Re: [Kamailio-Users] Register Request Forward
No two way communication is a problem of NAT in general. If session has been
established then it's not codec problem.
From: toqeer ali [mailto:toqee...@gmail.com]
Sent: Thursday, November 12, 2009 9:25 AM
To: Denis Putyato
Cc: users@lists.kamailio.org
Subject: Re: [Kamailio-Users] Register Request Forward
i have a problem with codec. when i call to canada from kamailio to PSTN, i
could not listen his/her voice as they listen( no two way communication)
but when call to other countries it works... Basically call to canada
support g729 can i define any codec information in kamailio to G729.
Basically the problem is only with local number when you dial with extension
1 and when you dial with 08 it works... 08 is international gateway of
canada...
X-lite i am using as a softphone.
please help
2009/11/11 Denis Putyato denis7...@mail.ru
can i use regular expression in $fu=XXX
You can use regexp. $fu=~expression
From: toqeer ali [mailto:toqee...@gmail.com]
Sent: Wednesday, November 11, 2009 3:08 PM
To: Denis Putyato
Cc: users@lists.kamailio.org
Subject: Re: [Kamailio-Users] Register Request Forward
Hmmm
it works now i just remove Record Route function and it works,,, now i can
dial from 602 to 601... thanks :).
i will start work on how i will register multiple extension via kamailio to
asterisk ... can i use regular expression in $fu=XXX instead of actual
extension extension...
2009/11/11 Denis Putyato denis7...@mail.ru
Registration is a process for telling SoftSwitch (in your variant -
Asterisk) where (on which username, IP address and port) the SoftSwitch must
send INVITE when received incoming call for 601. Try to understand where
your asterisk try to send INVITE when you try to call from 602 to 601.
ngrep utility will help you (http://ngrep.sourceforge.net/download.html)
From: toqeer ali [mailto:toqee...@gmail.com]
Sent: Wednesday, November 11, 2009 2:18 PM
To: Denis Putyato
Cc: users@lists.kamailio.org
Subject: Re: [Kamailio-Users] Register Request Forward
Here is the update:
what i have done so for is that I have Register extension 601 to asterisk
via kamailio and it is registered!.
When i call to 602 (configured on 192.168.0.1 -- asterisk box) it calls and
make connection and voice call also works (RTP Traffic is ok).
However, when i call to 601 from 602 (which is directly connected to
asterisk without kamailio ), 601 can't receive call via kamailio. This is
the main problem. Please guide me that how i will define this in kamailio
(ip = 192.168.0.2) that xlite with extension 601 can receive call through
602 which is another extension on Asterisk.
In a nutshell: 601 (registered through Kamailio to Asterisk) ---can call
--- 602 (registered via Asterisk)
However:602cannot call -- 601
please help.
2009/11/11 Denis Putyato denis7...@mail.ru
Try to use dlg_bridge(); function in dialog module
From: toqeer ali [mailto:toqee...@gmail.com]
Sent: Wednesday, November 11, 2009 12:13 PM
To: Denis Putyato
Cc: users@lists.kamailio.org
Subject: Re: [Kamailio-Users] Register Request Forward
that client send INVITE to asterisk via kamailio but no replay is in on
this INVITE
basically Asterisk receive the request and send back to kamailio but
kamailio can't send back to xlite..
2009/11/11 Denis Putyato denis7...@mail.ru
but call does not route back to me
You mean, that incoming call from asterisk to client via kamailio doesn't
work or that client send INVITE to asterisk via kamailio but no replay is in
on this INVITE?
From: toqeer ali [mailto:toqee...@gmail.com]
Sent: Wednesday, November 11, 2009 10:16 AM
To: Denis Putyato
Cc: us...@l
ists.kamailio.org
Subject: Re: [Kamailio-Users] Register Request Forward
Thanks Again for the prompt reply... Basicaly i want to route Register
Requests for multiple asterisk via kamailio ... by using pattern matching
of extensions...
is i am on the right path ?? or i can achieve this from dispatcher or load
balancer module Basically i want to register specific extensions to
specific asterisk boxes
from your previous reply i can send call to asterisk and register to as well
but call does not route back to me... any clue ...
2009/11/11 Denis Putyato denis7...@mail.r
The secret is that the other sip server was asterisk and i want route call
to asterisk via kamailio :).
I don't understand why do you want to do any session with asterisk via
kamailio: not directly to asterisk: but any way the decision is same -
send() or rewritehost()
route {
if ($fU==:is_method(INVITE))
record_route();
rewritehost(192.168.0.1);
t_relay();
}
Do this before authentication checking is made in kamailio.
From: toqeer ali [mailto:toqee...@gmail.com]
Sent: Wednesday, November 11, 2009 6:49 AM
To: Denis Putyato
Cc: users@lists.kamailio.org
Subject: Re: [Kamailio-Users] Register Request Forward
Thanks it
Re: [Kamailio-Users] record-route
sip show peer after success registration (with record route() using )?
From: users-boun...@lists.kamailio.org
[mailto:users-boun...@lists.kamailio.org] On Behalf Of toqeer ali
Sent: Thursday, November 12, 2009 10:33 AM
To: users@lists.kamailio.org
Subject: [Kamailio-Users] record-route
Let me describe my scenario:
SipUser1 ---register request kamailio (SIP Proxy) ---
forwarded register request--- Asterisk
1. In the above scenario, when I register to Asterisk through Kamailio,
it works fine when I give the record-route() function in the cfg file.
Further, using this, I can also call to another extension 602 which is also
registered to Asterisk but 602 cannot call to 601 which is registered via
kamailio.
2. However, If I remove the record-route() function without
unregistering 601 and restarting the server, 602 can call to 601 .
route {
if ($fU==601is_method(INVITE))
record_route();
rewritehost(10.0.10.111); //asterisk host
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
should i add anything in script further...
--
Toqeer Ali Syed
Red Hat Certified Engineer
mob: +92 321 9059916
___
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Re: [Kamailio-Users] Register Request Forward
Thanks Denis,
It is indeed the problem of NAT. We have tried with live ip and its work
fine. However, behind the NAT it is not working. I have installed RTPproxy ,
but still it is not working. Here are the configuration that we use in our
cfg file.
modparam(nathelper,received_avp, $avp(i:42))
modparam(nathelper, rtpproxy_sock, udp:127.0.0.1:8899)
modparam(nathelper, natping_interval, 30)
modparam(nathelper, ping_nated_only, 0)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from,
sip:pin...@127.0.0.1sip%3apin...@127.0.0.1
)
route {
## NAT Detection
#
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
.
and rtpproxy is also configured on the same server.
2009/11/12 Denis Putyato denis7...@mail.ru
No two way communication is a problem of NAT in general. If session has
been established then it’s not codec problem.
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Thursday, November 12, 2009 9:25 AM
*To:* Denis Putyato
*Cc:* users@lists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
i have a problem with codec. when i call to canada from kamailio to PSTN, i
could not listen his/her voice as they listen( no two way communication)
but when call to other countries it works... Basically call to canada
support g729 can i define any codec information in kamailio to G729.
Basically the problem is only with local number when you dial with
extension 1 and when you dial with 08 it works... 08 is international
gateway of canada...
X-lite i am using as a softphone.
please help
2009/11/11 Denis Putyato denis7...@mail.ru
“can i use regular expression in $fu=XXX”
You can use regexp. $fu=~”expression”
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Wednesday, November 11, 2009 3:08 PM
*To:* Denis Putyato
*Cc:* users@lists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
Hmmm
it works now i just remove Record Route function and it works,,, now i can
dial from 602 to 601... thanks :).
i will start work on how i will register multiple extension via kamailio to
asterisk ... can i use regular expression in $fu=XXX instead of actual
extension extension...
2009/11/11 Denis Putyato denis7...@mail.ru
Registration is a process for telling SoftSwitch (in your variant -
Asterisk) where (on which username, IP address and port) the SoftSwitch must
send INVITE when received incoming call for 601. Try to understand where
your asterisk try to send INVITE when you try to call from 602 to 601.
ngrep utility will help you (http://ngrep.sourceforge.net/download.html)
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Wednesday, November 11, 2009 2:18 PM
*To:* Denis Putyato
*Cc:* users@lists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
Here is the update:
what i have done so for is that I have Register extension 601 to asterisk
via kamailio and it is registered!.
When i call to 602 (configured on 192.168.0.1 -- asterisk box) it calls and
make connection and voice call also works (RTP Traffic is ok).
However, when i call to 601 from 602 (which is directly connected to
asterisk without kamailio ), 601 can't receive call via kamailio. This is
the main problem. Please guide me that how i will define this in kamailio
(ip = 192.168.0.2) that xlite with extension 601 can receive call through
602 which is another extension on Asterisk.
In a nutshell: 601 (registered through Kamailio to Asterisk) ---can call
--- 602 (registered via Asterisk)
However:602cannot call -- 601
please help.
2009/11/11 Denis Putyato denis7...@mail.ru
Try to use dlg_bridge(); function in dialog module
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Wednesday, November 11, 2009 12:13 PM
*To:* Denis Putyato
*Cc:* users@lists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
that client send INVITE to asterisk via kamailio but no replay is in on
this INVITE
basically Asterisk receive the request and send back to kamailio but
kamailio can't send back to xlite..
2009/11/11 Denis Putyato denis7...@mail.ru
«but call does not route back to me»
You mean, that incoming call from asterisk to client via kamailio doesn’t
work or that client send INVITE to asterisk via kamailio but no replay is in
on this INVITE?
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Wednesday, November 11, 2009 10:16 AM
*To:* Denis Putyato
*Cc:* us...@l
ists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
Thanks Again for the prompt reply... Basicaly i want to route Register
Requests for multiple asterisk via kamailio ... by using pattern matching
of extensions...
is i am on the
Re: [Kamailio-Users] Remote management and security
2009/11/11 Brian br...@proximosystems.com: Hi, Does anyone know how I can do remote management of Kamailio in a secure way? Looking at the documentation for MI_DATAGRAM and MI_XMLRPC, there does not seem to be a way to authenticate the request, or a way to restrict requests to specific IP ranges. I want to be able to send management commands remotely, but I don’t to make this wide open. Iptables is your only workaround at this moment (or wait to SR 3.0). -- Iñaki Baz Castillo i...@aliax.net ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] RADIUS for billing
Dear All, I am trying to configure openSER for billing.After a lot og googling, i came to know that RADIUS is needed for anabling billing module. I would like to know if RADIUS is a MUST for billing or can i manage using mysql alone. I have configured accounting module in openser.can i use mysql and a CDR tool for billing?.. -- Regards, Rajesh Sreedharan Software Trainee-Wireless Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :rajesh.sreedha...@siptech.com Website : www.siptech.com ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] SKYPE or voipstunt
whether skype or voipstunt are using SIP protocols. I have experienced that in many countries, where voice ports are blocked by the Telecom providers but still voipstunt and skype works for calling to PSTN. Please give me some links, for further reading on this. ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] SKYPE or voipstunt
Don't know about Voipstunt. Skype has an entirely proprietary protocol stack - no SIP, or anything else open. Masoom Alam wrote: whether skype or voipstunt are using SIP protocols. I have experienced that in many countries, where voice ports are blocked by the Telecom providers but still voipstunt and skype works for calling to PSTN. Please give me some links, for further reading on this. ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] RADIUS for billing
Dear Alex, Thanks for the valuable post.Otherwise, I would have been toddling with RADIUS.Kindly let me know how do i proceed with openSER and using Mysql to do the billing.. Alex Balashov wrote: RADIUS is not a must; you can use plain database records the 'acc' module puts out for billing. In fact, unless you have a specific need for RADIUS due to billing software that operates on its record format, or have some other reason for it, it is unnecessary complexity you should avoid. There are some technical and resilience advantages to using RADIUS, but in my opinion they hardly justify the complexity (especially for a new user) unless you have an identifiable need to use it. Rajesh wrote: Dear All, I am trying to configure openSER for billing.After a lot og googling, i came to know that RADIUS is needed for anabling billing module. I would like to know if RADIUS is a MUST for billing or can i manage using mysql alone. I have configured accounting module in openser.can i use mysql and a CDR tool for billing?.. -- Regards, Rajesh Sreedharan Software Trainee-Wireless Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :rajesh.sreedha...@siptech.com mailto:rajesh.sreedha...@siptech.com? Website : www.siptech.com http://www.siptech.com/ http://www.siptech.com ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Regards, Rajesh Sreedharan Software Trainee-Wireless Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :rajesh.sreedha...@siptech.com Website : www.siptech.com ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] RADIUS for billing
Rajesh wrote: Thanks for the valuable post.Otherwise, I would have been toddling with RADIUS.Kindly let me know how do i proceed with openSER and using Mysql to do the billing.. Import the 'acc' module table schema from the tarball into MySQL, configure the 'acc' module to log to the database, set the accounting flags in your code, and you're set. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Register Request Forward
My rtpproxy works now basically i am using kamailio 1.2 with LCR, but when i
dial one number via LCR it dial this number and it works but when i dial
second number it says *filtered destination*, i have to restart the
service and then it dial second number...
i think LCR not de-alocating the resources any idea ?
2009/11/12 Denis Putyato denis7...@mail.ru
1) I would done not ” modparam(nathelper, rtpproxy_sock, udp:
127.0.0.1:8899)” but “modparam(nathelper, rtpproxy_sock,
unix:/var/run/rtpproxy.sock)”
2) http://www.kamailio.org/docs/modules/1.5.x/nathelper.html#id2468157
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Thursday, November 12, 2009 2:10 PM
*To:* Denis Putyato
*Cc:* users@lists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
Thanks Denis,
It is indeed the problem of NAT. We have tried with live ip and its work
fine. However, behind the NAT it is not working. I have installed RTPproxy ,
but still it is not working. Here are the configuration that we use in our
cfg file.
modparam(nathelper,received_avp, $avp(i:42))
modparam(nathelper, rtpproxy_sock, udp:127.0.0.1:8899)
modparam(nathelper, natping_interval, 30)
modparam(nathelper, ping_nated_only, 0)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from,
sip:pin...@127.0.0.1sip%3apin...@127.0.0.1
)
route {
## NAT Detection
#
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
.
and rtpproxy is also configured on the same server.
2009/11/12 Denis Putyato denis7...@mail.ru
No two way communication is a problem of NAT in general. If session has
been established then it’s not codec problem.
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Thursday, November 12, 2009 9:25 AM
*To:* Denis Putyato
*Cc:* users@lists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
i have a problem with codec. when i call to canada from kamailio to PSTN, i
could not listen his/her voice as they listen( no two way communication)
but when call to other countries it works... Basically call to canada
support g729 can i define any codec information in kamailio to G729.
Basically the problem is only with local number when you dial with
extension 1 and when you dial with 08 it works... 08 is international
gateway of canada...
X-lite i am using as a softphone.
please help
2009/11/11 Denis Putyato denis7...@mail.ru
“can i use regular expression in $fu=XXX”
You can use regexp. $fu=~”expression”
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Wednesday, November 11, 2009 3:08 PM
*To:* Denis Putyato
*Cc:* us...@lists.kamailio..org users@lists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
Hmmm
it works now i just remove Record Route function and it works,,, now i can
dial from 602 to 601... thanks :).
i will start work on how i will register multiple extension via kamailio to
asterisk ... can i use regular expression in $fu=XXX instead of actual
extension extension...
2009/11/11 Denis Putyato denis7...@mail.ru
Registration is a process for telling SoftSwitch (in your variant -
Asterisk) where (on which username, IP address and port) the SoftSwitch must
send INVITE when received incoming call for 601. Try to understand where
your asterisk try to send INVITE when you try to call from 602 to 601.
ngrep utility will help you (http://ngrep.sourceforge.net/download.html)
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Wednesday, November 11, 2009 2:18 PM
*To:* Denis Putyato
*Cc:* us...@lists.kamailio..org users@lists.kamailio.org
*Subject:* Re: [Kamailio-Users] Register Request Forward
Here is the update:
what i have done so for is that I have Register extension 601 to asterisk
via kamailio and it is registered!.
When i call to 602 (configured on 192.168.0.1 -- asterisk box) it calls and
make connection and voice call also works (RTP Traffic is ok).
However, when i call to 601 from 602 (which is directly connected to
asterisk without kamailio ), 601 can't receive call via kamailio. This is
the main problem. Please guide me that how i will define this in kamailio
(ip = 192.168.0.2) that xlite with extension 601 can receive call through
602 which is another extension on Asterisk.
In a nutshell: 601 (registered through Kamailio to Asterisk) ---can call
--- 602 (registered via Asterisk)
However:602cannot call -- 601
please help.
2009/11/11 Denis Putyato denis7...@mail.ru
Try to use dlg_bridge(); function in dialog module
*From:* toqeer ali [mailto:toqee...@gmail.com]
*Sent:* Wednesday, November 11, 2009 12:13 PM
*To:* Denis Putyato
*Cc:* us...@lists.kamailio..org users@lists.kamailio.org
