Hi,I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running. What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that?Thank you!
Hello,
is your trunk provider requiring a username/password for the calls sent to it, or it is just IP based peering?
Cheers, Daniel
On 12/10/12 4:52 PM, andre second wrote:
Hi, I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running.
What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that? Thank you!
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi.Thanks for you reply!Provider is using Login and Password combination for making calls. --- On Thu, 12/13/12, Daniel-Constantin Mierla miconda@gmail.com wrote:
From: Daniel-Constantin Mierla miconda@gmail.com Subject: Re: [SR-Users] From sip phone to provider trunk To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" sr-users@lists.sip-router.org Cc: "andre second" andrei.beliy@yahoo.com Date: Thursday, December 13, 2012, 11:14 AM
Hello,
is your trunk provider requiring a username/password for the calls sent to it, or it is just IP based peering?
Cheers,
Daniel
On 12/10/12 4:52 PM, andre second wrote:
Hi, I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running.
What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that? Thank you!
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Still no success.Shall I use auth module maybe?
--- On Fri, 12/14/12, andre second andrei.beliy@yahoo.com wrote:
From: andre second andrei.beliy@yahoo.com Subject: Re: [SR-Users] From sip phone to provider trunk To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" sr-users@lists.sip-router.org, miconda@gmail.com Date: Friday, December 14, 2012, 9:22 AM
Hi.Thanks for you reply!Provider is using Login and Password combination for making calls. --- On Thu, 12/13/12, Daniel-Constantin Mierla miconda@gmail.com wrote:
From: Daniel-Constantin Mierla miconda@gmail.com Subject: Re: [SR-Users] From sip phone to provider trunk To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" sr-users@lists.sip-router.org Cc: "andre second" andrei.beliy@yahoo.com Date: Thursday, December 13, 2012, 11:14 AM
Hello,
is your trunk provider requiring a username/password for the calls sent to it, or it is just IP based peering?
Cheers,
Daniel
On 12/10/12 4:52 PM, andre second wrote:
Hi, I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running.
What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that? Thank you!
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
-----Inline Attachment Follows-----
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
look at uac module with its authentication functions. But be aware of its limitations. A proxy cannot authenticate fully RFC compliant in behalf of users, for that you need a b2bua -- however, there is a chance that works with uac workaround.
Cheers, Daniel
On 12/17/12 7:14 PM, andre second wrote:
Still no success. Shall I use auth module maybe?
--- On *Fri, 12/14/12, andre second /andrei.beliy@yahoo.com/* wrote:
From: andre second <andrei.beliy@yahoo.com> Subject: Re: [SR-Users] From sip phone to provider trunk To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users@lists.sip-router.org>, miconda@gmail.com Date: Friday, December 14, 2012, 9:22 AM Hi. Thanks for you reply! Provider is using Login and Password combination for making calls. --- On *Thu, 12/13/12, Daniel-Constantin Mierla /<miconda@gmail.com>/* wrote: From: Daniel-Constantin Mierla <miconda@gmail.com> Subject: Re: [SR-Users] From sip phone to provider trunk To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users@lists.sip-router.org> Cc: "andre second" <andrei.beliy@yahoo.com> Date: Thursday, December 13, 2012, 11:14 AM Hello, is your trunk provider requiring a username/password for the calls sent to it, or it is just IP based peering? Cheers, Daniel On 12/10/12 4:52 PM, andre second wrote:Hi, I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running. What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that? Thank you! _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda -----Inline Attachment Follows----- _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org </mc/compose?to=sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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andre second -
Daniel-Constantin Mierla