Normal Call clearning

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Thanks for your reply, but?my question is the same?sip server is working for other soft phone, not for pjsip?sample...? :(

?

Have a look @ my small web-page:
http://www.geocities.com/muki_champs

Regards, 
Mukesh Kumar, 
Sr.Software Engineer,
Mobile Application Developer.
Hyderabad. 
India. 
+91-9397845485 (M)?





________________________________
From: Gang Liu <gangban.lau@xxxxxxxxx>
To: pjsip list <pjsip at lists.pjsip.org>
Sent: Tuesday, January 13, 2009 12:48:25 PM
Subject: Re: Normal Call clearning

so you need check your sip server log file.Then you will know the reason.
There are so many buggy sip servers.
When I first time try to use pjsip app to talk to my company callcenter, there are many issues.After trace the sip proxy server log file, we found all are because of SIP proxy buggy implementation.It works well now after we change to another SIPProxy provider.

regards,
Gang


On Tue, Jan 13, 2009 at 12:25 PM, Mukesh Srivastav <muki_champs at yahoo.com> wrote:

yes, i am getting the response as Bye from the sip server, but?my question, why it is not happening for other sip servers for the same?apps, i didnt do any single line of code change.

Any thoughts.
?

Have a look @ my small web-page:
http://www.geocities.com/muki_champs

Regards, 
Mukesh Kumar, 
Sr.Software Engineer,
Mobile Application Developer.
Hyderabad. 
India. 
+91-9397845485 (M)?





________________________________
From: Gang Liu <gangban.lau@xxxxxxxxx>
To: pjsip list <pjsip at lists.pjsip.org>
Sent: Tuesday, January 13, 2009 9:52:54 AM
Subject: Re: Normal Call clearning


you need look into your sip server to find the answer.For example your sip server may disconnect this call when codec not matched by your server config.

regards,
Gang


On Tue, Jan 13, 2009 at 2:27 AM, Mukesh Srivastav <muki_champs at yahoo.com> wrote:

Hi,
?i am trying to?tes the pjsip apps,?but my?server always respond with the message, "Normal Call Clearning", i have tested the same sip server with other sipsoftphones, it is working, but i try with the pjsip code, i am getting the response as "Normal Call Clearning", 

any thoughts about it ?
?


Thanks,
Muki

_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip at lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org




_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip at lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090112/5848bf69/attachment.html>


[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux