*
>
> http://kb.asipto.com/asterisk:index
>
> ** **
>
> Good luck.
>
> ** **
>
> *From:* sr-users-boun...@lists.sip-router.org [mailto:
> sr-users-boun...@lists.sip-router.org] *On Behalf Of *Nishar M.H
> *Sent:* maandag 19 augustus 2013 6:40
> *To:* Kama
gt; And Asterisk do other tasks.
>
>
> On Sun, Aug 18, 2013 at 1:25 PM, Nishar M.H wrote:
>
>> Do you have cfg file that is connecting with Asterisk ?
>> As per my previous post in this forum i have created Trunk in Asterisk
>> and it shows registered.
nderstand the
> configuration file and its flow.
>
> --
> Sammy
>
> sent from a little smart phone
> On Aug 18, 2013 3:08 PM, "Nishar M.H" wrote:
>
>> Thanks sammy.
>>
>> But when i added
>>
>>
>> $rU=~&q
Asterisk Server
is connected or not ?
Regards,
Nishar Hamsa
On Sun, Aug 18, 2013 at 2:20 PM, Salman Zafar wrote:
> You may have to *restart *Kamailio service to make your new changes
> implemented from cfg.
>
>
> On Sun, Aug 18, 2013 at 3:16 PM, Nishar M.H wrote:
>
8/18/2013 04:59 AM, Nishar M.H wrote:
>
> I am comfortable with Asterisk.
>> Here i am facing the issues with kamailio. Last few days i have been
>> struggling in kamailio integration with asterisk.
>> http://lists.sip-router.org/**pipermail/sr-users/2013-**
>> Augu
hing.
>
> --
> Sammy
>
> sent from a little smart phone
> On Aug 18, 2013 10:13 AM, "Nishar M.H" wrote:
>
>> HI,
>>
>> I have used *if**(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")**)* and now i am
>> able to start the kamailio using kamctl sta
Thanks.
On Sun, Aug 18, 2013 at 1:33 PM, Alex Balashov wrote:
> Nishar,
>
>
> On 08/18/2013 04:59 AM, Nishar M.H wrote:
>
> I am comfortable with Asterisk.
>> Here i am facing the issues with kamailio. Last few days i have been
>> struggling in kamailio inte
routing etc
>
> SIP-Fundamentals are prerequisite you must understand a little more than
> basic SIP technology knowledge to get started.
>
>
> On Sun, Aug 18, 2013 at 12:50 PM, Nishar M.H wrote:
>
>> Hi,
>>
>> How can i learn *kamai
Hi,
How can i learn *kamailio.cfg* basics?
Is there any document for v4.0
--
Thanks & Regards,
*Nishar Hamsa
*
Sat, Aug 17, 2013 at 1:06 PM, SamyGo wrote:
> Check regular expression as the logs state. Modify with correct one.
>
> sent from a little smart phone
> On Aug 17, 2013 2:03 PM, "Nishar M.H" wrote:
>
>> HI,
>>
>> I have followed the same.
>&
where a user location DB
> search is made and an online user is found for the dialed destination and a
> successful call will be established.
>
> I think that should be all for now.
>
> BR,
> Sammy
>
>
>
>
>
> On Fri, Aug 16, 2013 at 4:06 AM, Victor V. Kustov wr
r the route[LOCATION] where a user location DB
> search is made and an online user is found for the dialed destination and a
> successful call will be established.
>
> I think that should be all for now.
>
> BR,
> Sammy
>
>
>
>
>
> On Fri, Aug 16, 2013 at 4:06 A
:30 PM, Nishar M.H wrote:
> Hi,
>
> I have Two Servers.
>
> Asterisk PBX Server : 192.168.20.196
> Kamailio Server : 192.168.20.208
> No. of users created in Kamailio : 10
> Using command : kamctl add 1000 12345
>
> Here i am using Kamailio as SIP Server and Asterisk as
registrations then
> you need to create a trunk in Asterisk and write corresponding dialplan in
> asterisk to receive calls from Kamailio;process them; and dial back out to
> kamailio.
>
> BR,
> Sammy
>
> On Thu, Aug 15, 2013 at 5:05 AM, Nishar M.H wrote:
>
>> Hi Sammy
Hamsa.
On Thu, Aug 15, 2013 at 4:05 PM, Nishar M.H wrote:
> Hi Sammy,
>
> Thanks for support. What are the changes do i have to make id DB here.
> One more thing i have to ask you is how the Asterisk communicate with
> Kamailio SIP users.
> Do i have to create SIP trunk in Aste
ot; according to your OS and see why starting of kamailio fails.
>
> BR,
> Sammy
>
>
>
> On Thu, Aug 15, 2013 at 3:37 AM, Nishar M.H wrote:
>
>>
>> Hi,
>>
>> I am Trying to make a Seperate SIP Server (Kamailio) and Asterisk PBX
>> Server.
>>
Hi,
I am Trying to make a Seperate SIP Server (Kamailio) and Asterisk PBX
Server.
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
But in Above link the issue is actually both are in same server. But I need
to make seperate server SIP Authentication and Asterisk.
I tri
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