Thanks a lot for quick response. On Sat, January 19, 2013 3:59 am, sr-users-requ...@lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users@lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-requ...@lists.sip-router.org > > You can reach the person managing the list at > sr-users-ow...@lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > > 1. Mid call announcement(kamailio server) > (madhumanju...@integramicro.com) > 2. Re: Mid call announcement(kamailio server) > (r...@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: madhumanju...@integramicro.com > Subject: [SR-Users] Mid call announcement(kamailio server) > To: sr-users@lists.sip-router.org > Message-ID: > <39960.61.8.152.138.1358509935.squir...@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: r...@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@lists.sip-router.org> > Message-ID: > <38001.212.40.242.42.1358518489.squir...@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 <rinorho...@gmail.com> > Subject: [SR-Users] CRITICAL: dialog [dlg_timer.c:205]: Trying to > update a bogus dlg - Kamailio 3.3.x To: sr-users@lists.sip-router.org > Message-ID: > <CAFvba4ipKqicMJ0qn9UNLXmhjGhf_83fzG=l5w2qvbpc7jc...@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@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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.sip-router.org/pipermail/sr-users/attachments/20130118/1888 > 8481/attachment-0001.htm> > > > ------------------------------ > > > Message: 4 > Date: Fri, 18 Jan 2013 15:36:04 +0200 > From: Rumen Mihailov <zealas1...@gmail.com> > Subject: [SR-Users] Huge undertaking and in need of advices from the > community To: sr-users@lists.sip-router.org > Message-ID: > <cam4mw6j1pnazuv16sbyudgcctmpuygbryb4-uwdfhvjehcg...@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.mihay...@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@lists.sip-router.org> > Message-ID: > <CAPScudahLCQBBpN8eV=haxcuhpyh+-2chkyzm-zwxmftetp...@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 <zealas1...@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@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" <msc...@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@lists.sip-router.org> > Message-ID: > <bea4bed164e2154a9a2508a60718f76e19689...@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-boun...@lists.sip-router.org > [mailto:sr-users-boun...@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" > <msc...@homeadvisor.com<mailto:msc...@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@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.sip-router.org/pipermail/sr-users/attachments/20130118/8ce9 > 6979/attachment.htm> > > > ------------------------------ > > > _______________________________________________ > sr-users mailing list sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > End of sr-users Digest, Vol 92, Issue 43 > **************************************** > >
Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd. _______________________________________________ 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
