22 Aug
2012
22 Aug
'12
4:26 p.m.
Hi All Kamailio Experts,
I have configured Kamailio (kamailio 3.1.5) as media server. All things
are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
Calls. For this purpose i have followed the web page :
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
In this page, some points are not clear for me , as given below:
(1) In case you use *sipregs* you have to create a record for each
extension where to set the 'name' to value of 'name' from *sipusers*.
The rest is populated by Asterisk from registrations.
(2) Be sure you configure Asterisk *to not authenticate* SIP requests
coming from Kamailio.
I am not sure that my local users chat is working through kamailio or
asterisk, who is used for authorization.
Any specific Web page to correct the issue will highly appreciated
according to my scenario.
Kindly guide me. Thanks in advance.
--
Best Regards,
Vijay Thakur
(Assistant Manager - Networks)
Mobile : +91 8744018065
Mail : vijay.thakur(a)loopmethods.com
Loop IT Methods Private Limited
1st Floor, B-10, Sector-7, Noida, (U.P) India
Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS)
Fax: +91 971 728 330
Web: www.loopmethods.com
LOOP Disclaimer
-------------------------------------------------------------------------------------------------
This message (including any attachments) contains confidential information intended for a
specific individual and purpose, and is protected by law. If you are not the intended
recipient, you should delete this message and are hereby notified that any disclosure,
copying, or distribution of this message, or the taking of any action based on it, is
strictly prohibited.
-----------------------------------------------------------------------------------------------------------------
23 Aug
23 Aug
9:54 a.m.
On 22.08.2012 14:26, Vijay Thakur wrote:
Hi All Kamailio Experts,
I have configured Kamailio (kamailio 3.1.5) as media server.
Kamailio is a SIP proxy, not a media server. Maybe you mean that you are
using Kamailio with rtpproxy as media relay.
All things
are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
Calls. For this purpose i have followed the web page :
If you wan to you Asterisk as PSTN gateway only, then there is no need
to follow this tutorial. This tutorial makes strong integration of
Kamailio and Asterisk. For PSTN gateway functionality there is no need
to integrate Kamailio and Asterisk - just configure Asterisk as gateway
and forwards PSTN calls from Kamailio to Asterisk (and vice versa)
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
In this page, some points are not clear for me , as given below:
(1) In case you use *sipregs* you have to create a record for each
extension where to set the 'name' to value of 'name' from *sipusers*.
The rest is populated by Asterisk from registrations.
(2) Be sure you configure Asterisk *to not authenticate* SIP requests
coming from Kamailio.
I am not sure that my local users chat is working through kamailio or
asterisk, who is used for authorization.
What do you mean with "not sure"? For instant messaging between users
there is no need to use Asterisk.
In above setup the authentication is done by Kamailio only.
regards
Klaus
Any specific Web page to correct the issue will highly
appreciated
according to my scenario.
Kindly guide me. Thanks in advance.
--
Best Regards,
Vijay Thakur
(Assistant Manager - Networks)
Mobile : +91 8744018065
Mail :vijay.thakur@loopmethods.com
Loop IT Methods Private Limited
1st Floor, B-10, Sector-7, Noida, (U.P) India
Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS)
Fax: +91 971 728 330
Web:www.loopmethods.com
LOOP Disclaimer
-------------------------------------------------------------------------------------------------
This message (including any attachments) contains confidential information intended for a
specific individual and purpose, and is protected by law. If you are not the intended
recipient, you should delete this message and are hereby notified that any disclosure,
copying, or distribution of this message, or the taking of any action based on it, is
strictly prohibited.
-----------------------------------------------------------------------------------------------------------------
_______________________________________________
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sr-users(a)lists.sip-router.org
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10:31 a.m.
Thanks for clearing the doubts. You are very right, i am using kamailio
as Media Relay.
Can you send me some specific document URL, from where i can configure
Asterisk as PSTN Gateway.
Can we set Kamailio and Asterisk in one server.
Thanks in advance.
Vijay
Thursday 23 August 2012 11:24 AM, Klaus Darilion wrote:
On 22.08.2012 14:26, Vijay Thakur wrote:
Hi All Kamailio Experts,
I have configured Kamailio (kamailio 3.1.5) as media server.
Kamailio is a SIP proxy, not a media server. Maybe you mean that you
are using Kamailio with rtpproxy as media relay.
All things
are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
Calls. For this purpose i have followed the web page :
If you wan to you Asterisk as PSTN gateway only, then there is no need
to follow this tutorial. This tutorial makes strong integration of
Kamailio and Asterisk. For PSTN gateway functionality there is no need
to integrate Kamailio and Asterisk - just configure Asterisk as
gateway and forwards PSTN calls from Kamailio to Asterisk (and vice
versa)
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
In this page, some points are not clear for me , as given below:
(1) In case you use *sipregs* you have to create a record for each
extension where to set the 'name' to value of 'name' from *sipusers*.
The rest is populated by Asterisk from registrations.
(2) Be sure you configure Asterisk *to not authenticate* SIP requests
coming from Kamailio.
I am not sure that my local users chat is working through kamailio or
asterisk, who is used for authorization.
What do you mean with "not sure"? For instant messaging between users
there is no need to use Asterisk.
In above setup the authentication is done by Kamailio only.
regards
Klaus
> Any specific Web page to correct the issue will highly appreciated
> according to my scenario.
>
> Kindly guide me. Thanks in advance.
>
> --
> Best Regards,
>
> Vijay Thakur
> (Assistant Manager - Networks)
> Mobile : +91 8744018065
> Mail :vijay.thakur@loopmethods.com
>
> Loop IT Methods Private Limited
> 1st Floor, B-10, Sector-7, Noida, (U.P) India
> Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011
> 178 (AUS)
> Fax: +91 971 728 330
> Web:www.loopmethods.com
>
> LOOP Disclaimer
>
-------------------------------------------------------------------------------------------------
> This message (including any attachments) contains confidential
> information intended for a specific individual and purpose, and is
> protected by law. If you are not the intended recipient, you should
> delete this message and are hereby notified that any disclosure,
> copying, or distribution of this message, or the taking of any action
> based on it, is strictly prohibited.
>
-----------------------------------------------------------------------------------------------------------------
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
1:42 p.m.
On 23.08.2012 08:31, Vijay Thakur wrote:
Thanks for clearing the doubts. You are very right, i
am using kamailio
as Media Relay.
Can you send me some specific document URL, from where i can configure
Asterisk as PSTN Gateway.
There is no such document. But configuring a PSTN gateway is already in
the default configuration file. Just search in the deafult configuration
file for "WITH_PSTN".
Can we set Kamailio and Asterisk in one server.
Yes, thats no problem. Either use 2 IP addresses on the same server, one
for Kamailio and one for Asterisk, or use the same IP address and
different ports.
regards
Klaus
PS: If you are building a public SIP service it is a good idea to not
use the default port 5060 to get rid of SIP port scanners.
Thanks in advance.
Vijay
Thursday 23 August 2012 11:24 AM, Klaus Darilion wrote:
On 22.08.2012 14:26, Vijay Thakur wrote:
Hi All Kamailio Experts,
I have configured Kamailio (kamailio 3.1.5) as media server.
Kamailio is a SIP proxy, not a media server. Maybe you mean that you
are using Kamailio with rtpproxy as media relay.
All things
are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound
Calls. For this purpose i have followed the web page :
If you wan to you Asterisk as PSTN gateway only, then there is no need
to follow this tutorial. This tutorial makes strong integration of
Kamailio and Asterisk. For PSTN gateway functionality there is no need
to integrate Kamailio and Asterisk - just configure Asterisk as
gateway and forwards PSTN calls from Kamailio to Asterisk (and vice
versa)
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb.
In this page, some points are not clear for me , as given below:
(1) In case you use *sipregs* you have to create a record for each
extension where to set the 'name' to value of 'name' from *sipusers*.
The rest is populated by Asterisk from registrations.
(2) Be sure you configure Asterisk *to not authenticate* SIP requests
coming from Kamailio.
I am not sure that my local users chat is working through kamailio or
asterisk, who is used for authorization.
What do you mean with "not sure"? For instant messaging between users
there is no need to use Asterisk.
In above setup the authentication is done by Kamailio only.
regards
Klaus
> Any specific Web page to correct the issue will highly appreciated
> according to my scenario.
>
> Kindly guide me. Thanks in advance.
>
> --
> Best Regards,
>
> Vijay Thakur
> (Assistant Manager - Networks)
> Mobile : +91 8744018065
> Mail :vijay.thakur@loopmethods.com
>
> Loop IT Methods Private Limited
> 1st Floor, B-10, Sector-7, Noida, (U.P) India
> Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011
> 178 (AUS)
> Fax: +91 971 728 330
> Web:www.loopmethods.com
>
> LOOP Disclaimer
>
-------------------------------------------------------------------------------------------------
> This message (including any attachments) contains confidential
> information intended for a specific individual and purpose, and is
> protected by law. If you are not the intended recipient, you should
> delete this message and are hereby notified that any disclosure,
> copying, or distribution of this message, or the taking of any action
> based on it, is strictly prohibited.
>
-----------------------------------------------------------------------------------------------------------------
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
4460
days inactive
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days old
3 comments
2 participants
participants (2)
-
Klaus Darilion -
Vijay Thakur