Re: [SR-Users] Kamailio Crashes when dispatcher failover is being applied

[Daniel-Constantin Mierla]

Hello,

can you send the backtrace with gdb from the core file? The logs say the
corefile was generated -- it should beeiter in / or in the workind
directory you set for kamailio.

Cheers,
Daniel


On 18/07/16 22:45, TEG AMJG wrote:
> Dear list
>
> Im quite new to Kamailio and i found something while trying to see the
> failover behaviour in Kamailio. I have a kamailio v4.3.5 load
> balancing two Asterisk boxes.
>
> This happens just when i stop one of the asterisk and make a call that
> is being load balanced to the same asterisk box that i stopped before,
> when trying to go to failure_route and dispatch it to the other box
> then Kamailio just crashes. I have debug=4 in the config file and it
> really doesnt say much about the error. This is the relative logging i
> could found:
> /
> Jul 18 16:30:13 kam2 /usr/sbin/kamailio[69548]: INFO: 

Re: [SR-Users] Managing concurrent ports

[Grant Bagdasarian]

Yeah, that make sense. I kinda figured it would :P
Alright, thanks guys! :D

-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex 
Balashov
Sent: donderdag 21 juli 2016 02:12
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Managing concurrent ports

On 07/20/2016 06:43 PM, Alex Balashov wrote:
> On 07/20/2016 06:39 PM, Grant Bagdasarian wrote:
>
>> Do you also store state in the database?
>
> No. In my experience, that attracts from the beauty of the experience.
>

*detracts

--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Managing concurrent ports

[Alex Balashov]

On 07/20/2016 06:43 PM, Alex Balashov wrote:

On 07/20/2016 06:39 PM, Grant Bagdasarian wrote:


Do you also store state in the database?


No. In my experience, that attracts from the beauty of the experience.



*detracts

--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Managing concurrent ports

[Alex Balashov]

On 07/20/2016 06:39 PM, Grant Bagdasarian wrote:


Do you also store state in the database?


No. In my experience, that attracts from the beauty of the experience.

--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Managing concurrent ports

[Grant Bagdasarian]

Oh, nice! Thanks!
Do you also store state in the database?

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Brooks Bridges
Sent: woensdag 20 juli 2016 23:03
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Managing concurrent ports

We utilize the dialog module for concurrency control with several tens of 
thousands of channels at several thousand cps with zero issues.  As long as you 
spec the hardware suitably and configure the module properly, you won't have 
any issues until you hit really crazy levels of traffic.  I think you'll run 
out of resources on the network interfaces long before you have problems with 
the module itself.

Brooks Bridges | Sr. Voice Services Engineer
O1 Communications
4359 Town Center Boulevard, Suite 217
El Dorado Hills, California 95762
office: 916.235.2097 | main: 888.444., Option 2
email: bbrid...@o1.com | web: 
www.o1.com

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Grant Bagdasarian
Sent: Tuesday, July 19, 2016 11:46 PM
To: sr-users@lists.sip-router.org
Subject: [SR-Users] Managing concurrent ports

Hello,

Are there any recommended modules for managing concurrent ports for different 
endpoints/customers/accounts/etc?
I believe the dialog and dialog_ng modules do exactly this, but how is the 
performance of these modules under high loads?
What is the release time of the ports?

Are there perhaps other modules?

Regards,

Grant
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Re: [SR-Users] Managing concurrent ports

[Brooks Bridges]

We utilize the dialog module for concurrency control with several tens of 
thousands of channels at several thousand cps with zero issues.  As long as you 
spec the hardware suitably and configure the module properly, you won't have 
any issues until you hit really crazy levels of traffic.  I think you'll run 
out of resources on the network interfaces long before you have problems with 
the module itself.

Brooks Bridges | Sr. Voice Services Engineer
O1 Communications
4359 Town Center Boulevard, Suite 217
El Dorado Hills, California 95762
office: 916.235.2097 | main: 888.444., Option 2
email: bbrid...@o1.com | web: 
www.o1.com

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Grant Bagdasarian
Sent: Tuesday, July 19, 2016 11:46 PM
To: sr-users@lists.sip-router.org
Subject: [SR-Users] Managing concurrent ports

Hello,

Are there any recommended modules for managing concurrent ports for different 
endpoints/customers/accounts/etc?
I believe the dialog and dialog_ng modules do exactly this, but how is the 
performance of these modules under high loads?
What is the release time of the ports?

Are there perhaps other modules?

Regards,

Grant
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Re: [SR-Users] Managing concurrent ports

[Alex Balashov]

The dialog module remains the best-practical way of doing this, and it 
performs fine under high loads.


It's either that, or build your own dialog state machine using htable or 
whatever. Except it won't be as good at dealing with all possible 
eventualities in the complex lifecycle of a dialog as the dialog module 
itself.


On 07/20/2016 02:46 AM, Grant Bagdasarian wrote:


Hello,

Are there any recommended modules for managing concurrent ports for
different endpoints/customers/accounts/etc?

I believe the dialog and dialog_ng modules do exactly this, but how is
the performance of these modules under high loads?

What is the release time of the ports?

Are there perhaps other modules?

Regards,

Grant



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--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] music on hold configuration

[Robert Boisvert]

Guarav,

It seems that your kamailio.cfg is not setup to use MySQL. It says,
"db_bind_mod(): Module db_mysql not found".  You might want to review
http://www.kamailio.org/docs/modules/devel/modules/db_mysql.html

Bob

On Wed, Jul 20, 2016 at 4:41 PM, Gaurav Bmotra 
wrote:

> Hi Robert:
>
> 1. all the MOH uncommented
> 2. mohq_process() set now
> 3. rtpproxy module is loaded
> 4. tables created
> 5. Audio dir is created and Audio file is there
>
> But when srtart the kamailio it gives the error
>
> 
> Jul 21 02:09:06 bmotra polkitd(authority=local)[2720]: Operator of
> unix-session:c2 successfully authenticated as unix-user:bmotra to gain
> TEMPORARY au
> Jul 21 02:09:06 bmotra systemd[1]: Starting Kamailio (OpenSER) - the Open
> Source SIP Server...
> -- Subject: Unit kamailio.service has begun start-up
> -- Defined-By: systemd
> -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel
> --
> -- Unit kamailio.service has begun starting up.
> Jul 21 02:09:07 bmotra kamailio[12523]: INFO:  [sctp_core.c:74]:
> sctp_core_check_support(): SCTP API not enabled - if you want to use it,
> load s
> Jul 21 02:09:07 bmotra kamailio[12523]: WARNING: 
> [socket_info.c:1371]: fix_hostname(): could not rev. resolve 192.168.1.103
> Jul 21 02:09:07 bmotra kamailio[12523]: WARNING: 
> [socket_info.c:1371]: fix_hostname(): could not rev. resolve 192.168.1.103
> Jul 21 02:09:07 bmotra kamailio[12523]: INFO:  [tcp_main.c:4656]:
> init_tcp(): using epoll_lt as the io watch method (auto detected)
> Jul 21 02:09:07 bmotra kamailio[12523]: loading modules under config path:
> /usr/lib/x86_64-linux-gnu/kamailio/modules/
> Jul 21 02:09:07 bmotra kamailio[12523]: Listening on
> Jul 21 02:09:07 bmotra kamailio[12523]:  udp: 127.0.0.1:5060
> Jul 21 02:09:07 bmotra kamailio[12523]:  udp:
> 192.168.1.103:5060
> Jul 21 02:09:07 bmotra kamailio[12523]:  tcp: 127.0.0.1:5060
> Jul 21 02:09:07 bmotra kamailio[12523]:  tcp:
> 192.168.1.103:5060
> Jul 21 02:09:07 bmotra kamailio[12523]: Aliases:
> Jul 21 02:09:07 bmotra kamailio[12523]:  tcp: localhost:5060
> Jul 21 02:09:07 bmotra kamailio[12523]:  udp: localhost:5060
> Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: INFO: rr
> [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob
> Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: INFO: rr [rr_mod.c:174]:
> mod_init(): outbound module not available
> Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: INFO: usrloc
> [hslot.c:51]: ul_init_locks(): locks array size 1024
> Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: ERROR: 
> [db.c:203]: db_bind_mod(): Module db_mysql not found. Missing loadmodule?
> Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: ERROR: mohqueue
> [mohq.c:235]: init_db(): Unable to bind DB API using
> mysql://kamailio:kamailiorw@loc
> Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: ERROR: 
> [sr_module.c:961]: init_mod(): Error while initializing module mohqueue
> (/usr/lib/x86_
> Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: INFO: 
> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized
> Jul 21 02:09:07 bmotra systemd[1]: kamailio.service: Control process
> exited, code=exited status=255
> Jul 21 02:09:07 bmotra systemd[1]: Failed to start Kamailio (OpenSER) -
> the Open Source SIP Server.
> -- Subject: Unit kamailio.service has failed
> -- Defined-By: systemd
> -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel
> --
> -- Unit kamailio.service has failed.
>
> HARE I HAVE ATACHED MY KAMAILIO CONFIG FILE
>
>
>
>
>
> On Wed, Jul 20, 2016 at 11:19 PM, Robert Boisvert 
> wrote:
>
>> Gaurav,
>>
>> Your request is a little vague but the following things may be helpful:
>>
>>1. Your configuration has commented out the module statements which
>>disables it.
>>   - #loadmodule "mohqueue.so"
>>   - #modparam ("mohqueue", "db_url",
>>   "mysql://kamailio:kamailiorw@localhost/kamailio")
>>   - #modparam ("mohqueue", "mohdir", "/var/kamailio/MOH")
>>2. There is no mohq_process() or mohq_send() functions in the main
>>loop to intercept INVITES to the queue.
>>http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#proc.func
>>3. It is not clear if the configuration has loaded the rtpproxy
>>module.
>>
>> http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#mod.depends
>>4. Have you created and populated the MOHQUEUES and MOHQCALLS
>>tables in the database?
>>http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#database
>>5. Did you create audio files and put them in the mohdir?
>>
>> http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#audiofiles
>>
>> Bob
>>
>> On Wed, Jul 20, 2016 at 1:25 PM, Gaurav Bmotra <
>> saigauravmehr...@gmail.com> wrote:
>>
>>> hi i m a new user of kamailio
>>> i 

Re: [SR-Users] Kamailio + rtpproxy + SRTP

[Vasiliy Ganchev]

Konstantin M. wrote
> I have some problem with SSL+rtpproxy, kamailio successfully decodes SIP,
> rtpproxy decodes SRTP frames, all is going to freeswitch unencrypted. FS
> transfers the call back, the call is going unencrypted though kamailio,
> kamailio encodes to SSL, but rtpproxy DOESN'T encodes RTP frames to SRTP.
> Hence, endpoint SIP equipment rejects the call with 488 Not Acceptable
> Here
> 
> Scenario
> 
> 1. Endpoint device -> SRTP -> kamailio -> RTP -> freeswitch
> 2. Freeswitch calls back
> 3. freeswitch -> RTP -> kamailio -> PROBLEM (SIP[TLS] + RTP (but not
> SRTP))
> -> Endpoint device
> 4. Endpoint device -> 488 Not Acceptable Here -> Kamailio
> 
> 
> Could somebody point me what's wrong here?
> 
> Thanks!
> 
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> sr-users@.sip-router

> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Hi!
488 is the SIP reply, and your problem is not rtp->srtp encryption (this is
this consequence), but that you called rtpproxy_manage with wrong
parameters.

e.g. if endpoint is browser it can answer 488 if SDP in  INVITE has RTP/AVPF
(it work only with AVP/SRTP)

can suggest to:
- check logs on endpoint
- check parameters of rtpproxy_manage
- use rtpengine instead of rtpproxy

cheers!



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[SR-Users] reading siptrace trace_on current value in script

[ananda agrawal]

Hi,

Is it possible to check wether the trace_on for siptrace module is on or
off from within the config script, e.g inside routing logic,

thannks in advance
-Anand
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[SR-Users] Trouble with Asterisk behind Kamailio

[Tickling Contest]

Hello,

I am following the instructions in
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
to configure my Kamailio and Asterisk (using PJSIP realtime with MySQL). I
am able to get Kamailio register the clients (tested with Zoiper
softphone), but having trouble getting any further to load balance a couple
of Asterisk servers. Here is a range of issues I have:

(a) The Kamailio server keeps sending UDP SIP messages to the Asterisk
server and it is not clear where to control what protocol to use to send
SIP messages Asterisk server(s).
(b) I followed the instructions for the dispatcher module in the wiki, but
it is not clear why I should use the #!define WITH_ASTERISK directive to
enumerate the bind IPs and ports (how does this reconcile with the list of
Asterisk servers in the dispatcher.list file?):

!ifdef WITH_ASTERISK
asterisk.bindip = "192.168.1.102" desc "Asterisk IP Address"
asterisk.bindport = "5060" desc "Asterisk Port"
kamailio.bindip = "192.168.1.101" desc "Kamailio IP Address"
kamailio.bindport = "5060" desc "Kamailio Port"

Also, where do I specify the protocol to use when sending to the Asterisk
servers? Is it auto detected? If so, why is Kamailio not detecting that the
Asterisk servers are only listening on TCP?

Any help is appreciated. My entire /etc/kamailio/kamailio.cfg file is in
here:
https://gist.github.com/ticklingcontest/c7437dbc2d8e838834ee4d852753f254. I
am using Kamailio 4.0.4 with Asterisk 13.7.2 using TCP for PJSIP.

Thanks!
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Re: [SR-Users] music on hold configuration

[Gaurav Bmotra]

Hi Robert:

1. all the MOH uncommented
2. mohq_process() set now
3. rtpproxy module is loaded
4. tables created
5. Audio dir is created and Audio file is there

But when srtart the kamailio it gives the error


Jul 21 02:09:06 bmotra polkitd(authority=local)[2720]: Operator of
unix-session:c2 successfully authenticated as unix-user:bmotra to gain
TEMPORARY au
Jul 21 02:09:06 bmotra systemd[1]: Starting Kamailio (OpenSER) - the Open
Source SIP Server...
-- Subject: Unit kamailio.service has begun start-up
-- Defined-By: systemd
-- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel
-- 
-- Unit kamailio.service has begun starting up.
Jul 21 02:09:07 bmotra kamailio[12523]: INFO:  [sctp_core.c:74]:
sctp_core_check_support(): SCTP API not enabled - if you want to use it,
load s
Jul 21 02:09:07 bmotra kamailio[12523]: WARNING: 
[socket_info.c:1371]: fix_hostname(): could not rev. resolve 192.168.1.103
Jul 21 02:09:07 bmotra kamailio[12523]: WARNING: 
[socket_info.c:1371]: fix_hostname(): could not rev. resolve 192.168.1.103
Jul 21 02:09:07 bmotra kamailio[12523]: INFO:  [tcp_main.c:4656]:
init_tcp(): using epoll_lt as the io watch method (auto detected)
Jul 21 02:09:07 bmotra kamailio[12523]: loading modules under config path:
/usr/lib/x86_64-linux-gnu/kamailio/modules/
Jul 21 02:09:07 bmotra kamailio[12523]: Listening on
Jul 21 02:09:07 bmotra kamailio[12523]:  udp: 127.0.0.1:5060
Jul 21 02:09:07 bmotra kamailio[12523]:  udp: 192.168.1.103:5060
Jul 21 02:09:07 bmotra kamailio[12523]:  tcp: 127.0.0.1:5060
Jul 21 02:09:07 bmotra kamailio[12523]:  tcp: 192.168.1.103:5060
Jul 21 02:09:07 bmotra kamailio[12523]: Aliases:
Jul 21 02:09:07 bmotra kamailio[12523]:  tcp: localhost:5060
Jul 21 02:09:07 bmotra kamailio[12523]:  udp: localhost:5060
Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: INFO: rr
[../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob
Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: INFO: rr [rr_mod.c:174]:
mod_init(): outbound module not available
Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: INFO: usrloc
[hslot.c:51]: ul_init_locks(): locks array size 1024
Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: ERROR:  [db.c:203]:
db_bind_mod(): Module db_mysql not found. Missing loadmodule?
Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: ERROR: mohqueue
[mohq.c:235]: init_db(): Unable to bind DB API using
mysql://kamailio:kamailiorw@loc
Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: ERROR: 
[sr_module.c:961]: init_mod(): Error while initializing module mohqueue
(/usr/lib/x86_
Jul 21 02:09:07 bmotra /usr/sbin/kamailio[12526]: INFO: 
[sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized
Jul 21 02:09:07 bmotra systemd[1]: kamailio.service: Control process
exited, code=exited status=255
Jul 21 02:09:07 bmotra systemd[1]: Failed to start Kamailio (OpenSER) - the
Open Source SIP Server.
-- Subject: Unit kamailio.service has failed
-- Defined-By: systemd
-- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel
-- 
-- Unit kamailio.service has failed.

HARE I HAVE ATACHED MY KAMAILIO CONFIG FILE





On Wed, Jul 20, 2016 at 11:19 PM, Robert Boisvert 
wrote:

> Gaurav,
>
> Your request is a little vague but the following things may be helpful:
>
>1. Your configuration has commented out the module statements which
>disables it.
>   - #loadmodule "mohqueue.so"
>   - #modparam ("mohqueue", "db_url",
>   "mysql://kamailio:kamailiorw@localhost/kamailio")
>   - #modparam ("mohqueue", "mohdir", "/var/kamailio/MOH")
>2. There is no mohq_process() or mohq_send() functions in the main
>loop to intercept INVITES to the queue.
>http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#proc.func
>3. It is not clear if the configuration has loaded the rtpproxy
>module.
>
> http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#mod.depends
>4. Have you created and populated the MOHQUEUES and MOHQCALLS
>tables in the database?
>http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#database
>5. Did you create audio files and put them in the mohdir?
>http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#audiofiles
>
> Bob
>
> On Wed, Jul 20, 2016 at 1:25 PM, Gaurav Bmotra  > wrote:
>
>> hi i m a new user of kamailio
>> i have sertup kamailio SIP to SIP call going well
>> but now i m tring to impliment Music On Hold
>> i m this
>>
>> http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#table.parms
>> but unable to impliment plz help me
>>
>> Hare i attache my kamailio config file
>>
>>
>> --
>>
>>
>>
>>
>>
>>
>>
>>
>> *Regards:*
>> Gaurav Bmotra
>> Smith Infotech
>> E-46, phase 8 , Indus Area S.A.S Nagar Mohali 160071
>>
>> ___
>> SIP 

Re: [SR-Users] WEBRTC Confusion (2)

[Zaka]

Colin,

There is gratitude for you for your time and exceptional consideration.
I am really indebted, mate! Here in Albanian culture "I owe you a coffee"
(no a dinner!) :)

So as per your advice, I shall stick to the run of the mill clients at this
stage, rather than reinvent a slightly better wheel.

With kindest regards,

Zaka




On 19 July 2016 at 21:51, Colin Morelli  wrote:

> Zakia,
>
> Alright, starting with the basics: IP based voice/video communications
> generally involve two distinct responsibilities: signaling and media.
>
> The job of the signaling layer is to exchange information about calls
> between two parties. For example, if I call you, I use my signaling
> protocol to talk to my signaling server to send you an invitation to
> connect. The server (somehow - implementation dependent) relays that
> information to you. Once you've received the information, you can
> answer/ignore/reject the invitation. The signaling layer is responsible for
> notifying me about your response. The ultimate goal of the signaling layer
> is to exchange media descriptions, that tell each party where/how they can
> communicate with each other. SIP is an example of a signaling protocol, and
> it uses SDPs to exchange information about media.
>
> The job of the media layer is to actually exchange the media between two
> endpoints. With the job of the signaling layer complete, I should have some
> information about how I can talk to you. With that information I received
> over the signaling layer, I should be able to open my local audio/video
> devices, and stream data to you regardless of where you are. The media
> layer is responsible for handling that data and ensuring it gets to its
> destination. WebRTC is a spec for a media implementation which uses
> STUN/TURN/ICE for NAT traversal and reliable media exchange. While it is
> commonly built on top of SIP, there is no reason it needs to be. SIP is not
> directly involved in WebRTC at all. Rather, WebRTC-capable SIP endpoints
> simply generate an SDP containing all the information WebRTC needs to know
> in order to take over the rest (what codecs should be used, ICE candidates
> where the remote party can be reached, etc).
>
> Now, Kamailio is a SIP server/proxy. This means it speaks/understands the
> SIP protocol. This also means that its primary concern is with signaling.
> If you want to use Kamailio as the signaling layer for your application -
> then you'll need to talk to it via SIP. sipml5 is a javascript based SIP
> client that can run in browser, but there are others as well (SIP.js,
> JsSIP). You'll probably want to use one of these rather than build your
> own. Now, because SIP is just the signaling layer - there's no reason you
> *need* to use SIP in order to work with WebRTC. However, the Kamailio
> group, as well as the SIP client authors, have spent a lot of time building
> and refining their products to make them production-grade and robust.
>
> If you want to use Kamailio, you need a SIP client. Even with WebRTC
> (remember, WebRTC is only about media). If you were feeling very
> adventurous (although I would never recommend it), you could build your own
> signaling layer and use your own custom client. But you'd also have to
> build a server component that speaks your new protocol.
>
> Bottom line: you probably just want to stick with Kamailio + one of the
> great javascript SIP clients I mentioned above. They're quite
> stable/reliable.
>
> Hopefully that helps.
>
> Best,
> Colin
>
> On Tue, Jul 19, 2016 at 3:11 PM Zaka  wrote:
>
>> Hello Colin:
>>
>> Thanks a lot! It solved the issue. Though I had got a hint by succeeding
>> to specify the address using *kamailio -l desired address/port *option,
>> yet never noticed the lack of *MY-WS_ADDRESS*
>>
>> Last, but not the least, in a browser supporting WEBRTC (FF, CHROME) do
>> we still need SIP ML5as on:  https://www.doubango.org/sipml5/
>>
>> If not, I wonder how will the WS or WSS URL post the login credentials to
>> the KAMAILIO WEBRTC server?
>>
>> BR,
>>
>> Zaka
>>
>>
>>
>> On 16 July 2016 at 18:11, Colin Morelli  wrote:
>>
>>> Zaka,
>>>
>>> I could be wrong here but I don't think you ever actually have a
>>> "listen" line for MY_WS_ADDR.
>>>
>>> I believe you have a typo, as you have listen=MY_IP_ADDR twice, once
>>> within the guard for WITH_WEBSOCKETS. Replace the one inside the if with
>>> MY_WS_ADDR and I think your problem should be resolved.
>>>
>>> Best,
>>> Colin
>>>
>>> On Sat, Jul 16, 2016 at 3:36 AM Zaka  wrote:
>>>
 Dear List:

 I am still stuck. Still unable to get Kamailio to listen on WEBRTC port*
 (in this case 8000)*

 Please advise if it has something to do with disabling TLS/MSRP?


 Please see the log & related config file in line here:

 *Log:*

  0(5724) INFO:  [ppcfg.c:82]: pp_subst_add(): ### added subst
 

Re: [SR-Users] music on hold configuration

[Robert Boisvert]

Gaurav,

Your request is a little vague but the following things may be helpful:

   1. Your configuration has commented out the module statements which
   disables it.
  - #loadmodule "mohqueue.so"
  - #modparam ("mohqueue", "db_url",
  "mysql://kamailio:kamailiorw@localhost/kamailio")
  - #modparam ("mohqueue", "mohdir", "/var/kamailio/MOH")
   2. There is no mohq_process() or mohq_send() functions in the main loop
   to intercept INVITES to the queue.
   http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#proc.func
   3. It is not clear if the configuration has loaded the rtpproxy module.
   http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#mod.depends
   4. Have you created and populated the MOHQUEUES and MOHQCALLS tables in
   the database?
   http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#database
   5. Did you create audio files and put them in the mohdir?
   http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#audiofiles

Bob

On Wed, Jul 20, 2016 at 1:25 PM, Gaurav Bmotra 
wrote:

> hi i m a new user of kamailio
> i have sertup kamailio SIP to SIP call going well
> but now i m tring to impliment Music On Hold
> i m this
>
> http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#table.parms
> but unable to impliment plz help me
>
> Hare i attache my kamailio config file
>
>
> --
>
>
>
>
>
>
>
>
> *Regards:*
> Gaurav Bmotra
> Smith Infotech
> E-46, phase 8 , Indus Area S.A.S Nagar Mohali 160071
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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[SR-Users] music on hold configuration

[Gaurav Bmotra]

hi i m a new user of kamailio
i have sertup kamailio SIP to SIP call going well
but now i m tring to impliment Music On Hold
i m this

http://www.kamailio.org/docs/modules/devel/modules/mohqueue.html#table.parms
but unable to impliment plz help me

Hare i attache my kamailio config file


-- 








*Regards:*
Gaurav Bmotra
Smith Infotech
E-46, phase 8 , Indus Area S.A.S Nagar Mohali 160071


kamailio.cfg
Description: Binary data
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Re: [SR-Users] Tsilo Tiimers

[Federico Cabiddu]

Hi,
to set the timeout for a specific transaction you can use tm module
function t_set_fr (
http://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_set_fr).
For example you could call it before suspending the transaction setting a
short timeout and then call it again in the reply route when receiving the
first provisional response from the client.

Regards,

Federico
On 20 Jul 2016 2:50 pm, "Nahum Nir"  wrote:

> Hello,
>
> How can I chage the timer waiting for the client to send the register.
> i.e. how much time to wait for the push?
>
> Thanks,
> Nir
>
>
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>
>
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[SR-Users] Tsilo Tiimers

[Nahum Nir]

Hello,

How can I chage the timer waiting for the client to send the register. i.e.
how much time to wait for the push?

Thanks,
Nir
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[SR-Users] Managing concurrent ports

[Grant Bagdasarian]

Hello,

Are there any recommended modules for managing concurrent ports for different 
endpoints/customers/accounts/etc?
I believe the dialog and dialog_ng modules do exactly this, but how is the 
performance of these modules under high loads?
What is the release time of the ports?

Are there perhaps other modules?

Regards,

Grant
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