[SR-Users] Iptel SIP used with Bria iPhone App
I'm using an iPhone Bria app and have input Iptel SIP information in the settings, however, when I try to dial a land line I get a Forbidden (403) error message. What could be wrong? ___ 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
Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
On 13 May 2013 09:46, zhengyw zhen...@neusoft.com wrote: Hello: I have followed the steps in the guide( http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb ): 1. Could you find any mistake in my configuration file? I don't see any obvious errors apart form the incorrect table structure in video1_sipregs which I already mentioned. 2. I also wonder how could I be sure of Be sure you update the listen IP and port as well if Asterisk is running on the same You do this using the bindaddr/udpbindaddr/tcpbindaddr settings in sip.conf. You have udpbindaddr=10.11.2.47:5080 set. Perhaps it would help if you instead simply set bindaddr=10.11.2.47:5080 - it will have the same affect, as tcp is disabled anyway. system with Kamailio and Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio. This is achieved by having no entry in the secret field of the video1_sipusers table, so that Asterisk does not try to authenticate the peer, as Kamailio has already done so. -Barry thank you very much PS：attachment is kamailio、asterisk's congifure file. Best Regards zhengyw - Original Message - From: Daniel-Constantin Mierla mico...@gmail.com To: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Sent: Monday, May 13, 2013 3:26 PM Subject: Re: [SR-Users] I need you help-about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Hello, I am not that familiar to troubleshoot asterisk configuration files, but from logs I could see the resulting URI is: INVITEsip:106@(null) SIP/2.0 That is wrong, meaning something incorrect is done when setting it in asterisk. Maybe someone else can help more with asterisk. Cheers, Daniel On 5/13/13 4:02 AM, zhengyw wrote: hello daniel: thank you very much! but I can't find the problem in the asterisk. attachment is asterisk's configure file, kamailio's configure file and data.can you help with this problem? ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04 Best Regards, zhengyw kamailio.cfg http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg sip.conf http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf extconfig.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf extensions.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf db_result.txt http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt -- View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-x-and-Asterisk-10-7-0-Realtime-Integration-tp118248p118319.html Sent from the Users mailing list archive at Nabble.com. ___ 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ 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 --- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. --- ___ 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 ___ 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
Re: [SR-Users] confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b
I can't install siremis 3.0 for kaimailio after install kamailio and begin to install siremis in CRIT.log of siremis/log, i get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all. 2013/5/14 sr-users-requ...@lists.sip-router.org Mailing list subscription confirmation notice for mailing list sr-users We have received a request from 218.79.223.111 for subscription of your email address, lia...@gmail.com, to the sr-users@lists.sip-router.org mailing list. To confirm that you want to be added to this mailing list, simply reply to this message, keeping the Subject: header intact. Or visit this web page: http://lists.sip-router.org/cgi-bin/mailman/confirm/sr-users/cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b Or include the following line -- and only the following line -- in a message to sr-users-requ...@lists.sip-router.org: confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b Note that simply sending a `reply' to this message should work from most mail readers, since that usually leaves the Subject: line in the right form (additional Re: text in the Subject: is okay). If you do not wish to be subscribed to this list, please simply disregard this message. If you think you are being maliciously subscribed to the list, or have any other questions, send them to sr-users-ow...@lists.sip-router.org. -- Future Lian ___ 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
Re: [SR-Users] confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b
You must create a new email with proper subject and content, otherwise nobody will look at such subjects and you will never get answers from the right people, because it is very likely to get filtered by spam. Cheers, Daniel On 5/14/13 12:53 PM, Future Lian wrote: I can't install siremis 3.0 for kaimailio after install kamailio and begin to install siremis in CRIT.log of siremis/log, i get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all. 2013/5/14 sr-users-requ...@lists.sip-router.org mailto:sr-users-requ...@lists.sip-router.org Mailing list subscription confirmation notice for mailing list sr-users We have received a request from 218.79.223.111 for subscription of your email address, lia...@gmail.com mailto:lia...@gmail.com, to the sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org mailing list. To confirm that you want to be added to this mailing list, simply reply to this message, keeping the Subject: header intact. Or visit this web page: http://lists.sip-router.org/cgi-bin/mailman/confirm/sr-users/cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b Or include the following line -- and only the following line -- in a message to sr-users-requ...@lists.sip-router.org mailto:sr-users-requ...@lists.sip-router.org: confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b Note that simply sending a `reply' to this message should work from most mail readers, since that usually leaves the Subject: line in the right form (additional Re: text in the Subject: is okay). If you do not wish to be subscribed to this list, please simply disregard this message. If you think you are being maliciously subscribed to the list, or have any other questions, send them to sr-users-ow...@lists.sip-router.org mailto:sr-users-ow...@lists.sip-router.org. -- Future Lian ___ 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ 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
Re: [SR-Users] 4.0 forking behaviour
On 5/14/13 2:29 AM, Alex Balashov wrote: Daniel, Thank you for your help. FYI, stopping my use of append_branch() everywhere solved the problem. I was unaware that it had become an essentially deprecated requirement. it was deprecated but should have stayed harmless. Do you touch anything related to next hop address after append_branch()? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ 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
Re: [SR-Users] Iptel SIP used with Bria iPhone App
On 5/13/13 10:40 PM, JE wrote: I'm using an iPhone Bria app and have input Iptel SIP information in the settings, however, when I try to dial a land line I get a Forbidden (403) error message. What could be wrong? afaik, iptel.org does not offer pstn termination out of the box, you have to configure your own gateway there. Anyhow, a proper answer you should get from the mailing list associated with the services: - http://lists.iptel.org/mailman/listinfo/services Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ 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
Re: [SR-Users] event_route
Hello, why do you need to process cancel requests? They have special routing requirements related to associated invite and sent from tm directly. The event route is for the requests sent by modules via tm. Haven't tried, but maybe onsend_route will capture it. Cheers, Daniel On 5/14/13 12:30 AM, Bruno Bresciani wrote: Hi All, in a call forking, after one branch answer the call (200 OK reply), a CANCEL SIP message has been sending to other/another branch(es) and I need to process this/these cancellations in configuration file. After reading some documentations, I discovered there is event_route[tm:local-request] block, which is executed when tm generates internally and sends a SIP request, Such cases are: SIP messages sent by msilo module SIP messages sent by presence server SIP messages sent by dialog module SIP messages sent via MI or CTL interfaces I didn't understand very well this cases, so I insert event_route block in my kamailio.cfg but neither CANCEL SIP message or other requests generated by tm module was handled by event_route. I must be using wrong concept to handle this CANCEL SIP message, it's possible handle this messages in configuration file? Best Regards ___ 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ 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
Re: [SR-Users] event_route
Thank's Daniel, I need process cancel requests to delete sessions on rtp proxy... In a call forking, when I need forking to multiple destinations on different network segments, requiring different rtpproxy parameters, I use the “extra_id_pv” and b parameter in the rtpproxy_offer() function to created a session RTP to each branch of calling in rtp proxy. When one of branch answer the call, I want delete the other(s) session(s) on rtp proxy (unforce_rtp_proxy()) using the CANCEL SIP message to do this, but I can't get handle it in configuration file... I am wrong trying use the CANCEL message to delete session rtp? I will try onsend_route to handle... Best Regards 2013/5/14 Daniel-Constantin Mierla mico...@gmail.com Hello, why do you need to process cancel requests? They have special routing requirements related to associated invite and sent from tm directly. The event route is for the requests sent by modules via tm. Haven't tried, but maybe onsend_route will capture it. Cheers, Daniel On 5/14/13 12:30 AM, Bruno Bresciani wrote: Hi All, in a call forking, after one branch answer the call (200 OK reply), a CANCEL SIP message has been sending to other/another branch(es) and I need to process this/these cancellations in configuration file. After reading some documentations, I discovered there is event_route[tm:local-request] block, which is executed when tm generates internally and sends a SIP request, Such cases are: SIP messages sent by msilo module SIP messages sent by presence server SIP messages sent by dialog module SIP messages sent via MI or CTL interfaces I didn't understand very well this cases, so I insert event_route block in my kamailio.cfg but neither CANCEL SIP message or other requests generated by tm module was handled by event_route. I must be using wrong concept to handle this CANCEL SIP message, it's possible handle this messages in configuration file? Best Regards ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ 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 ___ 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
Re: [SR-Users] Kamailio for Debian blog?
2013/5/13 Victor Seva linuxman...@torreviejawireless.org: I'm going to migrate Debian kamailio repository to git ASAP. Well, finally I've created the git repository with git-buildpackage importing just the 3.0.1-1 and 4.0.1-1 versions. I think it's enough ___ 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
Re: [SR-Users] kamailio 4.0.1 and nokia e72 udp
On 5/13/2013 2:17 PM, hiro wrote: It doesn't seem to be the router/NAT's problem though, as the Nokia itself binds to the right port at first, then gives up on it and changes to a port 20 higher instead. The second bind is also the one that it advertises in it's sdp. But that tip with listen for port changes is good, it would only be problematic if there are multiple concurrent calls from the same (perhaps NATted) IP, right? No, it would not be a problem because multiple calls would go to different destination UDP ports at the server. RTPproxy would be able to match them all dynamically even if the source port on the client (or clients) changes constantly during the calls. We have tested this extensively and has worked flawlessly for years. I works so well that even if the IP address on the client changes (like a DSL session going down and up again), the rtpproxy will match the new stream from the client immediatly. -- Technical Support http://www.cellroute.net On 5/13/13, Andres and...@telesip.net wrote: On 5/11/2013 4:29 PM, hiro wrote: using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind NAT registered via UDP I get no voice. The e72 strangely sends a single udp packet from a wrong port (49152) before the rtp stream should start. This quirk of the e72 doesn't seem to work well with rtpproxy if the following analysis is true: rtpproxy detects that single UDP packet from the wrong port and so we think that is where everything else will also come from and stop listening on other ports. we then also answer on that wrong port. Although all subsequent incoming packets arrive from the expected (49172) port sent also in the sdp and to the right one we had sent in the sdp earlier we never receive them, because we still listen on that wrong port with that one bogus packet. I have seen such behavior before from other cheap NAT routers. The solution was to keep rtpproxy in listen mode for port changes always. That way it will keep working no matter how many times the port changes on the client side. We are still running an older version of rtpproxy so I cannot comment on how to patch the lastest version. The version we have is 1.0.2 and the modification we did was to file main.c and commented the following aroubd line 1415: /*sp-canupdate[ridx] = 0;*/ Thats it. -- Technical Support http://www.cellroute.net ___ 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 -- Technical Support http://www.cellroute.net ___ 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
[SR-Users] Kamailio 3.3.x is crashing frequently...
Hi, When the Server is running it is crashing occasionally. (gdb) bac #0 0x0043359e in ?? () #1 0x in ?? () (gdb) file /usr/local/sbin/kamailio Reading symbols from /usr/local/sbin/kamailio...done. (gdb) bac #0 free_lump_list (lump_list=value optimized out) at data_lump.c:504 #1 del_nonshm_lump (lump_list=value optimized out) at data_lump.c:661 #2 0x7f571ff6cd9c in ?? () #3 0x008db480 in mem_pool () #4 0x7f571ff6c7e1 in ?? () #5 0x0139086e in ?? () #6 0x0039294546e1 in ?? () #7 0x0140c7d0 in ?? () #8 0x7f561f7acb80 in ?? () #9 0x7f561f3fa6e0 in ?? () #10 0x7f561f3fa6e0 in ?? () #11 0x0001 in ?? () #12 0x in ?? () What could be going wrong here? Krish Kura ___ 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
[SR-Users] home pbx server experience
I would like to share my experience with kamailio and other home pbx servers. Kamailio on my kirkwood home router for my 6 SIP users is perhaps overkill: I don't really need mysql and scalability. But at last I finally managed to make calling between registered users work stable. My voip clients only work in all NAT scenarios if I work around some bugs: to use csipsimple on android I had to change rtpproxy_manage() to rtpproxy_manage(c) in kamailio's default config, so that problems with conflicting c: entries in the SDP go away. I propose kamailio could ship with a special example kamailio-compatible.cfg that doesn't try to be RFC compliant, but compatible to the most common voip clients. Right now the only thing I would change for this is the option for rtpproxy_manage, but I'm sure others will know more common quirks that could safely be enabled to increase compatibility. I think this compatibility idea is what yate sticks to for their defaults. In freeswitch you also have to do it all manually, and it's much more work to figure things out in their enormous config files. The other SIP proxies I had tried before kamailio officially fit all my requirements, including support for multihomed dynamic IPs, but contrary to their claims it didn't work. Yate was easy to set up, but the default dialplan is more confusing than powerful and after having made everything work I realised yate was clogging my CPU and RAM and after some time always randomly stopped working. This is with only 2 users connected! It also wasn't possible to fix NAT sdp while leaving the codecs section in the SDP alone at the same time. I tried to debug the code, but the C++ was so complex that I had to give up. Freeswitch was much more difficult to setup, a multihomed setup with dynamic IP was super buggy and it also didn't help that the unintuitive configuration is all in complex unreadable XML configuration files. Kamailio and rtpproxy don't officially support dynamic IP address, but I can just restart both each time my DSL provider forces me to a new IP address. This happens automatically in the night and is no big hassle really. The most simple, least-featureful solution works best it seems. Now the last problem I have with kamailio: I don't know how to connect my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). I would like a simple way to do this, preferably without other features that always seem to complicate the matters. Is there something more lightweight and simple than asterisk, freeswitch and yate, that people use successfully for this task together with kamailio and rtpproxy? u ___ 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
Re: [SR-Users] home pbx server experience
Do you have an example of the home pbx config you like in kamailio? Fred Posner | LOD / Team Forrest ph. 503-914-0999 | f...@lod.com | qxork.com On 5/14/13 10:27 AM, u wrote: I would like to share my experience with kamailio and other home pbx servers. Kamailio on my kirkwood home router for my 6 SIP users is perhaps overkill: I don't really need mysql and scalability. But at last I finally managed to make calling between registered users work stable. My voip clients only work in all NAT scenarios if I work around some bugs: to use csipsimple on android I had to change rtpproxy_manage() to rtpproxy_manage(c) in kamailio's default config, so that problems with conflicting c: entries in the SDP go away. I propose kamailio could ship with a special example kamailio-compatible.cfg that doesn't try to be RFC compliant, but compatible to the most common voip clients. Right now the only thing I would change for this is the option for rtpproxy_manage, but I'm sure others will know more common quirks that could safely be enabled to increase compatibility. I think this compatibility idea is what yate sticks to for their defaults. In freeswitch you also have to do it all manually, and it's much more work to figure things out in their enormous config files. The other SIP proxies I had tried before kamailio officially fit all my requirements, including support for multihomed dynamic IPs, but contrary to their claims it didn't work. Yate was easy to set up, but the default dialplan is more confusing than powerful and after having made everything work I realised yate was clogging my CPU and RAM and after some time always randomly stopped working. This is with only 2 users connected! It also wasn't possible to fix NAT sdp while leaving the codecs section in the SDP alone at the same time. I tried to debug the code, but the C++ was so complex that I had to give up. Freeswitch was much more difficult to setup, a multihomed setup with dynamic IP was super buggy and it also didn't help that the unintuitive configuration is all in complex unreadable XML configuration files. Kamailio and rtpproxy don't officially support dynamic IP address, but I can just restart both each time my DSL provider forces me to a new IP address. This happens automatically in the night and is no big hassle really. The most simple, least-featureful solution works best it seems. Now the last problem I have with kamailio: I don't know how to connect my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). I would like a simple way to do this, preferably without other features that always seem to complicate the matters. Is there something more lightweight and simple than asterisk, freeswitch and yate, that people use successfully for this task together with kamailio and rtpproxy? u ___ 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 ___ 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
Re: [SR-Users] kamailio 4.0.1 and nokia e72 udp
makes sense, cool. now i just have to find a version of rtpproxy that supports this stuff :) On 5/14/13, Andres and...@telesip.net wrote: On 5/13/2013 2:17 PM, hiro wrote: It doesn't seem to be the router/NAT's problem though, as the Nokia itself binds to the right port at first, then gives up on it and changes to a port 20 higher instead. The second bind is also the one that it advertises in it's sdp. But that tip with listen for port changes is good, it would only be problematic if there are multiple concurrent calls from the same (perhaps NATted) IP, right? No, it would not be a problem because multiple calls would go to different destination UDP ports at the server. RTPproxy would be able to match them all dynamically even if the source port on the client (or clients) changes constantly during the calls. We have tested this extensively and has worked flawlessly for years. I works so well that even if the IP address on the client changes (like a DSL session going down and up again), the rtpproxy will match the new stream from the client immediatly. -- Technical Support http://www.cellroute.net On 5/13/13, Andres and...@telesip.net wrote: On 5/11/2013 4:29 PM, hiro wrote: using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind NAT registered via UDP I get no voice. The e72 strangely sends a single udp packet from a wrong port (49152) before the rtp stream should start. This quirk of the e72 doesn't seem to work well with rtpproxy if the following analysis is true: rtpproxy detects that single UDP packet from the wrong port and so we think that is where everything else will also come from and stop listening on other ports. we then also answer on that wrong port. Although all subsequent incoming packets arrive from the expected (49172) port sent also in the sdp and to the right one we had sent in the sdp earlier we never receive them, because we still listen on that wrong port with that one bogus packet. I have seen such behavior before from other cheap NAT routers. The solution was to keep rtpproxy in listen mode for port changes always. That way it will keep working no matter how many times the port changes on the client side. We are still running an older version of rtpproxy so I cannot comment on how to patch the lastest version. The version we have is 1.0.2 and the modification we did was to file main.c and commented the following aroubd line 1415: /*sp-canupdate[ridx] = 0;*/ Thats it. -- Technical Support http://www.cellroute.net ___ 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 -- Technical Support http://www.cellroute.net ___ 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
Re: [SR-Users] home pbx server experience
Hi... Use UAC module to manage registrations and play a little with the config (INVITE section) to forward output calls correctly. --- Edson. Em 14/05/2013 11:27, u escreveu: I would like to share my experience with kamailio and other home pbx servers. Kamailio on my kirkwood home router for my 6 SIP users is perhaps overkill: I don't really need mysql and scalability. But at last I finally managed to make calling between registered users work stable. My voip clients only work in all NAT scenarios if I work around some bugs: to use csipsimple on android I had to change rtpproxy_manage() to rtpproxy_manage(c) in kamailio's default config, so that problems with conflicting c: entries in the SDP go away. I propose kamailio could ship with a special example kamailio-compatible.cfg that doesn't try to be RFC compliant, but compatible to the most common voip clients. Right now the only thing I would change for this is the option for rtpproxy_manage, but I'm sure others will know more common quirks that could safely be enabled to increase compatibility. I think this compatibility idea is what yate sticks to for their defaults. In freeswitch you also have to do it all manually, and it's much more work to figure things out in their enormous config files. The other SIP proxies I had tried before kamailio officially fit all my requirements, including support for multihomed dynamic IPs, but contrary to their claims it didn't work. Yate was easy to set up, but the default dialplan is more confusing than powerful and after having made everything work I realised yate was clogging my CPU and RAM and after some time always randomly stopped working. This is with only 2 users connected! It also wasn't possible to fix NAT sdp while leaving the codecs section in the SDP alone at the same time. I tried to debug the code, but the C++ was so complex that I had to give up. Freeswitch was much more difficult to setup, a multihomed setup with dynamic IP was super buggy and it also didn't help that the unintuitive configuration is all in complex unreadable XML configuration files. Kamailio and rtpproxy don't officially support dynamic IP address, but I can just restart both each time my DSL provider forces me to a new IP address. This happens automatically in the night and is no big hassle really. The most simple, least-featureful solution works best it seems. Now the last problem I have with kamailio: I don't know how to connect my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). I would like a simple way to do this, preferably without other features that always seem to complicate the matters. Is there something more lightweight and simple than asterisk, freeswitch and yate, that people use successfully for this task together with kamailio and rtpproxy? u ___ 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 . ___ 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
[SR-Users] HELP! I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end.
I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end. in CRIT.log of siremis/log, I get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all. ___ 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
[SR-Users] Kamailio crash
Hi we have developed the custom module where in we have created shared memory area which is not used by any other module or process and we locked it using locks as recommended. but because of locks the kamailio is getting crashed every now and then . is it really important to protect the shared memory area with locks. if yes then how to avoid the crash ?? so to locate the crash we built kamailio wit h MM_DBG , and other GCC debug flags . but we are getting error /No symbol table info available/ . how to locate the bugs. the kamailio is built wit h 1 MB of shared memory on MIPS based VOIP gateway which is having * MB of flash and 16 MB or memory (RAM) . thanks in advance . ___ 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
