19, 2013 2:41 PM
> *To:* users@lists.opensips.org
> *Cc:* users-boun...@lists.opensips.org
> *Subject:* uac_replace_to problem 
>
>  
>
> Hi,
>
>  
>
> I am running opensips 1.8.3 with  do_routing module 
>
> A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)
>
>  
>
> After do_routing(,)
>
> I am setting the $rU and fixing  the To Tag header using uac_replace_to(
> and the call go to gw1 with correct TO tag as I set it ,
>
>  
>
> but when calls fails on gw1 ,then  I set the $rU in route[6] to go to
> second in route and it goes with bad TO header, since it goes with the same
> To header in the 1st invite 
>
> That target gw1 
>
> 1-how to fix the To header in the second invite to gw2
>
>  
>
>  
>
> loadmodule "uac.so"
>
> modparam("uac","restore_mode","auto")
>
>  
>
> uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");
>
>t_on_failure("1");
>
>  
>
> on failure_route[1] {
>
> .
>
> if (!t_check_status("487")) {
>
>  #xlog("route6---\n");
>
>  $avp(failure_count) = $avp(failure_count) +
> 1;  #480|486|603  
>
>route(6);
>
>  
>
> } 
>
>  
>
> Route[6]{
>
> .
>
> .
>
> .
>
> $rU = $var(prefix) + $avp(dst);
>
> uac_replace_to("sip:$var(prefix)$avp(dst)@$td");
>
>  
>
> }
>
>  
>
> Regards
>
>  
>
>
>
>
> 
>
> ___
>
> 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
>
>
___
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread M.Khaled W Chehab

Yes,

 

I am in need to change it every time I send the call to different trunk ,

Is there a way to restore the original header before changing it second time
as I try  

Uac_restore_to() before uac_replace_to but it didn't work  too 

 

Regards

 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Stefano Pisani
Sent: Wednesday, June 19, 2013 4:33 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] uac_replace_to problem

 

Hi ,
have you changed to twice?
This happens if you try to use uac_replace_to (or uac_replace_from) twice.

s

Il 19/06/2013 15.18, M.Khaled W Chehab ha scritto:

while i am using uac_replace_to in failover route branch i can find that TO
header is not changed(sip user part ) but appended an new raw  as I want it
to be To: "971552448304" 

 

SIP to address:
sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55

SIP to address User Part: 835822971552448304

SIP to address Host Part: xx.xx.xx.55sip

SIP to address Host Port: 808971552448...@xx.xx.xx.55

 

 

please advice 

 

regards

 

 

From: M.Khaled W Chehab [mailto:kche...@icucall.com] 
Sent: Wednesday, June 19, 2013 2:41 PM
To: users@lists.opensips.org
Cc: users-boun...@lists.opensips.org
Subject: uac_replace_to problem 

 

Hi,

 

I am running opensips 1.8.3 with  do_routing module 

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

 

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using uac_replace_to( and
the call go to gw1 with correct TO tag as I set it ,

 

but when calls fails on gw1 ,then  I set the $rU in route[6] to go to second
in route and it goes with bad TO header, since it goes with the same To
header in the 1st invite 

That target gw1 

1-how to fix the To header in the second invite to gw2

 

 

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

 

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td
 ");

   t_on_failure("1");

 

on failure_route[1] {

.

if (!t_check_status("487")) {

 #xlog("route6---\n");

 $avp(failure_count) = $avp(failure_count) + 1;
#480|486|603  

   route(6);

 

} 

 

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td
 ");

 

}

 

Regards

 






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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread Stefano Pisani


Hi ,
have you changed to twice?
This happens if you try to use uac_replace_to (or uac_replace_from) twice.

s

Il 19/06/2013 15.18, M.Khaled W Chehab ha scritto:


while i am using uac_replace_to in failover route branch i can find 
that TO header is not changed(sip user part ) but appended an new raw 
 as I want it to be To: "971552448304" 


SIP to address: 
sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55


SIP to address User Part: 835822971552448304

SIP to address Host Part: xx.xx.xx.55sip

SIP to address Host Port: 808971552448...@xx.xx.xx.55 



please advice

regards

*From:*M.Khaled W Chehab [mailto:kche...@icucall.com]
*Sent:* Wednesday, June 19, 2013 2:41 PM
*To:* users@lists.opensips.org
*Cc:* users-boun...@lists.opensips.org
*Subject:* uac_replace_to problem

Hi,

I am running opensips 1.8.3 with  do_routing module

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using 
uac_replace_to( and the call go to gw1 with correct TO tag as I set it ,


but when calls fails on gw1 ,then  I set the $rU in route[6] to go to 
second in route and it goes with bad TO header, since it goes with the 
same To header in the 1^st invite


That target gw1

1-how to fix the To header in the second invite to gw2

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td 
");


t_on_failure("1");

on failure_route[1] {

.

if (!t_check_status("487")) {

#xlog("route6---\n");

$avp(failure_count) = $avp(failure_count) + 1; #480|486|603

route(6);

}

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td 
");


}

Regards



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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread M.Khaled W Chehab

Please can you show me what do you mean by coding it ,what do you mean
branch route  for example is it route (6) I am using , you can find my
config below 

 

while i am using uac_replace_to in failover route branch i can find that TO
header is not changed(sip user part ) but appended an new raw  as I want it
to be To: "971552448304" 

 

SIP to address:
sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55

SIP to address User Part: 835822971552448304

SIP to address Host Part: xx.xx.xx.55sip

SIP to address Host Port: 808971552448...@xx.xx.xx.55

 

 

Thanks in advance 

 

regards

 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Flavio Goncalves
Sent: Wednesday, June 19, 2013 3:13 PM
To: OpenSIPS users mailling list
Cc: users-boun...@lists.opensips.org
Subject: Re: [OpenSIPS-Users] uac_replace_to problem

 

Hi Khaleb, 

 

Set uac_replace_to and uac_replace_from in a branch_route. 

 

Flavio E. Goncalves

 

 

2013/6/19 M.Khaled W Chehab 

Hi,

 

I am running opensips 1.8.3 with  do_routing module 

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

 

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using uac_replace_to( and
the call go to gw1 with correct TO tag as I set it ,

 

but when calls fails on gw1 ,then  I set the $rU in route[6] to go to second
in route and it goes with bad TO header, since it goes with the same To
header in the 1st invite 

That target gw1 

1-how to fix the To header in the second invite to gw2

 

 

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

 

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");

   t_on_failure("1");

 

on failure_route[1] {

.

if (!t_check_status("487")) {

 #xlog("route6---\n");

 $avp(failure_count) = $avp(failure_count) + 1;
#480|486|603  

   route(6);

 

} 

 

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td");

 

}

 

Regards

 


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Re: [OpenSIPS-Users] Setting up opensips

2013-06-19 Thread Bogdan-Andrei Iancu

Hello Stuart,

Maybe your calls are negatively replied by OpenSIPS ? are you doing any
t_relay() after the LB part ?

It seems a script error to me.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/18/2013 11:56 AM, Stuart Mills wrote:
> Hi,
>  
> I wonder if anyone can help me.
>  
> I’ve setup OpenSIPS 1.9.1-tls stable on Debian 7, I have installed the
> mysql module, create the database schema and gone through make
> menuselect to create a load balance script, saved and generated the
> script, all seemed to go according to plan, after that I created some
> load balance nodes using some examples as a guide.
>  
> The problem I seem to be having is when the calls arrive they show up
> on the UAS_transactions count but the call doesn’t seem to be passed
> onto any of the load balance nodes.
>  
> Is there a way i can see what OpenSIPS is trying to do with the call?
>  
> Regards,
>  
> Stuart Mills
>  
>  
>
>
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread Laszlo

BTW, looks like you doing uac_replace_to twice. once in global, and once in
a per-branch route.
You can do it only once, I'm sure you see something about this in the
syslog.

try what Flavio suggested, so do these changes only in the branch route.

-Laszlo


2013/6/19 M.Khaled W Chehab 

> while i am using uac_replace_to in failover route branch i can find that
> TO header is not changed(sip user part ) but appended an new raw  as I want
> it to be To: "971552448304" 
>
> ** **
>
> SIP to address: sip:835822971552448...@xx.xx.xx.55sip
> :808971552448...@xx.xx.xx.55
>
> SIP to address User Part: 835822971552448304
>
> SIP to address Host Part: xx.xx.xx.55sip
>
> SIP to address Host Port: 808971552448...@xx.xx.xx.55
>
> ** **
>
> ** **
>
> please advice 
>
> ** **
>
> regards
>
> ** **
>
> ** **
>
> *From:* M.Khaled W Chehab [mailto:kche...@icucall.com]
> *Sent:* Wednesday, June 19, 2013 2:41 PM
> *To:* users@lists.opensips.org
> *Cc:* users-boun...@lists.opensips.org
> *Subject:* uac_replace_to problem 
>
> ** **
>
> Hi,
>
> ** **
>
> I am running opensips 1.8.3 with  do_routing module 
>
> A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)
>
> ** **
>
> After do_routing(,)
>
> I am setting the $rU and fixing  the To Tag header using uac_replace_to(
> and the call go to gw1 with correct TO tag as I set it ,
>
> ** **
>
> but when calls fails on gw1 ,then  I set the $rU in route[6] to go to
> second in route and it goes with bad TO header, since it goes with the same
> To header in the 1st invite 
>
> That target gw1 
>
> 1-how to fix the To header in the second invite to gw2
>
> ** **
>
> ** **
>
> loadmodule "uac.so"
>
> modparam("uac","restore_mode","auto")
>
> ** **
>
> uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");
>
>t_on_failure("1");
>
> ** **
>
> on failure_route[1] {
>
> .
>
> if (!t_check_status("487")) {
>
>  #xlog("route6---\n");
>
>  $avp(failure_count) = $avp(failure_count) +
> 1;  #480|486|603  
>
>route(6);
>
>  
>
> } 
>
> ** **
>
> Route[6]{
>
> .
>
> .
>
> .
>
> $rU = $var(prefix) + $avp(dst);
>
> uac_replace_to("sip:$var(prefix)$avp(dst)@$td");
>
> ** **
>
> }
>
> ** **
>
> Regards
>
> ** **
>
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>
>
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread M.Khaled W Chehab

while i am using uac_replace_to in failover route branch i can find that TO
header is not changed(sip user part ) but appended an new raw  as I want it
to be To: "971552448304" 

 

SIP to address:
sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55

SIP to address User Part: 835822971552448304

SIP to address Host Part: xx.xx.xx.55sip

SIP to address Host Port: 808971552448...@xx.xx.xx.55

 

 

please advice 

 

regards

 

 

From: M.Khaled W Chehab [mailto:kche...@icucall.com] 
Sent: Wednesday, June 19, 2013 2:41 PM
To: users@lists.opensips.org
Cc: users-boun...@lists.opensips.org
Subject: uac_replace_to problem 

 

Hi,

 

I am running opensips 1.8.3 with  do_routing module 

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

 

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using uac_replace_to( and
the call go to gw1 with correct TO tag as I set it ,

 

but when calls fails on gw1 ,then  I set the $rU in route[6] to go to second
in route and it goes with bad TO header, since it goes with the same To
header in the 1st invite 

That target gw1 

1-how to fix the To header in the second invite to gw2

 

 

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

 

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");

   t_on_failure("1");

 

on failure_route[1] {

.

if (!t_check_status("487")) {

 #xlog("route6---\n");

 $avp(failure_count) = $avp(failure_count) + 1;
#480|486|603  

   route(6);

 

} 

 

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td");

 

}

 

Regards

 

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Re: [OpenSIPS-Users] opensips control panel

2013-06-19 Thread Flavio Goncalves

Hi,

Please, check your DB parameters in db.inc.php.

Flavio E. Goncalves




2013/6/19 Nandini madhu 

> Dear All,
> Greetings,
>
> i have got the mi_xmlrpc.so file. but in control panel the error is:-
>
> *Failed to issue query, error message : MDB2 Error: no such table*
>
> thanks in advance
>
> please help me.
>
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Re: [OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread Flavio Goncalves

Hi Khaleb,

Set uac_replace_to and uac_replace_from in a branch_route.

Flavio E. Goncalves



2013/6/19 M.Khaled W Chehab 

> Hi,
>
> ** **
>
> I am running opensips 1.8.3 with  do_routing module 
>
> A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)
>
> ** **
>
> After do_routing(,)
>
> I am setting the $rU and fixing  the To Tag header using uac_replace_to(
> and the call go to gw1 with correct TO tag as I set it ,
>
> ** **
>
> but when calls fails on gw1 ,then  I set the $rU in route[6] to go to
> second in route and it goes with bad TO header, since it goes with the same
> To header in the 1st invite 
>
> That target gw1 
>
> 1-how to fix the To header in the second invite to gw2
>
> ** **
>
> ** **
>
> loadmodule "uac.so"
>
> modparam("uac","restore_mode","auto")
>
> ** **
>
> uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");
>
>t_on_failure("1");
>
> ** **
>
> on failure_route[1] {
>
> .
>
> if (!t_check_status("487")) {
>
>  #xlog("route6---\n");
>
>  $avp(failure_count) = $avp(failure_count) +
> 1;  #480|486|603  
>
>route(6);
>
>  
>
> } 
>
> ** **
>
> Route[6]{
>
> .
>
> .
>
> .
>
> $rU = $var(prefix) + $avp(dst);
>
> uac_replace_to("sip:$var(prefix)$avp(dst)@$td");
>
> ** **
>
> }
>
> ** **
>
> Regards
>
> ** **
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] uac_replace_to problem

2013-06-19 Thread M.Khaled W Chehab

Hi,

 

I am running opensips 1.8.3 with  do_routing module 

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

 

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using uac_replace_to( and
the call go to gw1 with correct TO tag as I set it ,

 

but when calls fails on gw1 ,then  I set the $rU in route[6] to go to second
in route and it goes with bad TO header, since it goes with the same To
header in the 1st invite 

That target gw1 

1-how to fix the To header in the second invite to gw2

 

 

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

 

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td");

   t_on_failure("1");

 

on failure_route[1] {

.

if (!t_check_status("487")) {

 #xlog("route6---\n");

 $avp(failure_count) = $avp(failure_count) + 1;
#480|486|603  

   route(6);

 

} 

 

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td");

 

}

 

Regards

 

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[OpenSIPS-Users] opensips control panel

2013-06-19 Thread Nandini madhu

Dear All,
Greetings,

i have got the mi_xmlrpc.so file. but in control panel the error is:-

*Failed to issue query, error message : MDB2 Error: no such table*

thanks in advance

please help me.
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Re: [OpenSIPS-Users] Lync Client through OpenSIPS

2013-06-19 Thread Bogdan-Andrei Iancu

Hello Jens,

Try to do topology hiding on OpenSIPS based on dialog module (for calls
between Lync clients and Lync server) - I remember Lync having some
issues with dealing with a proxy.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/17/2013 01:32 PM, Jens Grönningberg wrote:
> Hi Bogdan,
>
> thanks for your response. We use OpenSIPS to take care of NAT
> traversal in a server that acts as the firewall for our organization.
> It already works for normal calls. Now we need to be able to get Lync
> clients from the Internet to be able to connect to the Lync server
> through our OpenSIPS. That's what we are having trouble with. OpenSIPS
> seems to do everything fine according to the network captures of the
> SIP traffic I've made. It is the Lync server who seems to behave
> differently when the SIP traffic is passing through OpenSIPS. I
> already asked in Microsoft forums without luck, that's why I wanted to
> ask here whether someone successfully achieved this kind of integration.
>
> Thanks. Regards,
>
> Jens
>
>
> On Thu, Jun 13, 2013 at 7:15 PM, Bogdan-Andrei Iancu
> mailto:bog...@opensips.org>> wrote:
>
> Hello Jens,
>
> I successfully integrated OpenSIPS (acting as SBC) with Lync,
> typically
> for calls (no presence and IM).
>
> What kind o integration are you trying to do (what is the purpose of
> OpenSIPs in your network) ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 06/13/2013 12:03 PM, Jens Grönningberg wrote:
> > Hi! Has someone managed to get the Microsoft Lync client
> communicate with
> > the Lync Server through OpenSIPS? I tried but the Lync Server
> behaved
> > differently, refusing to send BENOTIFY requests. This caused
> presence not
> > to work properly, and IM not to work at all. I would be happy to
> hear if
> > someone had success with this.
> >
> > Thanks.
> >
> > ___
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> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
>
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Re: [OpenSIPS-Users] opensips bad accounting

2013-06-19 Thread Bogdan-Andrei Iancu

Hello Khaled,

If opensips is not responding to the BYEs is because of your config (for
sequential requests) - you have to check were the BYE gets lost in your
routing logic.

IF the BYE is lost, the dialog (from opensips perspective) is still
ongoing and the dialog will be terminated due a timeout - this may
explain the duration in CDR.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/17/2013 12:10 PM, M.Khaled W Chehab wrote:
>
> Hi,
>
>  
>
> I am using  opensips 1.8.3 with acc module for accounting  with cdr flag
>
> When I  restart  opensips using opensipsctl restart , and while
> getting back to wireshark capture I recognize that opensips is not
> responding to the BYE sip messages that been  sent from provider  or
> from  user agent  for the connected calls before I restart opensips
>
> And the duration will be written in the cdr flag  50372  about 839
> minutes moreover   I find a row in acc table where method = bye
>  having same call-ID  and the time field  is  showing a difference 10
> hours from the  method invite
>
>  
>
> 1-how to restart safely with no error in accounting
>
> Please advice
>
>  
>
> Regards
>
>  
>
>
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Re: [OpenSIPS-Users] Call Generator

2013-06-19 Thread Maciej Bylica

Thanks Adrian,

I will take a look on this...

Mac.


2013/6/18 Adrian Georgescu 

> See sipclient package, it contains this tool that does pretty much all you
> are looking for:
>
> http://sipsimpleclient.org/projects/sipsimpleclient/wiki/Sip_audio_session
>
> Adrian
>
> On Jun 18, 2013, at 7:03 PM, Maciej Bylica  wrote:
>
> Hello,
>
> I am looking for call generator that is capable of:
> - generating and in the same time pick up the call (the call will traverse
> infrastructure under testing and get back to generator)
> - generating SIP + RTP calls. There must be many .wav or mp3 files
> possible to be used
> - heaving random call duration
> - heaving a possibility to set Called and Called numbers random in
> specified ranges (like 1122233[0-9]{4} for instance).
>
> Have you tested any call generator that has aforementioned functionality
> implemented?
> I know that Opensips could be used for this purpose, but i am looking for
> the ready-to-run product.
>
> Thanks,
> Mac.
>
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Re: [OpenSIPS-Users] SIP reinvite /update

2013-06-19 Thread Bogdan-Andrei Iancu

Hello Khaled,

I'm not fully understanding the question.  You cannot hangup an INVITE,
but a call. Also a call cannot have a TO tag, but only a SIP message can
have.

Could you please rephrase ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/17/2013 12:01 PM, M.Khaled W Chehab wrote:
>
> Hi,
>
>  
>
> I am using opensips 1.8.3
>
> How to stop the  sip invite update or the sip reinvite behavior when a
> call has a has_totag without hanging up   the first  invite, as I
> already set
>
> if (has_totag()) {
>
> if (loose_route()) {
>
> }
>
> t_check_trans();  
>
> }
>
>  
>
>  
>
> Regards
>
>  
>
>  
>
>
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Re: [OpenSIPS-Users] Opensips Critical Hutdown

2013-06-19 Thread Bogdan-Andrei Iancu

Hello Khaled,

That messages tells you that the OpenSIPS shutdown took longer than 20
seconds (and the shutdown was interrupted).

Why the shutdown takes more than 20 secs - it may be because there is a
large amount of data that needs to be flushed to DB (like usrloc,
dialogs, etc) or it may be because of a bug or so.

The shutdown is done explicitly by you or opensips crashes ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 06/16/2013 02:30 PM, M.Khaled W Chehab wrote:
>
> Hi,
>
>  
>
> I am using opensips 1.8.3 .
>
> Opensips stops suddenly and I can find this error in my log file
>
> CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying...
>
>  
>
>  
>
> Please advice
>
> Regards
>
>  
>
>
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Re: [OpenSIPS-Users] [Bulk] Re: Setting up opensips

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] [Bulk] Re: Setting up opensips

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] Setting up opensips

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] uac_replace_to problem

Re: [OpenSIPS-Users] opensips control panel

Re: [OpenSIPS-Users] uac_replace_to problem

[OpenSIPS-Users] uac_replace_to problem

[OpenSIPS-Users] opensips control panel

Re: [OpenSIPS-Users] Lync Client through OpenSIPS

Re: [OpenSIPS-Users] opensips bad accounting

Re: [OpenSIPS-Users] Call Generator

Re: [OpenSIPS-Users] SIP reinvite /update

Re: [OpenSIPS-Users] Opensips Critical Hutdown

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