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.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

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Nishar Hamsa






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Thanks & Regards,

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Nishar Hamsa






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Thanks & Regards,

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