[SR-Users] sr-users Digest, Vol 92, Issue 43

Tue Jan 22 04:34:32 CET 2013

Thanks a lot for quick response.

On Sat, January 19, 2013 3:59 am, sr-users-request at lists.sip-router.org
wrote:
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> Today's Topics:
>
>
> 1. Mid call announcement(kamailio server)
> (madhumanjusha at integramicro.com)
> 2. Re: Mid call announcement(kamailio server)
> (rabs at dimension-virtual.com)
> 3. CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus
> dlg - Kamailio 3.3.x (Rinor Hoxha) 4. Huge undertaking and in need of
> advices from the	community (Rumen Mihailov)
> 5. Re: Huge undertaking and in need of advices from the
> community (Stoyan Mihaylov) 6. Re: Invite - negotiate to asterisk as peer
> not kamailio (Scott, Matt)
>
>
>
> ----------------------------------------------------------------------
>
>
> Message: 1
> Date: Fri, 18 Jan 2013 06:52:15 -0500 (EST)
> From: madhumanjusha at integramicro.com
> Subject: [SR-Users] Mid call announcement(kamailio server)
> To: sr-users at lists.sip-router.org
> Message-ID:
> <39960.61.8.152.138.1358509935.squirrel at mail.integramicro.com>
> Content-Type: text/plain;charset=iso-8859-1
>
>
> Hello,
> As of now am working on Kamailio server.
> My task is to announce in between call,my sip clients are EKIGA.
> I should able to make my kamailio server act as controller and media
> server,for that purpose my first step is : invite hold to called party.
> second step is:: invite(sdp for .wav file) to caller and then am sending
> rtp streaming.In wireshark am able to see rtp packets,but caller is not
> playing the .wav file which am streaming through ortp library from
> kamailio.
>
> Can anyone help me out with this problem?On top of it,is this possible in
>  reality?
>
>
> Thanks & Regards,
> Manjusha A.
> Integra Micro Software Services (P) Ltd.
>
>
>
>
>
>
>
> ------------------------------
>
>
> Message: 2
> Date: Fri, 18 Jan 2013 15:14:49 +0100 (CET)
> From: rabs at dimension-virtual.com
> Subject: Re: [SR-Users] Mid call announcement(kamailio server)
> To: "SIP Router - Kamailio \(OpenSER\) and SIP Express Router \(SER\)
> - Users	Mailing List" <sr-users at lists.sip-router.org>
> Message-ID:
> <38001.212.40.242.42.1358518489.squirrel at www.rodriguezfeo.es>
> Content-Type: text/plain;charset=iso-8859-1
>
>
>> Hello,
>> As of now am working on Kamailio server.
>> My task is to announce in between call,my sip clients are EKIGA.
>> I should able to make my kamailio server act as controller and media
>> server,for that purpose my first step is : invite hold to called party.
>> second step is:: invite(sdp for .wav file) to caller and then am
>> sending rtp streaming.In wireshark am able to see rtp packets,but caller
>> is not playing the .wav file which am streaming through ortp library
>> from kamailio.
>>
>> Can anyone help me out with this problem?On top of it,is this possible
>> in reality?
>
> No, it's not possible using only kamailio, and thats because kamailio
> it's a SIP proxy, it have nothing to do with media, for what you are
> triying to get you need to router your calls throught a B2BUA, like sems
> or asterisk.
>
> Best regards
>
>
>
>
>
> ------------------------------
>
>
> Message: 3
> Date: Fri, 18 Jan 2013 14:11:05 +0100
> From: Rinor Hoxha <rinorhoxha at gmail.com>
> Subject: [SR-Users] CRITICAL: dialog [dlg_timer.c:205]: Trying to
> update a bogus dlg - Kamailio 3.3.x To: sr-users at lists.sip-router.org
> Message-ID:
> <CAFvba4ipKqicMJ0qn9UNLXmhjGhf_83fzG=L5w2qVBpc7JC1GQ at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> Hi list,
>
>
> I'm having some issues with dialog module. From time to time I get the
> following error:
>
> ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)
>
> [root at proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" |
> sort | uniq -c
>
> 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy
> /usr/local/kamailio/sbin/kamailio[4100]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy
> /usr/local/kamailio/sbin/kamailio[4100]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy
> /usr/local/kamailio/sbin/kamailio[4100]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy
> /usr/local/kamailio/sbin/kamailio[4100]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy
> /usr/local/kamailio/sbin/kamailio[4100]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil)
>
> This one was flooding my log file: tl=0x2aeeea1c6840
>
>
> 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy
> /usr/local/kamailio/sbin/kamailio[4100]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)
>
> 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy
> /usr/local/kamailio/sbin/kamailio[4102]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy
> /usr/local/kamailio/sbin/kamailio[4102]:
> CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
> tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)
>
> where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this
> happening and how to fix it. (I suspect that may be some issues with our
> server memory...checking now...and will inform). However wanted to raise
> this issue in case anyone else is having the same.
>
> Thanks, Rinor
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> ------------------------------
>
>
> Message: 4
> Date: Fri, 18 Jan 2013 15:36:04 +0200
> From: Rumen Mihailov <zealas1662 at gmail.com>
> Subject: [SR-Users] Huge undertaking and in need of advices from the
> community To: sr-users at lists.sip-router.org
> Message-ID:
> <CAM4MW6j1pNazuV16SbYudgCctMpUYGBRyB4-UWdFHVjEHcG4UQ at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
>
> Hi all,
>
>
> Thank you for all the help till now.
>
>
> Until now I was trying to setup completely redundant setup with
> Kamailio + asterisk boxes and for the moment i have the following
> configuration working.
>
> 2x Kamailios (Active/Passive)
> 2x Asterisks Kamailio is load balancing I can add more boxes if needed
> MySQL Cluster for all database needs
>
>
> Now the time has come and I will need to made this setup available for
> around 10K users. I will need to become an ITSP for an Internet Provider
> that is missing the telephony part from his package, so I have a couple of
> questions.
>
> 1. Is this setup OK to cover the needs of the customer base ? hardware
> will be decent 2. The end users will most probably need ATA adapters. What
> is the best cost effective solution I can go for ? 3. Do I need any other
> piece of software apart from the SIP Proxy and asterisk as I need the cost
> of implementation to be as low as possible. 4. Pricing...I can see that
> there are a lot of plans of the competitors that include 1000 minutes for
> free...Do you have any statistics what is the avarage % usage of prepaid
> minutes ? 5. Are there any other "stones" that I might hit and should be
> aware of ?
>
> Thank you for the information
>
>
> Best regards,
> Rumen
>
>
>
>
> ------------------------------
>
>
> Message: 5
> Date: Fri, 18 Jan 2013 19:17:59 +0200
> From: Stoyan Mihaylov <stoyan.v.mihaylov at gmail.com>
> Subject: Re: [SR-Users] Huge undertaking and in need of advices from
> the	community To: "SIP Router - Kamailio (OpenSER) and SIP Express Router
> (SER) -
> Users	Mailing List" <sr-users at lists.sip-router.org>
> Message-ID:
> <CAPScudahLCQBBpN8eV=hAXCuhpyh+-2Chkyzm-ZwXMfteTpxsw at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
>
> 1. From what I read - 1 Kamailio server should be good enough for up
> to 50 K users. 3. You need Kamailio, Asterisk, Database (MySQL), and of
> course web server (Apache) for GUI - setup of clients, allow clients to
> see their calls.... You can use on separate server rtpproxy. 4. Pricing,
> depends of country. For USA, where most calls cost less then cent, 1000
> minutes are possible, but for Bulgaria - you cannot offer such minutes.
>
> 2. Adapters are very important. There are lot of cheap adapters, which
> do perfect job, but they did some troubles for me. Small and acceptable
> troubles, but for me. One of problems we saw is with power supply - after
> a year or so we saw how some of devices stopped working. And we just
> replaced their power supply. Also - you can use modified/adapted SIP
> clients - for desktop, for Android or iPhone. Some of devices, can be used
> as routers. For me standard is Grandstream - they are not cheap, but they
> just work - and I had no problems with them. I use also different unnamed
> devices, and I can try to find mails and addresses of companies for
> contacts, but we purchased them long ago.
>
>
>
> On Fri, Jan 18, 2013 at 3:36 PM, Rumen Mihailov <zealas1662 at gmail.com>
> wrote:
>
>> Hi all,
>>
>>
>> Thank you for all the help till now.
>>
>>
>> Until now I was trying to setup completely redundant setup with
>> Kamailio + asterisk boxes and for the moment i have the following
>> configuration working.
>>
>> 2x Kamailios (Active/Passive)
>> 2x Asterisks Kamailio is load balancing I can add more boxes if needed
>> MySQL Cluster for all database needs
>>
>>
>> Now the time has come and I will need to made this setup available for
>> around 10K users. I will need to become an ITSP for an Internet Provider
>> that is missing the telephony part from his package, so I have a couple
>> of questions.
>>
>> 1. Is this setup OK to cover the needs of the customer base ? hardware
>> will be decent 2. The end users will most probably need ATA adapters.
>> What is the
>> best cost effective solution I can go for ? 3. Do I need any other piece
>> of software apart from the SIP Proxy and asterisk as I need the cost of
>> implementation to be as low as possible. 4. Pricing...I can see that
>> there are a lot of plans of the competitors that include 1000 minutes
>> for free...Do you have any statistics what is the avarage % usage of
>> prepaid minutes ? 5. Are there any other "stones" that I might hit and
>> should be aware of ?
>>
>> Thank you for the information
>>
>>
>> Best regards,
>> Rumen
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> ------------------------------
>
>
> Message: 6
> Date: Fri, 18 Jan 2013 16:57:34 +0000
> From: "Scott, Matt" <mscott at homeadvisor.com>
> Subject: Re: [SR-Users] Invite - negotiate to asterisk as peer not
> kamailio To: "'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER)
> -
> Users	Mailing List'" <sr-users at lists.sip-router.org>
> Message-ID:
> <BEA4BED164E2154A9A2508A60718F76E19689107 at pexkc001.servicemagic.com>
> Content-Type: text/plain; charset="us-ascii"
>
>
> Let's see if I can do this, so something like this?
> Using the PSTN Source IP to decide on the asterisk peer?
> Not really sure on what variable dispatch will return as it's chosen ip?
>
>
> IP_auth ->Dispatch->Force Socket->Relay
>
>
> route{
>
> if (!allow_source_address("1")) { sl_send_reply("403", "Forbidden"); exit;
> };
> # load-balance dispatching on gateways group '1'
> if(!ds_select_dst("1", "10")) {
> send_reply("404", "No destination"); exit; }
> xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
>
> t_on_failure("RTF_DISPATCH");
>
> if ($si == "1.2.3.1") { force_send_socket($sndto:5065);
> } else if ($si == "1.2.3.2") {
> force_send_socket($sndto:5070);
> }
> t_relay();
>
> return;
>
>
> Most definitely, thank you for your time.
>
>
> Matt Scott
>
>
>
>
>
>
>
>
>
>
> From: sr-users-bounces at lists.sip-router.org
> [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Olle E.
> Johansson
> Sent: Friday, January 18, 2013 1:09 AM
> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
> Mailing List
> Subject: Re: [SR-Users] Invite - negotiate to asterisk as peer not
> kamailio
>
>
> 17 jan 2013 kl. 18:07 skrev "Scott, Matt"
> <mscott at homeadvisor.com<mailto:mscott at homeadvisor.com>>:
>
>
>
> Pstn->Kamailio->dispatcher->asterisk
>
>
>
> When calls come in, they are sent to asterisk, and the peer is negotiated
> as Kamailio. How can I have different peer settings, for example dtmf?
> I have an Asterisk branch that separates peers beyond the proxy
> (pinetree-1.4) based on the via received headers.
> I have it in production for Asterisk 1.4 in multiple places, but don't
> know the current state in relationship to latest source code.
>
>
>
> I'm thinking I want the call to come in, then when the invite is sent to
> asterisk, the invite should be from the pstn-peer and not Kamailio, make
> sense? Then I can have separate peers, with their own dtmf/audio settings,
> but I see the invite as:
>
> Remember that Asterisk first match incoming calls on users with the FROM
> header. You can separate based on the From header. But for PSTN, that
> doesn't really help...
>
> If you have a limited set of profiles you can add port numbers to
> Asterisk peers, like have a peer on port 5060 and another on the kamailio
> IP on 5065. Have kamailio open both sockets and use force_socket to send
> from the proper one.
>
> /O
>
>
>
>
> INVITE sip:+1866NXXNXXX@<kamailio<sip:+1866NXXNXXX@%3ckamailio> ip>:5060
> SIP/2.0
> Via: SIP/2.0/UDP <Kamailio ip>;branch=z9hG4bKc087.12db3c91.0
> Via: SIP/2.0/UDP <pstn-peer ip>:5060;branch=z9hG4bK0cB824c55100a4f2b8b
> From: <sip:+1NXXNXXXXXXXX@<pstn-peer ip>:5060;isup-oli=0>;tag=gK0c731df8
> To: <sip:+1NXXNXXNXXX@<kamailio ip>:5060>
> Call-ID: 1024240378_131584110@<pstn-peer ip>
>
>
> Any help or suggestions are very welcome..
>
>
>
> Matt Scott
>
>
>
>
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org<mailto:sr-users at lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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>
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>
> End of sr-users Digest, Vol 92, Issue 43
> ****************************************
>
>

Thanks & Regards,
Manjusha A.
Integra Micro Software Services (P) Ltd.