opensipstack-devel

Re: [OpenSIPStack] Call Forwarding Loop Detected and Notify

[Joegen E. B. <joe...@gm...>]

Julian,

Please send in the logs.  If possible create a bug report in 
bugs.opensourcesip.org and attach the logs there together with the 
description of the bug.  Please also include the procedure to replicate 
it.  Thanks.

Joegen


Julian F. Tasis, III wrote:
> Good day,
>
> I grabe the latest cvs this morning and tested the patch. Notify is working fine but for the call forwarding functions, the logs says "loop detected" when redirecting the call. 
>
> Thanks for the patch.
>
> Julian
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[OpenSIPStack] Call Forwarding Loop Detected and Notify

[Julian F. T. I. <jf...@id...>]

Good day,

I grabe the latest cvs this morning and tested the patch. Notify is =
working fine but for the call forwarding functions, the logs says "loop =
detected" when redirecting the call.=20

Thanks for the patch.

Julian

Re: [OpenSIPStack] Testing of Refer and NOTIFY

[Joegen E. B. <joe...@gm...>]

Julian F. Tasis, III wrote:
> Good day,
>
> I'm testing opensbc refer functionality and NOTIFY. The sip client is replying 400 messages. I looked on the packet and found out that the content-length from the NOTIFY message is incorrect. The last \r\n was not counted. 
>
>   

Patched.   Try CVS head.

Re: [OpenSIPStack] Call Forwarding Problem

[Joegen E. B. <joe...@gm...>]

Julian F. Tasis, III wrote:
> Good day,
>
> Hi, I'm having a problem with call forward right now. This is the call setup I'm doing. 
>
> client1: 8000
> client2: 1212
> client3: 1512
>
> a. all of the said client ared registered on osbc.
> b. 1212 is set to always call forward to 1512.
> c. client1 calls 1212 
> d. client2 responded the invite with 302 move temporarily, contact header contains 1512 as the next address.
> e. opensbc replied ACK to 302. 
>
> Opensbc did not send the invite to 1512.
>
> I looked on the log files, opensbc created the invite going to 1512 for redirection but it has encountered a transport error and the current transaciton was queued for deletion. 
>   


This is caused by the failure of OpenSBC to perform a separate routing 
for 3xx forwarded back to the SBC local domain.   The correct behavior 
should do a reg db lookup.  I have provided a patch for this problem in 
CVS.  Please test again and let me know if the problem is fixed.

[OpenSIPStack] Testing of Refer and NOTIFY

[Julian F. T. I. <jf...@id...>]

Good day,

I'm testing opensbc refer functionality and NOTIFY. The sip client is =
replying 400 messages. I looked on the packet and found out that the =
content-length from the NOTIFY message is incorrect. The last \r\n was =
not counted.=20

Here's my capture.

Julian

Re: [OpenSIPStack] Issue Regarding RTP seems to be bypassing OpenSBC when using Gafachi as a service provider.

[Joegen E. B. <joe...@gm...>]

Would Gafachi be able to route RTP to 192.168.96.115?  This sees to be 
an internal private address.  You need to set the Static RTP Address to 
the public interface of your NAT router and make sure that your NAT maps 
the RTP ports properly.  RTP port range of OpenSBC 30000-35000.  Lastly, 
run your SBC in B2BUA mode.  Your INVITEs are getting proxied and RTP 
proxy does not apply to plain proxied requests.


Ashok Gupta wrote:
> Hello,
>  
> I am having some issue in getting RTP packets when using Gafachi as a 
> service provider.
>  
>
> When I use a network configuration like
>  
> Xlite UA (Outbound) (.64) <--------> OpenSBC (.115) <-----------> Xlite UA
> (Inbound) (.60)
>  
> * Outbound UA is registered on OpenSBC
>  
> * Inbound UA is registered on OpenSBC
>  
> * A route exists from OpenSBC to Xlite UA (Inbound) (.60)
>  
> Rtp packets are coming fine at opensbc.
>  
>
> But when I am having a configuration like
>  
> Xlite UA (Outbound) (.64) <--------> OpenSBC (.115) <-----------> Xlite UA
> (Inbound) (.60)
>  
> * Outbound UA is registered on Gafachi       //Sip Proxy is set to opensbc
>  
> * Inbound UA is registered on Gafachi
>  
> * A route exists from OpenSBC to Gafachi UA (Inbound) (.60)
>  
>
> I am not getting RTP packets at opensbc side (SIP packets are coming 
> fine,RTP seems to be bypassing OpenSBC).When I looked at traces I 
> found that In this case when opensbc Route sip Invite to Gafachi SDP 
> is not getting change.
>  
> In my config settings with Gafachi  I am having
>  
> SBC Mode  FullMode
> Interface Address  opensbc machine Ip(192.168.96.115)
> Alwayes Proxy media true
> Static RTP media address opensbc machine Ip(192.168.96.115)
>  
> B2BUA Routes--> [sip:*@gafachi] sip:"sipproxy address"
> Relay Route--> [sip:*@gafachi] sip:"sipproxy address"
>  
> Upper Registration Routes-->[sip:*@gafachi] sip:"sipproxy address"
>  
>
> Also
> >>Do we need a external NAT router to have this possible?
>  
> Attached are the level 5 logs with Gafachi call and without gafachi call.
> I'll be thankful for any guidance on what's going wrong.
>  
> Regards
> Ashok
>  
>  
>  
>  
>  
>
>  

Re: [OpenSIPStack] Question Regarding opening the RTP sockets in openSBC.

[Joegen E. B. <joe...@gm...>]

Yes currently this is a limitation.  OpenSBC utilizes two threads per 
connection for the RTP proxy.    Media aggregation allows you to use a 
finite set of threads to handle multiple number of rtp sessions.     
Media aggregation code is incomplete and has never been tested.

Ashish Khare wrote:
> Hi joegen,
> Sorry for clarifying once again.
> As of now openSBC is limited by the number of threads, the system can have,
> in handling the RTP streams, as 2 threads are needed for  one RTP session.
> Please reply.
>
> -Ashish
>
>
> On 8/21/07, Joegen E. Baclor <joe...@gm...> wrote:
>   
>> That is TRUE.   If you check the Media Interface handler...  there are
>> inactive code there that pertains to media aggregation.   This code
>> allows for the RTP streams to be aggregated by a limited number of
>> threads which will allow OpenSBC to just be limited by the CPU in terms
>> of max concurrency.   However, the OpenSBC team does not have time right
>> now to revisit this "incomplete" segment.  Feel free to work on this
>> code and send back your patches.
>>
>> Ashish Khare wrote:
>>     
>>> Hi Joegen,
>>>
>>> Thanks for the reply.
>>> But if this is the case, then concurrent media sessions is
>>>       
>> not  restricted
>>     
>>> to the maximun number
>>> of threads the system / machine can have.
>>> How can we scale the system to accomodate lets say 1 million subscriber
>>> which can lead to 1 million media sessions concurrently?
>>>
>>> Thanks,
>>> Ashish
>>>
>>> On 8/21/07, Joegen E. Baclor <joe...@gm...> wrote:
>>>
>>>       
>>>> Ashish Khare wrote:
>>>>
>>>>         
>>>>> Hi,
>>>>>
>>>>> I have a question regarding the opening of RTP sockets in the openSBC.
>>>>> When openSBC recieved the INVITE request, it creates the new rtp/rtcp
>>>>> sockets for that leg in the new thread.
>>>>> Similarly when openSBC sends the outgoing INVITE in the same Call
>>>>> session, it creates another new rtp/rtcp sockets for the call leg 2 in
>>>>> the new thread instance.
>>>>> So altogether 2 threads are there for one call between 2 parties to
>>>>> handle RTP/RTCP packets.
>>>>> Then if there are N * 2 subscribers are talking to each other, then it
>>>>> means there are N * 2 threads handling the RTP streams.
>>>>> is my understanding correct?
>>>>>
>>>>>
>>>>>
>>>>>           
>>>> yes.
>>>>
>>>>
>>>>         
>> -------------------------------------------------------------------------
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>>>       
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>>     
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Re: [OpenSIPStack] Question Regarding opening the RTP sockets in openSBC.

[Ashish K. <ash...@gm...>]

Hi joegen,
Sorry for clarifying once again.
As of now openSBC is limited by the number of threads, the system can have,
in handling the RTP streams, as 2 threads are needed for  one RTP session.
Please reply.

-Ashish


On 8/21/07, Joegen E. Baclor <joe...@gm...> wrote:
>
> That is TRUE.   If you check the Media Interface handler...  there are
> inactive code there that pertains to media aggregation.   This code
> allows for the RTP streams to be aggregated by a limited number of
> threads which will allow OpenSBC to just be limited by the CPU in terms
> of max concurrency.   However, the OpenSBC team does not have time right
> now to revisit this "incomplete" segment.  Feel free to work on this
> code and send back your patches.
>
> Ashish Khare wrote:
> > Hi Joegen,
> >
> > Thanks for the reply.
> > But if this is the case, then concurrent media sessions is
> not  restricted
> > to the maximun number
> > of threads the system / machine can have.
> > How can we scale the system to accomodate lets say 1 million subscriber
> > which can lead to 1 million media sessions concurrently?
> >
> > Thanks,
> > Ashish
> >
> > On 8/21/07, Joegen E. Baclor <joe...@gm...> wrote:
> >
> >> Ashish Khare wrote:
> >>
> >>> Hi,
> >>>
> >>> I have a question regarding the opening of RTP sockets in the openSBC.
> >>> When openSBC recieved the INVITE request, it creates the new rtp/rtcp
> >>> sockets for that leg in the new thread.
> >>> Similarly when openSBC sends the outgoing INVITE in the same Call
> >>> session, it creates another new rtp/rtcp sockets for the call leg 2 in
> >>> the new thread instance.
> >>> So altogether 2 threads are there for one call between 2 parties to
> >>> handle RTP/RTCP packets.
> >>> Then if there are N * 2 subscribers are talking to each other, then it
> >>> means there are N * 2 threads handling the RTP streams.
> >>> is my understanding correct?
> >>>
> >>>
> >>>
> >> yes.
> >>
> >>
> >
> -------------------------------------------------------------------------
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> >
> >
> >
>
>
> -------------------------------------------------------------------------
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Re: [OpenSIPStack] Question Regarding opening the RTP sockets in openSBC.

[Joegen E. B. <joe...@gm...>]

That is TRUE.   If you check the Media Interface handler...  there are 
inactive code there that pertains to media aggregation.   This code 
allows for the RTP streams to be aggregated by a limited number of 
threads which will allow OpenSBC to just be limited by the CPU in terms 
of max concurrency.   However, the OpenSBC team does not have time right 
now to revisit this "incomplete" segment.  Feel free to work on this 
code and send back your patches. 

Ashish Khare wrote:
> Hi Joegen,
>
> Thanks for the reply.
> But if this is the case, then concurrent media sessions is not  restricted
> to the maximun number
> of threads the system / machine can have.
> How can we scale the system to accomodate lets say 1 million subscriber
> which can lead to 1 million media sessions concurrently?
>
> Thanks,
> Ashish
>
> On 8/21/07, Joegen E. Baclor <joe...@gm...> wrote:
>   
>> Ashish Khare wrote:
>>     
>>> Hi,
>>>
>>> I have a question regarding the opening of RTP sockets in the openSBC.
>>> When openSBC recieved the INVITE request, it creates the new rtp/rtcp
>>> sockets for that leg in the new thread.
>>> Similarly when openSBC sends the outgoing INVITE in the same Call
>>> session, it creates another new rtp/rtcp sockets for the call leg 2 in
>>> the new thread instance.
>>> So altogether 2 threads are there for one call between 2 parties to
>>> handle RTP/RTCP packets.
>>> Then if there are N * 2 subscribers are talking to each other, then it
>>> means there are N * 2 threads handling the RTP streams.
>>> is my understanding correct?
>>>
>>>
>>>       
>> yes.
>>
>>     
> -------------------------------------------------------------------------
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> Still grepping through log files to find problems?  Stop.
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> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] Question Regarding opening the RTP sockets in openSBC.

[Ashish K. <ash...@gm...>]

Hi Joegen,

Thanks for the reply.
But if this is the case, then concurrent media sessions is not  restricted
to the maximun number
of threads the system / machine can have.
How can we scale the system to accomodate lets say 1 million subscriber
which can lead to 1 million media sessions concurrently?

Thanks,
Ashish

On 8/21/07, Joegen E. Baclor <joe...@gm...> wrote:
>
> Ashish Khare wrote:
> > Hi,
> >
> > I have a question regarding the opening of RTP sockets in the openSBC.
> > When openSBC recieved the INVITE request, it creates the new rtp/rtcp
> > sockets for that leg in the new thread.
> > Similarly when openSBC sends the outgoing INVITE in the same Call
> > session, it creates another new rtp/rtcp sockets for the call leg 2 in
> > the new thread instance.
> > So altogether 2 threads are there for one call between 2 parties to
> > handle RTP/RTCP packets.
> > Then if there are N * 2 subscribers are talking to each other, then it
> > means there are N * 2 threads handling the RTP streams.
> > is my understanding correct?
> >
> >
> yes.
>

Re: [OpenSIPStack] Question Regarding opening the RTP sockets in openSBC.

[Joegen E. B. <joe...@gm...>]

Ashish Khare wrote:
> Hi,
>  
> I have a question regarding the opening of RTP sockets in the openSBC.
> When openSBC recieved the INVITE request, it creates the new rtp/rtcp 
> sockets for that leg in the new thread.
> Similarly when openSBC sends the outgoing INVITE in the same Call 
> session, it creates another new rtp/rtcp sockets for the call leg 2 in 
> the new thread instance.
> So altogether 2 threads are there for one call between 2 parties to 
> handle RTP/RTCP packets.
> Then if there are N * 2 subscribers are talking to each other, then it 
> means there are N * 2 threads handling the RTP streams.
> is my understanding correct?
>  
>  
yes. 

Re: [OpenSIPStack] Maximum Capacity of openSBC.

[Joegen E. B. <joe...@gm...>]

Ashish Khare wrote:
> Hi,
>  
> I want to know how many Sessions ( SIP Sessions + RTP Sessions ) can 
> openSBC concurrently held.
> What is openSBC maximum capacity?
Currently maximum (tested) capacity is at 1000 concurrent connections.   
CPS is currently timed at 30 calls per second throughput.     OpenSBC is 
tested to accommodate up to 400 connections when media is proxied.

Re: [OpenSIPStack] Call Forwarding Problem

[Joegen E. B. <joe...@gm...>]

You forgot to attach the files.

Julian F. Tasis, III wrote:
> Good day,
>
> Hi, I'm having a problem with call forward right now. This is the call setup I'm doing. 
>
> client1: 8000
> client2: 1212
> client3: 1512
>
> a. all of the said client ared registered on osbc.
> b. 1212 is set to always call forward to 1512.
> c. client1 calls 1212 
> d. client2 responded the invite with 302 move temporarily, contact header contains 1512 as the next address.
> e. opensbc replied ACK to 302. 
>
> Opensbc did not send the invite to 1512.
>
> I looked on the log files, opensbc created the invite going to 1512 for redirection but it has encountered a transport error and the current transaciton was queued for deletion. 
>
> Attached are my captures and the log files. 
>
>  Thanks for the help!
>
> Julian
>   
> ------------------------------------------------------------------------
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
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>
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>   

[OpenSIPStack] Call Forwarding Problem

[Julian F. T. I. <jf...@id...>]

Good day,

Hi, I'm having a problem with call forward right now. This is the call =
setup I'm doing.=20

client1: 8000
client2: 1212
client3: 1512

a. all of the said client ared registered on osbc.
b. 1212 is set to always call forward to 1512.
c. client1 calls 1212=20
d. client2 responded the invite with 302 move temporarily, contact =
header contains 1512 as the next address.
e. opensbc replied ACK to 302.=20

Opensbc did not send the invite to 1512.

I looked on the log files, opensbc created the invite going to 1512 for =
redirection but it has encountered a transport error and the current =
transaciton was queued for deletion.=20

Attached are my captures and the log files.=20

 Thanks for the help!

Julian

[OpenSIPStack] Question Regarding opening the RTP sockets in openSBC.

[Ashish K. <ash...@gm...>]

Hi,

I have a question regarding the opening of RTP sockets in the openSBC.
When openSBC recieved the INVITE request, it creates the new rtp/rtcp
sockets for that leg in the new thread.
Similarly when openSBC sends the outgoing INVITE in the same Call session,
it creates another new rtp/rtcp sockets for the call leg 2 in the new thread
instance.
So altogether 2 threads are there for one call between 2 parties to handle
RTP/RTCP packets.
Then if there are N * 2 subscribers are talking to each other, then it means
there are N * 2 threads handling the RTP streams.
is my understanding correct?

[OpenSIPStack] Maximum Capacity of openSBC.

[Ashish K. <ash...@gm...>]

Hi,

I want to know how many Sessions ( SIP Sessions + RTP Sessions ) can openSBC
concurrently held.
What is openSBC maximum capacity?

thanks,
Ashish

Re: [OpenSIPStack] Configurable Timer B

[Gustavo C. <cur...@ho...>]

Hi joegen,
=20
Thanks for your help. I get the value in OpenSBC::OnConfigChanged() instead=
 of  OpenSBCDaemon::OnConfigChanged().
=20
This was the code added in OpenSBC::OnConfigChanged():
=20
  GetStack().SetTimerBDuration( config.GetInteger( configKeySection, "SIP T=
imer B Duration", 10000 ) );
=20
In the class SIPTimerManager:
.....
public:
  void SetTimerBDuration( DWORD duration ){ m_TimerBDuration =3D duration; =
}
  DWORD GetTimerBDuration(){ return m_TimerBDuration; }
private:
  DWORD m_TimerBDuration;
.....
=20
and finnaly in void InviteClientTransaction::HandleStateIdle():
=20
 StartTimer_B(m_TimerManager.GetTimerBDuration());
=20
It's seem to work fine. But i'm not sure if there is a possible collision b=
etween  SetTimerBDuration() when is configured from the http admin and GetT=
imerBDuration() in the transaction layer.
=20
Regards
=20
Gustavo


> Date: Fri, 17 Aug 2007 17:16:29 +0800> To: cur...@gm...; ope=
nsi...@li...> From: joe...@gm...> CC: j=
oeg...@gm...> Subject: Re: [OpenSIPStack] Configurable Timer B> >=
 > Hi gustavo,> > You need to edit oss-application.conf.xml and ad a new it=
em in the > general parameter section. Example:> > <Item Comment=3D"" Type=
=3D"integer" Default=3D"32000" Minimum=3D"5000" > Maximum=3D"32000" Length=
=3D"50">SIP Timer B Duration</Item>> > > This should add a new item in http=
 admin general parameter when you > restart opensbc. You may get the value =
of timer B in method > OpenSBCDaemon::OnConfigChanged() by calling> > int t=
imerbDuration =3D config.GetInterger( configKeySection, "SIP Timer B > Dura=
tion", 32000 );> > > Gustavo Curetti wrote:> > Hi Joegen:> > > > I want to =
make configurable the timer B via http admin. How could i do this? Where i =
have to add the variable and the methods?> > > > Thanks for your help> > > =
> Gustavo> > ______________________________________________________________=
___> > Expr=E9sate - dise=F1a tu p=E1gina de inicio de Live.com como m=E1s =
te guste.> > http://www.live.com/getstarted> > ----------------------------=
---------------------------------------------> > This SF.net email is spons=
ored by: Splunk Inc.> > Still grepping through log files to find problems? =
Stop.> > Now Search log events and configuration files using AJAX and a bro=
wser.> > Download your FREE copy of Splunk now >> http://get.splunk.com/> >=
 _______________________________________________> > opensipstack-devel mail=
ing list> > ope...@li...> > https://lists.sourc=
eforge.net/lists/listinfo/opensipstack-devel> >> >> > > > > > -------------=
------------------------------------------------------------> This SF.net e=
mail is sponsored by: Splunk Inc.> Still grepping through log files to find=
 problems? Stop.> Now Search log events and configuration files using AJAX =
and a browser.> Download your FREE copy of Splunk now >> http://get.splunk.=
com/> _______________________________________________> opensipstack-devel m=
ailing list> ope...@li...> https://lists.source=
forge.net/lists/listinfo/opensipstack-devel
_________________________________________________________________
Expr=E9sate - dise=F1a tu p=E1gina de inicio de Live.com como m=E1s te gust=
e.
http://www.live.com/getstarted=

Re: [OpenSIPStack] Configurable Timer B

[<jo...@op...>]

Hi gustavo,

You need to edit oss-application.conf.xml and ad a new item in the=20
general parameter section.  Example:

<Item Comment=3D"" Type=3D"integer" Default=3D"32000" Minimum=3D"5000"=20
Maximum=3D"32000" Length=3D"50">SIP Timer B Duration</Item>


This should add a new item in http admin general parameter when you=20
restart opensbc.  You may get the value of timer B in method=20
OpenSBCDaemon::OnConfigChanged() by calling

int timerbDuration =3D config.GetInterger( configKeySection, "SIP Timer B=
=20
Duration", 32000 );


Gustavo Curetti wrote:
> Hi Joegen:
> =20
> I want to make configurable the timer B via http admin. How could i do =
this? Where i have to add the variable and the methods?
> =20
> Thanks for your help
> =20
> Gustavo
> _________________________________________________________________
> Expr=E9sate - dise=F1a tu p=E1gina de inicio de Live.com como m=E1s te =
guste.
> http://www.live.com/getstarted
> -----------------------------------------------------------------------=
--
> This SF.net email is sponsored by: Splunk Inc.
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> Download your FREE copy of Splunk now >>  http://get.splunk.com/
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> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>  =20

Re: [OpenSIPStack] IVRHandling End Call

[<jo...@op...>]

Try calling connection->DestroyConnection() in the IVR handler.  

Julian F. Tasis, III wrote:
> Good day,
>
> Hi I got the call transfer back to IVR when B rejected the call and record it if possible. One thing I can't do right that is ending the call. I wanted the media server to initiate the BYE. How can I do that? Is it the same with the transferCall where in the invite message is created from scratch?
>
>
> Thanks,
> Julian
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>
>   

[OpenSIPStack] IVRHandling End Call

[Julian F. T. I. <jf...@id...>]

Good day,

Hi I got the call transfer back to IVR when B rejected the call and =
record it if possible. One thing I can't do right that is ending the =
call. I wanted the media server to initiate the BYE. How can I do that? =
Is it the same with the transferCall where in the invite message is =
created from scratch?


Thanks,
Julian

[OpenSIPStack] Configurable Timer B

[Gustavo C. <cur...@ho...>]

Hi Joegen:
=20
I want to make configurable the timer B via http admin. How could i do this=
? Where i have to add the variable and the methods?
=20
Thanks for your help
=20
Gustavo
_________________________________________________________________
Expr=E9sate - dise=F1a tu p=E1gina de inicio de Live.com como m=E1s te gust=
e.
http://www.live.com/getstarted=

Re: [OpenSIPStack] IVR

[Joegen E. B. <joe...@gm...>]

Try this in your route:

[sip:1800*] sip:5000@localhost:65090

Andre Silo wrote:
> Hello Julian,
>    
>   Have you tried all incoming call from outside the network and let Media Server answer with a voice prompt?
>    
>   1800 number -> OSBC -> 5000 Media Server
>
>        
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>   

[OpenSIPStack] IVR

[Andre S. <eds...@ya...>]

Hello Julian,
   
  Have you tried all incoming call from outside the network and let Media Server answer with a voice prompt?
   
  1800 number -> OSBC -> 5000 Media Server

       
---------------------------------
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Re: [OpenSIPStack] IVR handling on transfer call rejection

[Andre S. <eds...@ya...>]

Tha wave format must be:
   
  File Type: 8000Hz, 16-bit, Mono
File Format: Windows PCM


"Julian F. Tasis, III" <jf...@id...> wrote:
  Hi Andre,

I did not experienced a call being dropped after 2 rings. The log file
will tell you what went wrong on that particular call.

By the way I tried to record my voice using the the sound recorder in
windows, but still can't hear it. I'll try to record again and test it.

> Andre,
>
> Tons of information can be found in the logs. Have you even tried
> looking at it? What do you see?
>
>
> Andre Silo wrote:
>> Julian,
>>
>> Have you tried calling from a remote phone connecting to OSBC? I
>> tested it but after two rings on the local account the line just got
>> disconnected. Is there a parameter that extends the number of rings
>> before it gets rejected?
>>
>> "Julian F. Tasis, III" wrote:
>> Hi,
>>
>> Thanks for the reply. I transfer the call back to the IVR when it is
>> rejected and just added some internal header and it worked.
>>
>> By the way I modify the IVRTransferCall a little bit so that when the
>> media server is routing the call to B's destination, B will see the
>> profile of the caller and not the media server.
>>
>> May I know how did you record all the wav files OpenSBC is using? I
>> tried
>> recording my own wav files but I can't hear my recordings hehe.
>>
>> Regards,
>> Julian F. Tasis, III
>>
>> Re: [OpenSIPStack] IVR handling on transfer call rejection
>> From: - 2007-08-15 03:30
>> There is more work to be done before you will be able to do this. Ryan
>> is working on improvements to the API to allow the transfered
>> connection to reconnect to the IVR once the outbound call is
>> disconnected. Watch the updates.
>>
>>
>> Julian F. Tasis, III wrote:
>>
>>> Good day,
>>>
>>> Hi, I was playing around with the IVRHandler lately and getting to know
>>> how it works. Just got a problem when transferring a call to B
>>> destination. If the B rejects the call, I can't notify A that the call
>>> is
>>> dropped. I wanted to play also a specific speil to A to notify it. I
>>> tried
>>> getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but
>>> with no success. Is there any other way around to do that?
>>>
>>>
>>> Regards,
>>> Julian F. Tasis, III
>>>
>>
>>
>>
>>
>> -------------------------------------------------------------------------
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>> Still grepping through log files to find problems? Stop.
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>> ope...@li...
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>>
>>
>>
>> ---------------------------------
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>> -------------------------------------------------------------------------
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>> Still grepping through log files to find problems? Stop.
>> Now Search log events and configuration files using AJAX and a browser.
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>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>
>


Regards,
Julian F. Tasis, III
Product Planning and Development

Ideawurx Inc.
7th Floor, V.A. Rufino Bldg.,
6784 Ayala Ave.
Makati City, Philippines 1226

Tel: +(632) 8111530
Fax: +(632) 8189767

jf...@id...
http://www.ideawurx.com.ph

Ideawurx is a telecommunications solutions provider that develops and
markets a range of customized computer telephony integration ( CTI )
solutions.

"Our IDEAS will WORK for YOU!"


-------------------------------------------------------------------------
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Re: [OpenSIPStack] IVR handling on transfer call rejection

[Julian F. T. I. <jf...@id...>]

Hi Andre,

I did not experienced a call being dropped after 2 rings. The log file
will tell you what went wrong on that particular call.

By the way I tried to record my voice using the the sound recorder in
windows, but still can't hear it. I'll try to record again and test it.

> Andre,
>
> Tons of information can be found in the logs.    Have you even tried
> looking at it?  What do you see?
>
>
> Andre Silo wrote:
>> Julian,
>>
>>   Have you tried calling from a remote phone connecting to OSBC? I
>> tested it but after two rings on the local account the line just got
>> disconnected. Is there a parameter that extends the number of rings
>> before it gets rejected?
>>
>> "Julian F. Tasis, III" <jf...@id...> wrote:
>>   Hi,
>>
>> Thanks for the reply. I transfer the call back to the IVR when it is
>> rejected and just added some internal header and it worked.
>>
>> By the way I modify the IVRTransferCall a little bit so that when the
>> media server is routing the call to B's destination, B will see the
>> profile of the caller and not the media server.
>>
>> May I know how did you record all the wav files OpenSBC is using? I
>> tried
>> recording my own wav files but I can't hear my recordings hehe.
>>
>> Regards,
>> Julian F. Tasis, III
>>
>> Re: [OpenSIPStack] IVR handling on transfer call rejection
>> From: - 2007-08-15 03:30
>> There is more work to be done before you will be able to do this. Ryan
>> is working on improvements to the API to allow the transfered
>> connection to reconnect to the IVR once the outbound call is
>> disconnected. Watch the updates.
>>
>>
>> Julian F. Tasis, III wrote:
>>
>>> Good day,
>>>
>>> Hi, I was playing around with the IVRHandler lately and getting to know
>>> how it works. Just got a problem when transferring a call to B
>>> destination. If the B rejects the call, I can't notify A that the call
>>> is
>>> dropped. I wanted to play also a specific speil to A to notify it. I
>>> tried
>>> getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but
>>> with no success. Is there any other way around to do that?
>>>
>>>
>>> Regards,
>>> Julian F. Tasis, III
>>>
>>
>>
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems? Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >> http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>
>> ---------------------------------
>> Pinpoint customers who are looking for what you sell.
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems?  Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >>  http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>
>


Regards,
Julian F. Tasis, III
Product Planning and Development

Ideawurx Inc.
7th Floor, V.A. Rufino Bldg.,
6784 Ayala Ave.
Makati City, Philippines 1226

Tel: +(632) 8111530
Fax: +(632) 8189767

jf...@id...
http://www.ideawurx.com.ph

Ideawurx is a telecommunications solutions provider that develops and
markets a range of customized computer telephony integration ( CTI )
solutions.

"Our IDEAS will WORK for YOU!"