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 Asterisk ?

Regards,

Nishar Hamsa.

On Thu, Aug 15, 2013 at 3:49 PM, SamyGo govoiper@gmail.com wrote:

Hi Nishar,

Once you've separate Asterisk and Kamailio installation you'll need to modify the variables and DB parameters in kamailio.cfg. Simply copying the configurations file may give you errors. Please see the log files "syslog" or "messages" 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 nisharmh85@gmail.com wrote:

...

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 tried to follow the the link step by step. But issue is actually, i made some changes to make separate server for SIP server and PBX. But issue that i am facing is when i copied kamailio.cfg from the link, I tried to start kamctl after that but i am got :

INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed

Can you help me out. ?

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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 again,

Please explain the whole situation and your desired setup. Once the whole picture is clear then anyone will be able to guide you. As I can imagine you might need to change the DBURL mysql://XXX:XXX@DBSERVER/DBNAME string in your kamailio.cfg ensure that the Kamailio server access the DB

The other thing you might need to consider is changing the IP addresses for Asterisk. And if you are not using Asterisk as REGISTRAR server,or not using SIP Realtime, and you just need kamailio to handle 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 nisharmh85@gmail.com 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 Asterisk ?

Regards,

Nishar Hamsa.

On Thu, Aug 15, 2013 at 3:49 PM, SamyGo govoiper@gmail.com wrote:

...

Hi Nishar,

Once you've separate Asterisk and Kamailio installation you'll need to modify the variables and DB parameters in kamailio.cfg. Simply copying the configurations file may give you errors. Please see the log files "syslog" or "messages" 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 nisharmh85@gmail.com wrote:

...

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 tried to follow the the link step by step. But issue is actually, i made some changes to make separate server for SIP server and PBX. But issue that i am facing is when i copied kamailio.cfg from the link, I tried to start kamctl after that but i am got :

INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed

Can you help me out. ?

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

Design :

5 Buildings and Data Center

Extensions :

Building 1: 1000 to 1010 Building 2: 2000 to 2010 Building 3: 3000 to 3010 Building 4: 4000 to 4010 Building 5: 5000 to 5010

In Data Center I going to keep the both Asterisk and Kamailio

Outside Calls are going through Asterisk Server. In Asterisk Server i have installed Telephonic Card. The Dialout : will be like _X. in Asterisk For the internal call I want use Kamailio Server. In each location IP Phone will be there

I have Installed Kamailio and Asterisk PBX in Separate Server.

Regards,

Nishar Hamsa.

On Thu, Aug 15, 2013 at 4:30 PM, Nishar M.H nisharmh85@gmail.com 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 media. In Asterisk Server i am going to use Digium Telephonic Card. So all calls will be going through Asterisk.

So if any kamailio SIP user Trying call Outside then It should go through Asterisk.

This Is my Setup. Right i want test with out Realtime asterisk. What are the changes and configurations i have to do here ?

Thanks.

Regards,

Nishar Hamsa.

On Thu, Aug 15, 2013 at 4:15 PM, SamyGo govoiper@gmail.com wrote:

...

Hi again,

Please explain the whole situation and your desired setup. Once the whole picture is clear then anyone will be able to guide you. As I can imagine you might need to change the DBURL mysql://XXX:XXX@DBSERVER/DBNAME string in your kamailio.cfg ensure that the Kamailio server access the DB

The other thing you might need to consider is changing the IP addresses for Asterisk. And if you are not using Asterisk as REGISTRAR server,or not using SIP Realtime, and you just need kamailio to handle 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 nisharmh85@gmail.com 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 Asterisk ?

Regards,

Nishar Hamsa.

On Thu, Aug 15, 2013 at 3:49 PM, SamyGo govoiper@gmail.com wrote:

...

Hi Nishar,

Once you've separate Asterisk and Kamailio installation you'll need to modify the variables and DB parameters in kamailio.cfg. Simply copying the configurations file may give you errors. Please see the log files "syslog" or "messages" 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 nisharmh85@gmail.comwrote:

...

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 tried to follow the the link step by step. But issue is actually, i made some changes to make separate server for SIP server and PBX. But issue that i am facing is when i copied kamailio.cfg from the link, I tried to start kamctl after that but i am got :

INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed

Can you help me out. ?

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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

--

Thanks & Regards,

---

*Nishar Hamsa

---

Dear Nishar,

From the mentioned URL the configurations need the following changes.

On the top start on configs you need to use: WITH_PSTN and might not need WITH_ASTERISK

#!KAMAILIO #!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_PSTN

The insert your asterisk IP in the following line:

pstn.gw_ip = "<AsteriskIPHere>" desc "PSTN GW Address"

Next your route[LOCATION] is called after the route[PSTN] from default config file so you'll have to put proper regexp/conditions to avoid your regular Extension to Extension calls to go out to PSTN-Asterisk.

Now in route[PSTN] add another condition to return from PSTN route when a local extension is dialed.

if(!($rU=~"^(+|00)[1-9][0-9]{3,20}$") || $rU=~"^([1-5]01[0-9]$"))

return;

That will ensure that your regular extensions set don't route out to PSTN asterisk and infact will enter 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 AM, Victor V. Kustov coyote@bks.tv wrote:

Hi, Nishar M.H!

...

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed

see log for details. do not use dumb copy/paste for kamailio config.

-- WBR, Victor I use FREE operation system: 3.10.4-calculate GNU/Linux

---

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,

I have followed the same.

When i run :

*kamctl start*

I got the same error like :

*INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed*

Here is the log file :

Aug 17 12:57:31 kamailio-VirtualBox kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp: using epoll_lt as the io watch method (auto detected) Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: rr [../outbound/api.h:49]: Failed to import bind_ob Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: ERROR: <core> [rvalue.c:3026]: Bad regular expression "^([1-5]01[0-9]$"(853,50-853,66) Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: ERROR: <core> [route.c:1214]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:857

Some issues is there with the pattern.

Regards,

Nishar Hamsa

On Sat, Aug 17, 2013 at 11:29 AM, SamyGo govoiper@gmail.com wrote:

Dear Nishar,

From the mentioned URL the configurations need the following changes.

On the top start on configs you need to use: WITH_PSTN and might not need WITH_ASTERISK

#!KAMAILIO #!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_PSTN

The insert your asterisk IP in the following line:

pstn.gw_ip = "<AsteriskIPHere>" desc "PSTN GW Address"

Next your route[LOCATION] is called after the route[PSTN] from default config file so you'll have to put proper regexp/conditions to avoid your regular Extension to Extension calls to go out to PSTN-Asterisk.

Now in route[PSTN] add another condition to return from PSTN route when a local extension is dialed.

if(!($rU=~"^(+|00)[1-9][0-9]{3,20}$") || $rU=~"^([1-5]01[0-9]$"))

return;

That will ensure that your regular extensions set don't route out to PSTN asterisk and infact will enter 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 AM, Victor V. Kustov coyote@bks.tv wrote:

...

Hi, Nishar M.H!

...

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed

see log for details. do not use dumb copy/paste for kamailio config.

-- WBR, Victor I use FREE operation system: 3.10.4-calculate GNU/Linux

---

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

---

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,

I have used *if**(!($rU=~"^(+|00)[1-9][0-9]{3,20}$")**)* and now i am able to start the kamailio using kamctl start

Here is the log below.:

Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3773]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Aug 18 08:28:54 kamailio-VirtualBox /usr/local/sbin/kamailio[3771]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command domain_dump is not available Aug 18 08:49:13 kamailio-VirtualBox /usr/local/sbin/kamailio[3764]: NOTICE: acc [acc.c:275]: ACC: call missed: timestamp=1376801353;method=INVITE;from_tag=ae70066a;to_tag=e90f3c6b;call_id=NmEwMzE3ZDBkNGEwZmRhODMwNDMzZDNkZmVjNGVmODM.;code=486;reason=Busy Here;src_user=1000;src_domain=192.168.20.208;src_ip=192.168.20.186;dst_ouser=1000;dst_user=1000;dst_domain=192.168.20.186

I am able to call sip to sip for internal purpose.

In Asterisk Server i have created a trunk and after the command "sip show peers" it show like below : localhost*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description Realtime *kamailio 192.168.20.208 5060 OK (1 ms)* 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]

Now how can i test the calls are going through Asterisk ?

Thanks in advance.

Regards,

Nishar Hamsa

On Sat, Aug 17, 2013 at 1:06 PM, SamyGo govoiper@gmail.com 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" nisharmh85@gmail.com wrote:

...

HI,

I have followed the same.

When i run :

*kamctl start*

I got the same error like :

*INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed*

Here is the log file :

Aug 17 12:57:31 kamailio-VirtualBox kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp: using epoll_lt as the io watch method (auto detected) Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: rr [../outbound/api.h:49]: Failed to import bind_ob Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: ERROR: <core> [rvalue.c:3026]: Bad regular expression "^([1-5]01[0-9]$"(853,50-853,66) Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: ERROR: <core> [route.c:1214]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:857

Some issues is there with the pattern.

Regards,

Nishar Hamsa

On Sat, Aug 17, 2013 at 11:29 AM, SamyGo govoiper@gmail.com wrote:

...

Dear Nishar,

From the mentioned URL the configurations need the following changes.

On the top start on configs you need to use: WITH_PSTN and might not need WITH_ASTERISK

#!KAMAILIO #!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_PSTN

The insert your asterisk IP in the following line:

pstn.gw_ip = "<AsteriskIPHere>" desc "PSTN GW Address"

Next your route[LOCATION] is called after the route[PSTN] from default config file so you'll have to put proper regexp/conditions to avoid your regular Extension to Extension calls to go out to PSTN-Asterisk.

Now in route[PSTN] add another condition to return from PSTN route when a local extension is dialed.

if(!($rU=~"^(+|00)[1-9][0-9]{3,20}$") || $rU=~"^([1-5]01[0-9]$"))

return;

That will ensure that your regular extensions set don't route out to PSTN asterisk and infact will enter 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 AM, Victor V. Kustov coyote@bks.tv wrote:

...

Hi, Nishar M.H!

...

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed

see log for details. do not use dumb copy/paste for kamailio config.

-- WBR, Victor I use FREE operation system: 3.10.4-calculate GNU/Linux

---

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

---

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

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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

Thanks sammy.

But when i added

$rU=~"^([1-5]01[0-9]$"

into that code it shows some errors. Thats why remove that line and used *if **(!($rU=~"^(+|00)[1-9][0-9]{3,20}$")**) *there.*

* * * Now,

I have created users in :

Asterisk Server (192.168.20.196) : 5555 Kamailio Server (192.168.20.208) : 1000

If i want to communicate from 5555 to 1000, what are the customization require on both server ?

Like *5555 <------> 1000*

How can i get the user events in asterisk that i have created in kamailio server* ? *

*

* *Regards,

* *Nishar Hamsa. * * * * *

On Sun, Aug 18, 2013 at 1:39 PM, SamyGo govoiper@gmail.com wrote:

Since you've removed the second regex then only way to make call reach to Asterisk via PSTN route is when you dial anything that matches the mentioned regexp in your email I.e 001456223456 and look at your Asterisk cli for anything.

-- Sammy

sent from a little smart phone On Aug 18, 2013 10:13 AM, "Nishar M.H" nisharmh85@gmail.com 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 start

Here is the log below.:

Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3773]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Aug 18 08:28:54 kamailio-VirtualBox /usr/local/sbin/kamailio[3771]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command domain_dump is not available Aug 18 08:49:13 kamailio-VirtualBox /usr/local/sbin/kamailio[3764]: NOTICE: acc [acc.c:275]: ACC: call missed: timestamp=1376801353;method=INVITE;from_tag=ae70066a;to_tag=e90f3c6b;call_id=NmEwMzE3ZDBkNGEwZmRhODMwNDMzZDNkZmVjNGVmODM.;code=486;reason=Busy Here;src_user=1000;src_domain=192.168.20.208;src_ip=192.168.20.186;dst_ouser=1000;dst_user=1000;dst_domain=192.168.20.186

I am able to call sip to sip for internal purpose.

In Asterisk Server i have created a trunk and after the command "sip show peers" it show like below : localhost*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description Realtime *kamailio 192.168.20.208 5060 OK (1 ms)* 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]

Now how can i test the calls are going through Asterisk ?

Thanks in advance.

Regards,

Nishar Hamsa

On Sat, Aug 17, 2013 at 1:06 PM, SamyGo govoiper@gmail.com 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" nisharmh85@gmail.com wrote:

...

HI,

I have followed the same.

When i run :

*kamctl start*

I got the same error like :

*INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed*

Here is the log file :

Aug 17 12:57:31 kamailio-VirtualBox kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp: using epoll_lt as the io watch method (auto detected) Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: rr [../outbound/api.h:49]: Failed to import bind_ob Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: ERROR: <core> [rvalue.c:3026]: Bad regular expression "^([1-5]01[0-9]$"(853,50-853,66) Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: ERROR: <core> [route.c:1214]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:857

Some issues is there with the pattern.

Regards,

Nishar Hamsa

On Sat, Aug 17, 2013 at 11:29 AM, SamyGo govoiper@gmail.com wrote:

...

Dear Nishar,

From the mentioned URL the configurations need the following changes.

On the top start on configs you need to use: WITH_PSTN and might not need WITH_ASTERISK

#!KAMAILIO #!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_PSTN

The insert your asterisk IP in the following line:

pstn.gw_ip = "<AsteriskIPHere>" desc "PSTN GW Address"

Next your route[LOCATION] is called after the route[PSTN] from default config file so you'll have to put proper regexp/conditions to avoid your regular Extension to Extension calls to go out to PSTN-Asterisk.

Now in route[PSTN] add another condition to return from PSTN route when a local extension is dialed.

if(!($rU=~"^(+|00)[1-9][0-9]{3,20}$") || $rU=~"^([1-5]01[0-9]$"))

return;

That will ensure that your regular extensions set don't route out to PSTN asterisk and infact will enter 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 AM, Victor V. Kustov coyote@bks.tvwrote:

...

Hi, Nishar M.H!

> >ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio >start failed

see log for details. do not use dumb copy/paste for kamailio config.

-- WBR, Victor I use FREE operation system: 3.10.4-calculate GNU/Linux

---

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

---

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

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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

Thanks Sammy. Really Sorry for that. until now i have never used Kamailio or other SIP servers. I am trying to learn that.

Once again thanks for your support.

Regards,

Nishar Hamsa

On Sun, Aug 18, 2013 at 2:40 PM, SamyGo govoiper@gmail.com wrote:

Oh Dear Nishar,

Please google and see that there is an error in the regexp which gives error..don't you see an extra ( on the start..make a regexp yourself..what's the big deal?

I expect that at the very least you can write dial plan in asterisk to dial call TO kamailio peer when user 5XXX dials 1XXX or you need help in there too? If that is OK then you may need to at least tell your Kamailio to accept calls from asterisk source ip:port Please look at the route[FROM_ASTERISK] or enable WITH_IPAUTH and put IP&Port address of Asterisk in address table to get past the Authentication for calls coming in from asterisk to kamailio.

Please do some mailing list searching and working on your part as well before asking mailing list to give exact working solutions for your requirements.

Daniels' blog are THE BEST thing you will ever find to understand the configuration file and its flow.

-- Sammy

sent from a little smart phone On Aug 18, 2013 3:08 PM, "Nishar M.H" nisharmh85@gmail.com wrote:

...

Thanks sammy.

But when i added

$rU=~"^([1-5]01[0-9]$"

into that code it shows some errors. Thats why remove that line and used *if**(!($rU=~"^(+|00)[1-9][0-9]{3,20}$")**) *there.*

Now,

I have created users in :

Asterisk Server (192.168.20.196) : 5555 Kamailio Server (192.168.20.208) : 1000

If i want to communicate from 5555 to 1000, what are the customization require on both server ?

Like *5555 <------> 1000*

How can i get the user events in asterisk that i have created in kamailio server* ?

*Regards,

*Nishar Hamsa.

On Sun, Aug 18, 2013 at 1:39 PM, SamyGo govoiper@gmail.com wrote:

...

Since you've removed the second regex then only way to make call reach to Asterisk via PSTN route is when you dial anything that matches the mentioned regexp in your email I.e 001456223456 and look at your Asterisk cli for anything.

-- Sammy

sent from a little smart phone On Aug 18, 2013 10:13 AM, "Nishar M.H" nisharmh85@gmail.com 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 start

Here is the log below.:

Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3757]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 18 08:24:45 kamailio-VirtualBox /usr/local/sbin/kamailio[3773]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Aug 18 08:28:54 kamailio-VirtualBox /usr/local/sbin/kamailio[3771]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command domain_dump is not available Aug 18 08:49:13 kamailio-VirtualBox /usr/local/sbin/kamailio[3764]: NOTICE: acc [acc.c:275]: ACC: call missed: timestamp=1376801353;method=INVITE;from_tag=ae70066a;to_tag=e90f3c6b;call_id=NmEwMzE3ZDBkNGEwZmRhODMwNDMzZDNkZmVjNGVmODM.;code=486;reason=Busy Here;src_user=1000;src_domain=192.168.20.208;src_ip=192.168.20.186;dst_ouser=1000;dst_user=1000;dst_domain=192.168.20.186

I am able to call sip to sip for internal purpose.

In Asterisk Server i have created a trunk and after the command "sip show peers" it show like below : localhost*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description Realtime *kamailio 192.168.20.208 5060 OK (1 ms)* 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]

Now how can i test the calls are going through Asterisk ?

Thanks in advance.

Regards,

Nishar Hamsa

On Sat, Aug 17, 2013 at 1:06 PM, SamyGo govoiper@gmail.com 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" nisharmh85@gmail.com wrote:

...

HI,

I have followed the same.

When i run :

*kamctl start*

I got the same error like :

*INFO: Starting Kamailio :

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed*

Here is the log file :

Aug 17 12:57:31 kamailio-VirtualBox kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp: using epoll_lt as the io watch method (auto detected) Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: rr [../outbound/api.h:49]: Failed to import bind_ob Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) support disabled Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: ERROR: <core> [rvalue.c:3026]: Bad regular expression "^([1-5]01[0-9]$"(853,50-853,66) Aug 17 12:57:31 kamailio-VirtualBox /usr/local/sbin/kamailio[2110]: ERROR: <core> [route.c:1214]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:857

Some issues is there with the pattern.

Regards,

Nishar Hamsa

On Sat, Aug 17, 2013 at 11:29 AM, SamyGo govoiper@gmail.com wrote:

> Dear Nishar, > > From the mentioned URL the configurations need the following changes. > > On the top start on configs you need to use: WITH_PSTN and might not > need WITH_ASTERISK > > #!KAMAILIO > #!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_PSTN > > The insert your asterisk IP in the following line: > > > pstn.gw_ip = "<AsteriskIPHere>" desc "PSTN GW Address" > > > Next your route[LOCATION] is called after the route[PSTN] from > default config file so you'll have to put proper regexp/conditions to avoid > your regular Extension to Extension calls to go out to PSTN-Asterisk. > > Now in route[PSTN] add another condition to return from PSTN route > when a local extension is dialed. > > if(!($rU=~"^(+|00)[1-9][0-9]{3,20}$") || $rU=~"^([1-5]01[0-9]$")) > > > return; > > > That will ensure that your regular extensions set don't route out to > PSTN asterisk and infact will enter 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 AM, Victor V. Kustov coyote@bks.tvwrote: > >> Hi, Nishar M.H! >> >> > >> >ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio >> >start failed >> >> see log for details. do not use dumb copy/paste for kamailio config. >> >> -- >> WBR, Victor >> I use FREE operation system: 3.10.4-calculate GNU/Linux >> >> _______________________________________________ >> 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 >> > > > _______________________________________________ > 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 > >

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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

--

Thanks & Regards,

---

*Nishar Hamsa

---

---

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

---

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