Re: [SR-Users] dns lookup and store IP in avp..
On Mon, Mar 19, 2012 at 4:56 PM, MingHon gming...@gmail.com wrote: Thanks Daniel... :) On Mon, Mar 19, 2012 at 4:18 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 3/19/12 8:22 AM, Klaus Darilion wrote: No. If you really need it, then you can use exec() and a tool like dig or host. apart of exec module functions, another option would be using an embedded language script (Lua/Perl...). Cheers, Daniel regards Klaus On 19.03.2012 03:19, MingHon wrote: Hi All, is this function available?? http://sourceforge.net/**tracker/?func=detailaid=** 2687695group_id=139143atid=**743023http://sourceforge.net/tracker/?func=detailaid=2687695group_id=139143atid=743023 http://sourceforge.net/**tracker/?func=detailaid=** 2687695group_id=139143atid=**743023http://sourceforge.net/tracker/?func=detailaid=2687695group_id=139143atid=743023 Thanks :) -- Regards, MingHon __**_ 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-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.**php/kamailio-advanced-**training/http://www.asipto.com/index.php/kamailio-advanced-training/ -- Regards, MingHon -- Regards, MingHon ___ 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] dns lookup and store IP in avp..
And Klaus.. :) On Mon, Mar 19, 2012 at 4:56 PM, MingHon gming...@gmail.com wrote: On Mon, Mar 19, 2012 at 4:56 PM, MingHon gming...@gmail.com wrote: Thanks Daniel... :) On Mon, Mar 19, 2012 at 4:18 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 3/19/12 8:22 AM, Klaus Darilion wrote: No. If you really need it, then you can use exec() and a tool like dig or host. apart of exec module functions, another option would be using an embedded language script (Lua/Perl...). Cheers, Daniel regards Klaus On 19.03.2012 03:19, MingHon wrote: Hi All, is this function available?? http://sourceforge.net/**tracker/?func=detailaid=** 2687695group_id=139143atid=**743023http://sourceforge.net/tracker/?func=detailaid=2687695group_id=139143atid=743023 http://sourceforge.net/**tracker/?func=detailaid=** 2687695group_id=139143atid=**743023http://sourceforge.net/tracker/?func=detailaid=2687695group_id=139143atid=743023 Thanks :) -- Regards, MingHon __**_ 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-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.**php/kamailio-advanced-**training/http://www.asipto.com/index.php/kamailio-advanced-training/ -- Regards, MingHon -- Regards, MingHon -- Regards, MingHon ___ 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] dns lookup and store IP in avp..
Hi All, is this function available?? http://sourceforge.net/tracker/?func=detailaid=2687695group_id=139143atid=743023 Thanks :) -- Regards, MingHon ___ 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] commands out of sync you can't run this command now
Hello, i attached the log/messages files please take a look... Cheers, MingHon On Tue, Dec 13, 2011 at 4:58 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 12/13/11 9:24 AM, Daniel-Constantin Mierla wrote: Hello, are you using any private developed module or extension? There seems to be a buffer overflow as well. if you don't have any private extension, can you send directly to me the entire log from start of kamailio with following global parameters: debug=3 memlog=4 memdbg=4 errata - the values of the global parameters were wrong -- actually the above three lines should be: debug=3 memlog=1 memdbg=1 Cheers, Daniel Cheers, Daniel On 12/13/11 8:51 AM, MingHon wrote: im using version: kamailio 3.1.4 (i386/linux) after kamctl start it return INFO: Starting Kamailio : INFO: started (pid: 3589) ps aux | grep 3589 root 3609 0.0 0.0 4012 704 pts/0S+ 15:48 0:00 grep 3589 here is the log message. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Commands out of sync; you can't run this command now Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db_query.c:101]: error while submitting query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db.c:366]: error in db_query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db.c:405]: querying version for table location Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: usrloc [dlist.c:491]: error during table version check. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: : core [mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c (address 0x838b134) end overwritten(0, 0)! Dec 13 15:43:41 hostname kamailio: ERROR: core [daemonize.c:307]: Main process exited before writing to pipe On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 12/12/11 4:09 PM, MingHon wrote: btw im using kamailio 3.1 which one exactly? Just send output of 'kamailio -V'. Also, watch the log messages with debug=3 and get the one related to the error messages to see which module is executing that command. Cheers, Daniel, On Mon, Dec 12, 2011 at 11:08 PM, MingHon gming...@gmail.com wrote: Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon -- Regards, MingHon ___ 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://linkedin.com/in/miconda -- http://twitter.com/miconda -- Regards, MingHon ___ 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://linkedin.com/in/miconda -- http://twitter.com/miconda ___ 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://linkedin.com/in/miconda -- http://twitter.com/miconda -- Regards, MingHon -- Regards, MingHon ___ 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] Fwd: commands out of sync you can't run this command now
ya debug is set to 3 already.. also #!define WITH_DEBUG dont hav any -d command.. On Tue, Dec 13, 2011 at 9:06 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: I cannot see any DEBUG messages, are you sure the debug=3 in your config and you dont have other -d command line parameters? Cheers, Daniel On 12/13/11 12:58 PM, MingHon wrote: Hi, the attachment was 2mb i remove it from approval. i did sentto ur email too.. anyway i attached the file with 7 zip.. reduced to 60k now.. -- Regards, MingHon On Tue, Dec 13, 2011 at 7:00 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, have you sent them to me or to mailing list? On the mailing list I could not find an attachment, also I didn't get an email on myself only. Cheers, Daniel On 12/13/11 10:39 AM, MingHon wrote: Hello, i attached the log/messages files please take a look... Cheers, MingHon On Tue, Dec 13, 2011 at 4:58 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 12/13/11 9:24 AM, Daniel-Constantin Mierla wrote: Hello, are you using any private developed module or extension? There seems to be a buffer overflow as well. if you don't have any private extension, can you send directly to me the entire log from start of kamailio with following global parameters: debug=3 memlog=4 memdbg=4 errata - the values of the global parameters were wrong -- actually the above three lines should be: debug=3 memlog=1 memdbg=1 Cheers, Daniel Cheers, Daniel On 12/13/11 8:51 AM, MingHon wrote: im using version: kamailio 3.1.4 (i386/linux) after kamctl start it return INFO: Starting Kamailio : INFO: started (pid: 3589) ps aux | grep 3589 root 3609 0.0 0.0 4012 704 pts/0S+ 15:48 0:00 grep 3589 here is the log message. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Commands out of sync; you can't run this command now Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db_query.c:101]: error while submitting query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db.c:366]: error in db_query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db.c:405]: querying version for table location Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: usrloc [dlist.c:491]: error during table version check. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: : core [mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c (address 0x838b134) end overwritten(0, 0)! Dec 13 15:43:41 hostname kamailio: ERROR: core [daemonize.c:307]: Main process exited before writing to pipe On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 12/12/11 4:09 PM, MingHon wrote: btw im using kamailio 3.1 which one exactly? Just send output of 'kamailio -V'. Also, watch the log messages with debug=3 and get the one related to the error messages to see which module is executing that command. Cheers, Daniel, On Mon, Dec 12, 2011 at 11:08 PM, MingHon gming...@gmail.com wrote: Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon -- Regards, MingHon ___ 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://linkedin.com/in/miconda -- http://twitter.com/miconda -- Regards, MingHon ___ 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://linkedin.com/in/miconda -- http://twitter.com/miconda ___ 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://linkedin.com/in/miconda -- http://twitter.com/miconda -- Regards, MingHon -- Regards, MingHon ___ 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://linkedin.com/in/miconda
[SR-Users] commands out of sync you can't run this command now
Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon ___ 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] commands out of sync you can't run this command now
btw im using kamailio 3.1 On Mon, Dec 12, 2011 at 11:08 PM, MingHon gming...@gmail.com wrote: Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon -- Regards, MingHon ___ 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] commands out of sync you can't run this command now
im using version: kamailio 3.1.4 (i386/linux) after kamctl start it return INFO: Starting Kamailio : INFO: started (pid: 3589) ps aux | grep 3589 root 3609 0.0 0.0 4012 704 pts/0S+ 15:48 0:00 grep 3589 here is the log message. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Commands out of sync; you can't run this command now Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db_query.c:101]: error while submitting query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db.c:366]: error in db_query Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: core [db.c:405]: querying version for table location Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: ERROR: usrloc [dlist.c:491]: error during table version check. Dec 13 15:43:41 hostname /usr/local/sbin/kamailio[3589]: : core [mem/q_malloc.c:146]: BUG: qm_*: fragm. 0x838b11c (address 0x838b134) end overwritten(0, 0)! Dec 13 15:43:41 hostname kamailio: ERROR: core [daemonize.c:307]: Main process exited before writing to pipe On Tue, Dec 13, 2011 at 12:53 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 12/12/11 4:09 PM, MingHon wrote: btw im using kamailio 3.1 which one exactly? Just send output of 'kamailio -V'. Also, watch the log messages with debug=3 and get the one related to the error messages to see which module is executing that command. Cheers, Daniel, On Mon, Dec 12, 2011 at 11:08 PM, MingHon gming...@gmail.com wrote: Hi List, im doing a fresh install, everything installed successfully but when i try to start kamailio with kamctl start i receive this error. when im using mysql 5.5 rpm version i get this error if install with mysql 5.0 rpm i dont receive that error it run perfectly. is mysql 5.5 incompatible yet? Thanks, -- Regards, MingHon -- Regards, MingHon ___ 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://linkedin.com/in/miconda -- http://twitter.com/miconda -- Regards, MingHon ___ 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] question bout UA and BOX in same NAT
Hi, Thanks Daniel. It work like charm :) Cheers, On Thu, Sep 8, 2011 at 10:22 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 9/8/11 3:12 AM, MingHon wrote: Hi, Sorry, let me explain my situation. my box is behind nat with private ip 192.168.2.3 and nat ip is 60.x.x.x and asterisk is same nat as the box with private ip 192.168.2.23 now let say i got 2 UA. one UA is located on different nat and another UA is same nat with the box. UA1- private ip 10.10.10.123 and nat ip is 60.y.y.y UA2- private ip 192.168.2.100 and nat ip is 60.x.x.x with advertised_address = 60.x.x.x UA1 successfully register. UA2 registration fail. because when box send option method, UA2 reply 200OK to 60.x.x.x is there anyway to rewrite the via header when sending to UA2 ? then maybe is better to use set_advertise_address based on source IP, not only as global option: http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertised_address So depending who is sending the request, you set the right advertised address per request. Cheers, Daniel Thanks and Regards, MingHon On Wed, Sep 7, 2011 at 4:40 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 9/6/11 8:53 AM, MingHon wrote: Hello List, Can i have UA in the same nat with the box and also UA in different nat at the same time? My box is currrently working behind nat with ip 192.168.2.3 and advertised_address is set to sip.mydomain.com is it possible to append the via header replace the sip.mydomain.com to 192.168.2.3 ? not sure I understand what you want to achieve. First you say that you want advertised address to be sip.mydomain.com but then you want it to be 192.168.2.3, which would be if you don't set advertised address parameter. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda ___ 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.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda -- Regards, MingHon ___ 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] question bout UA and BOX in same NAT
Hi, Sorry, let me explain my situation. my box is behind nat with private ip 192.168.2.3 and nat ip is 60.x.x.x and asterisk is same nat as the box with private ip 192.168.2.23 now let say i got 2 UA. one UA is located on different nat and another UA is same nat with the box. UA1- private ip 10.10.10.123 and nat ip is 60.y.y.y UA2- private ip 192.168.2.100 and nat ip is 60.x.x.x with advertised_address = 60.x.x.x UA1 successfully register. UA2 registration fail. because when box send option method, UA2 reply 200OK to 60.x.x.x is there anyway to rewrite the via header when sending to UA2 ? Thanks and Regards, MingHon On Wed, Sep 7, 2011 at 4:40 PM, Daniel-Constantin Mierla mico...@gmail.comwrote: Hello, On 9/6/11 8:53 AM, MingHon wrote: Hello List, Can i have UA in the same nat with the box and also UA in different nat at the same time? My box is currrently working behind nat with ip 192.168.2.3 and advertised_address is set to sip.mydomain.com is it possible to append the via header replace the sip.mydomain.com to 192.168.2.3 ? not sure I understand what you want to achieve. First you say that you want advertised address to be sip.mydomain.com but then you want it to be 192.168.2.3, which would be if you don't set advertised address parameter. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda ___ 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] bypass rtp traffic.
anyone?? ___ 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] bypass rtp traffic.
Hello List, im still trying but no luck. asterisk canreinvite already set to yes now im testing in lan i setup kamailio and asterisk in same lan kamailiortpproxy on 192.168.2.3 and asterisk on 192.168.2.23 canreinvite=yes in asterisk. when both ua in the same lan register directly to asterisk the reinvite work. both ua will have and direct media flow [ua1][ua2] | | x | v [asterisk] when ua register to kamailio the audio work and the reinvite message is same as the first invite message. [ua1][kamailio][ua2] | ^ | | | | v | [asterisk] how do i stop the media flow between kamailio and asterisk? make kamailio relay the rtp between both ua. [ua1][kamailio][ua2] | ^ x x | | v | [asterisk] anyone could give some hint? thanks in adv. -- Regards, MingHon ___ 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] bypass rtp traffic.
Hi klaus, here is my ngrep i paste it pastebin pls take a look. http://pastebin.com/eHtMXvEx thanks in adv. -- Regards, MingHon ___ 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] bypass rtp traffic.
Hi, after asterisk reinvite i get status 491: request pending. after few seconds i hang up both UA then one of the UA will start ring. please advice. -- Thanks and Regards, MingHon ___ 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] bypass rtp traffic.
Hello, anyone? currently my setup look like this. when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk. [UA1] --(rtp)-- [Kamailio/RTPProxy] --(rtp)-- [UA2] ^ | RTP TRAFFIC | v [ASTERISK] what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio but not to asterisk. can kamailio handle the rtp traffic it own? [UA1] --(rtp)-- [Kamailio/RTPProxy] --(rtp)-- [UA2] ^ | X | v [ASTERISK] Thanks in advance. -- Regards, MingHon ___ 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] bypass rtp traffic.
Hi, yup i tried canreinivte=yes in sip.conf and also in the extension database. urm how bout having direct rtp traffic and also relay rtp traffic in my setup? example, UA1 and UA2 is at the same nat. so UA1 and UA2 will have direct rtp traffic. UA1 --(rtp)-- UA2. and UA3 and UA4 both behind different nat will need relay rtp traffic. when invite compare the received:ip_address. UA3 --(rtp)-- KAMAILIO --(rtp)-- UA4 issit possible to have both? urm ya what is the variable for the received:ip_address ? Thanks. -- Regards, MingHon On Thu, Jul 14, 2011 at 11:22 PM, Carsten Bock cars...@ng-voice.com wrote: Hi, have you tried canreinvite=yes on your Asterisk-box? If that does not help, there is probably no way to make the RTP-Traffic bypass your asterisk box... Carsten 2011/7/14 MingHon gming...@gmail.com: Hello, anyone? currently my setup look like this. when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk. [UA1] --(rtp)-- [Kamailio/RTPProxy] --(rtp)-- [UA2] ^ | RTP TRAFFIC | v [ASTERISK] what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio but not to asterisk. can kamailio handle the rtp traffic it own? [UA1] --(rtp)-- [Kamailio/RTPProxy] --(rtp)-- [UA2] ^ | X | v [ASTERISK] Thanks in advance. -- Regards, MingHon -- Carsten Bock http://www.ng-voice.com mailto:cars...@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 ~ Checkout SIP-Provider CE: http://www.sipwise.com/products/spce/overview/ ___ 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] bypass rtp traffic.
Hi, i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still trying to send rtp traffic to asterisk. and asterisk did not forward the rtp traffic back to kamailio/rtpproxy then i will get no audio on the ua. please adv. thanks, Regards, MingHon ___ 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] bypass rtp traffic.
Hi List, i would like to know is it possible to bypass the rtp traffic forwarding to asterisk server? my kamailio and rtpproxy is on the same box and asterisk is on the other box. can kamailio/rtpproxy handle the rtp traffic without forwarding to asterisk box? thanks in advance. -- Regards, MingHon ___ 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] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi, Thanks! do you think willl any other version will work? like version 3.0.x? Cheers, MingHon ___ 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] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Alright thank you very much. i wil try to install it from trunk. Will report back later. :) Regards, MingHon ___ 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] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi!! i finally able to replace the ip addr in the sdp body. but there is still some issue. will check tmr. anyway thank you very much. =) -- Regards, MingHon ___ 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] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
hello List,
anyone could give some hints??
im still unable to rewrite the sdp body.
hope to hear from you all.
thanks
--
Regards,
MingHon
On Tue, Jul 5, 2011 at 3:49 PM, MingHon gming...@gmail.com wrote:
Hi List,
im facing an issue that my kamailio proxy did not replace the ip address in
the invite and 200OK sdp body.
my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user
my kamailio is listening on 192.168.1.3, also
define: advertised_address=175.136.223.112; advertised_port=5060;
and my asterisk is on 192.168.1.23.
sip signalling and rtp port forwarded to kamailio.
uacs from another nat register successfully.
if i put 2 lines of force_rtp_proxy(fcow,175.136.223.112);
i will get double ip addr in c and o but kamailio ignore my ip addr.
example i will get
c=IN IP4 192.168.1.3192.168.1.3
here is part of my simple script.
hope you can help.
thank you very much.
---cfg---
route[RTPPROXY] {
#!ifdef WITH_NAT
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy(fcow,175.136.223.112);
#force_rtp_proxy(fcow,175.136.223.112);
xlog(L_INFO,offer);
}
if (!has_totag()) add_rr_param(;nat=yes);
#!endif
return;
}
--
and here is the wireshark for uac INVITE and OK.
---INVITE-
ve0
EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0
Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080
Max-Forwards: 69
From: 101 sip:1...@aextddns.dyndns.info;tag=as032358a3
To: sip:102@192.168.1.3:5060
Contact: sip:102@192.168.1.23:5080
Call-ID: 416f6e09674ae9671bb7144a1cb11...@aextddns.dyndns.info
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Tue, 05 Jul 2011 07:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 327
v=0
o=root 1639709788 1639709788 IN IP4 192.168.1.3
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.3
t=0 0
m=audio 10072 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
---200OK---
e90
ElE;pX4tSIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.200:5062
;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416
Record-Route: sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes
From: 101 sip:1...@aextddns.dyndns.info;tag=1796959074
To: sip:1...@aextddns.dyndns.info;tag=as2e4c0125
Call-ID: 1985782590@192.168.2.200
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.23:5080
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 403900934 403900934 IN IP4 192.168.1.23
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.23
t=0 0
m=audio 14420 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
My kamailio log.
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3
INFO: script: offer
-
double force_rtp_proxy
kamailio - asterisk [INVITE]-
Pyi-}E7V@:#pINVITE sip:1...@aextddns.dyndns.info SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=640933430;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0
Via: SIP/2.0/UDP 192.168.2.200:5062
;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648
From: 101 sip:1...@aextddns.dyndns.info;tag=640933430
To: sip:1...@aextddns.dyndns.info
Call-ID: 1909950509@192.168.2.200
CSeq: 21 INVITE
Contact: sip:101@175.138.21.31:2788
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
User-Agent: T20 9.41.0.80
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 334
v=0
o=20073 20073 IN IP4 192.168.1.3192.168.1.3
s=SDP data
c=IN IP4 192.168.1.3192.168.1.3
t=0 0
m=audio 1006410064 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=nortpproxy:yes
a=nortpproxy:yes
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi Carsten, I tried RTPProxy using external address. It work out from the box without any problem. UA successfully registered and RTP-Traffic relay work. Yup but now im trying to put the Box behind the nat. i refer to this thread http://lists.sip-router.org/pipermail/sr-users/2011-June/069223.html im able to put rtpproxy behind nat but need to modify the SDP body put in the proper ip address. so im trying to use force_rtp_proxy(co,external-address); to modify the (c=) and (o=) in SDP body. but in the SDP body i didnt see any changes of the ip address i tried adding several flags f,r,w but no luck. if i double my line force_rtp_proxy(co,external-address); in my SDP body i will get double private ip (c=192.168.1.3192.168.1.3) and (o=192.168.1.3192.168.1.3) thanks in advance, -- Regards, MingHon ___ 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] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi Ovidiu, Thanks for the info. But why the command still available in the doc. http://www.kamailio.org/docs/modules/stable/modules_k/rtpproxy.html#id3024166 anyway i will try the trunk version. but may i know which version? is this ok? # svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5kamailio Thanks in advance. Cheers, MingHon On Thu, Jul 7, 2011 at 6:39 AM, Ovidiu Sas o...@voipembedded.com wrote: Can you try the trunk version? I remember fixing a bug related to the external IP, but I don't remember if it is available on a stable version. Regards, Ovidiu sas On Wed, Jul 6, 2011 at 11:16 AM, MingHon gming...@gmail.com wrote: Hi Carsten, I tried RTPProxy using external address. It work out from the box without any problem. UA successfully registered and RTP-Traffic relay work. Yup but now im trying to put the Box behind the nat. i refer to this thread http://lists.sip-router.org/pipermail/sr-users/2011-June/069223.html im able to put rtpproxy behind nat but need to modify the SDP body put in the proper ip address. so im trying to use force_rtp_proxy(co,external-address); to modify the (c=) and (o=) in SDP body. but in the SDP body i didnt see any changes of the ip address i tried adding several flags f,r,w but no luck. if i double my line force_rtp_proxy(co,external-address); in my SDP body i will get double private ip (c=192.168.1.3192.168.1.3) and (o=192.168.1.3192.168.1.3) thanks in advance, -- Regards, MingHon ___ 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] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi, thanks.. may i know which version of kamailio? what the different between the stable tar ball and the git version? http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git is 3.1 okie? Cheers, MingHon ___ 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] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi List,
im facing an issue that my kamailio proxy did not replace the ip address in
the invite and 200OK sdp body.
my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user
my kamailio is listening on 192.168.1.3, also
define: advertised_address=175.136.223.112; advertised_port=5060;
and my asterisk is on 192.168.1.23.
sip signalling and rtp port forwarded to kamailio.
uacs from another nat register successfully.
if i put 2 lines of force_rtp_proxy(fcow,175.136.223.112);
i will get double ip addr in c and o but kamailio ignore my ip addr. example
i will get
c=IN IP4 192.168.1.3192.168.1.3
here is part of my simple script.
hope you can help.
thank you very much.
---cfg---
route[RTPPROXY] {
#!ifdef WITH_NAT
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy(fcow,175.136.223.112);
#force_rtp_proxy(fcow,175.136.223.112);
xlog(L_INFO,offer);
}
if (!has_totag()) add_rr_param(;nat=yes);
#!endif
return;
}
--
and here is the wireshark for uac INVITE and OK.
---INVITE-
ve0
EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0
Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080
Max-Forwards: 69
From: 101 sip:1...@aextddns.dyndns.info;tag=as032358a3
To: sip:102@192.168.1.3:5060
Contact: sip:102@192.168.1.23:5080
Call-ID: 416f6e09674ae9671bb7144a1cb11...@aextddns.dyndns.info
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Tue, 05 Jul 2011 07:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 327
v=0
o=root 1639709788 1639709788 IN IP4 192.168.1.3
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.3
t=0 0
m=audio 10072 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
---200OK---
e90
ElE;pX4tSIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.200:5062
;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416
Record-Route: sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes
From: 101 sip:1...@aextddns.dyndns.info;tag=1796959074
To: sip:1...@aextddns.dyndns.info;tag=as2e4c0125
Call-ID: 1985782590@192.168.2.200
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.23:5080
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 403900934 403900934 IN IP4 192.168.1.23
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.23
t=0 0
m=audio 14420 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
My kamailio log.
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3
INFO: script: offer
-
double force_rtp_proxy
kamailio - asterisk [INVITE]-
Pyi-}E7V@:#pINVITE sip:1...@aextddns.dyndns.info SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=640933430;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0
Via: SIP/2.0/UDP 192.168.2.200:5062
;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648
From: 101 sip:1...@aextddns.dyndns.info;tag=640933430
To: sip:1...@aextddns.dyndns.info
Call-ID: 1909950509@192.168.2.200
CSeq: 21 INVITE
Contact: sip:101@175.138.21.31:2788
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
User-Agent: T20 9.41.0.80
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 334
v=0
o=20073 20073 IN IP4 192.168.1.3192.168.1.3
s=SDP data
c=IN IP4 192.168.1.3192.168.1.3
t=0 0
m=audio 1006410064 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=nortpproxy:yes
a=nortpproxy:yes
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3
INFO: script: offer
---LOG--
--
Regards,
MingHon
___
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] two uac behind same nat one uac remain Cflag:: 0
Hi, Thanks for reply but i guess this is because of my sip alg issue. when my first uac register using src port 5062 and private ip but my sip alg change the private ip to public ip and and using the same src port 5062 so kamailio unable to detect the uac behind nat. im currently having issue in force_rtp_proxy. here is the link http://lists.sip-router.org/pipermail/sr-users/2011-July/069262.html i unable to replace the old ip address in c and o to a new ip address. please advice.. thanks.. -- Regards, MingHon ___ 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] force_rtp_proxy[flag, ipaddress] not rewriting c and o ?
Hi, how do i know is it invoked? i try putting xlog after force_rtp_proxy(rfco,publicip); line xlog(force_rtp); when i invite i saw force_rtp in /var/log/messages thanks. -- Regards, MingHon ___ 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] force_rtp_proxy[flag, ipaddress] not rewriting c and o ?
im trying to replace the ip address in c= and o= with force_rtp_proxy.
i tried different flag co,rc,ro,fo,fc,rfco but still doesnt work.
hope some one can help me with this..
below is my noob simple script.
hope someone can help..
thanks you :)
route[RTPPROXY] {
#!ifdef WITH_NAT
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
xlog(L_INFO,force_rtp_proxy $rm from $fu
(IP:$si:$sp)\n);
force_rtp_proxy(rfco,publicip);
}
if (!has_totag()) add_rr_param(;nat=yes);
#!endif
return;
}
--
Regards,
MingHon
___
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 + rtpproxy behind nat possible?
will kamailio as proxy behind nat and UACs behind another nat work? port forward sip and rtp done in the router. UACs register successfully but no audio. advertised_address = public_ip advertised_port = sip both define after the line of listen=public_ip please advice. thanks. -- Regards, MingHon ___ 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 + rtpproxy behind nat possible?
Hi, Im using kamailio 3.1.4 as proxy and rtpproxy 1.2.1 in a same server and asterisk 1.6 is at another server both server in the same lan. im new at kamailio and rtpproxy. do you have to source or example mind to share? i try googling but i cant find any. thank you very much. -- Regards, MingHon ___ 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 + rtpproxy behind nat possible?
Hi, Can you give me more detail, could you guide me how to? thanks you very much.. -- Regards, MingHon ___ 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] rtpproxy and kamailio doesnt work out from the box.
Hi, Thanks for the suggestion. i tried running rtpproxy on private. rtpproxy -l public_ip -s udp:192.168.2.3:7722 -u user and in kamailio cfg. modparam(rtpproxy, rtpproxy_sock, udp:192.168.2.3:7722) but still the same issue.. thanks. -- Regards, MingHon ___ 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] rtpproxy and kamailio doesnt work out from the box.
Hi List, i tcpdump on my kamailio server. when uac registered with source port 5060 the uac will remain Cflag:: 0 and did not have the Received: field. i also tried some iphone sip apps. some uac app will register with src port 5060 and some will not. anyone have solution for this? or how to prevent uac register with src port 5060 ? thanks! -- Regards, MingHon ___ 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] two uac behind same nat one uac remain Cflag:: 0
Hi List,
both uac register successfully but the issue is the first uac who register
the Cflag remain 0.
im using nat_uac_test(19).
thank you in advance.
here is the ul show:
AOR:: 102
Contact:: sip:102@175.136.221.60:5062 Q=
Expires:: 103
Callid:: 678936531@175.136.221.60
Cseq:: 2
User-agent:: T22 7.3.0.50
Received:: sip:175.136.221.60:1056
State:: CS_SYNC
Flags:: 0
Cflag:: 192
Socket:: udp:60.53.82.184:5060
Methods:: 16383
AOR:: 103
Contact:: sip:103@175.136.221.60:5062 Q=
Expires:: 62
Callid:: 1621958925@175.136.221.60
Cseq:: 3
User-agent:: Yealink SIP-T18 18.0.0.70
State:: CS_SYNC
Flags:: 0
Cflag:: 0
Socket:: udp:60.53.82.184:5060
Methods:: 16383
and here is the wireshark on kamailio for ua 103 and 102.
E@ \5Rw REGISTER sip:aextddns.dyndns.info SIP/2.0
Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK2080454766
From: 103 sip:103@175.136.221.60;tag=681558690
To: 103 sip:1...@aextddns.dyndns.info
Call-ID: 1621958925@175.136.221.60
CSeq: 2 REGISTER
Contact: sip:103@175.136.221.60:5062
Authorization: Digest username=103, realm=aextddns.dyndns.info,
nonce=Tgp7mU4Kem3Qqyd66hwhWgmU8bNpL463, uri=si$
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T18 18.0.0.70
Expires: 3600
Content-Length: 0
E@5RPSIP/2.0 200 OK
Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK2080454766;rport=5062
From: 103 sip:103@175.136.221.60;tag=681558690
To: 103 sip:1...@aextddns.dyndns.info
;tag=cbf8071f5b02e3e2fd329ee5d72c3f65.d24e
Call-ID: 1621958925@175.136.221.60
CSeq: 2 REGISTER
Contact: sip:103@175.136.221.60:5062;expires=120
Server: kamailio (3.1.4 (i386/linux))
Content-Length: 0
--
E@ j5R 7REGISTER sip:aextddns.dyndns.info SIP/2.0
Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK1080036976
From: 102 sip:102@175.136.221.60;tag=544575245
To: 102 sip:1...@aextddns.dyndns.info
Call-ID: 678936531@175.136.221.60
CSeq: 2 REGISTER
Contact: sip:102@175.136.221.60:5062
Authorization: Digest username=102, realm=aextddns.dyndns.info,
nonce=Tgp8AU4KetWGb79MEU5MUHq+xA4xy+OW, uri=si$
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: T22 7.3.0.50
Expires: 3600
Content-Length: 0
E@5R {SIP/2.0 200 OK
Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK1080036976;rport=1056
From: 102 sip:102@175.136.221.60;tag=544575245
To: 102 sip:1...@aextddns.dyndns.info
;tag=cbf8071f5b02e3e2fd329ee5d72c3f65.5cb8
Call-ID: 678936531@175.136.221.60
CSeq: 2 REGISTER
Contact: sip:102@175.136.221.60:5062;expires=120;received=sip:
175.136.221.60:1056
Server: kamailio (3.1.4 (i386/linux))
Content-Length: 0
--
Regards,
MingHon
___
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] rtpproxy and kamailio doesnt work out from the box.
Hi, i registered 3 uac behind same nat successfully but when i try to call each other i didnt get any audio. but if i use uac 102 and 103 to call into the voicemail i heard the audio but not for 101. kamailio is listening 60.48.218.61 and 192.168.2.3 rtpproxy is running. asterisk is at 192.168.2.23. here is my ul show. AOR:: 102 Contact:: sip:102@175.136.221.60:5062 Q= Expires:: 3110 Callid:: 721498432@175.136.221.60 Cseq:: 2 User-agent:: T22 7.3.0.50 Received:: sip:175.136.221.60:1024 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.48.218.61:5060 Methods:: 16383 AOR:: 103 Contact:: sip:103@175.136.221.60:5062 Q= Expires:: 3114 Callid:: 1499738216@175.136.221.60 Cseq:: 2 User-agent:: Yealink SIP-T18 18.0.0.70 Received:: sip:175.136.221.60:1025 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.48.218.61:5060 Methods:: 16383 AOR:: 101 Contact:: sip:101@175.136.221.60:5062 Q= Expires:: 3097 Callid:: 166053301@175.136.221.60 Cseq:: 2 User-agent:: T20 9.41.0.80 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:60.48.218.61:5060 Methods:: 16383 and may i know why uac 101 did not have the received: field? please some one could give a hand on this? the audio really cant get thru i really have no idea. thank you -- Regards, MingHon ___ 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] rtpproxy and kamailio doesnt work out from the box.
Hi, i fixed the audio issue for 102 to 103 vice versa. by fixing the canreinvite in asterisk. from uac the rtp packet will route to kamailio den forward to asterisk. can we bypass the rtp packet going to asterisk? and here is the update for uac 101 issue. when 101 call to voicemail or 102/103 there is no audio. in wireshark i saw 101 send rtp packet to a private ip belong to asterisk. but if 102/103 call to 101 both uac got audio. i realize this is because 101 is the first uac registered before 102/103 and because it did not have the received: field in ul show. please adv. -- Regards, MingHon ___ 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] rtpproxy and kamailio doesnt work out from the box.
hey list, i currently test on my asterisk 1.6 box on on pc with private ip and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc with public ip and private ip. everything installed successfully. i follow this tutorial realtime intergration with asterisk http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb my config. rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user #!define WITH_NAT modparam(rtpproxy, rtpproxy_sock, udp:127.0.0.1:7722) modparam(nathelper, sipping_from, sip:pinger@publicip) nat_uac_test(19) # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); is my config correct? may i know the best version work out from the box? please advice.. thanks in adv. :) -- Regards, MingHon ___ 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] rtpproxy and kamailio doesnt work out from the box.
Hi, Yup i did define #!define WITH_NAT at the beginning of config file. it doesnt work. been struggle for a month hope you can help. and also in the cfg im listening to both iface. listen=public ip listen=192.168.2.3 [kamailio ip] Regards, MingHon ___ 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] rtpproxy and kamailio doesnt work out from the box.
Hi, which firewall do you mean? the uac firewall or the kamailio firewall? kamailio doesnt have firewall. kamailio and rtpproxy in centos 5.4 firewall and selinux disabled. eth0 is configured for ppp0e and eth1 is the private ip and echo 1 /proc/sys/net/ipv4/ip_forward iptables -t nat -A POSTROUTING -o ppp0 -j MASQUERADE and for asterisk is in fedora14 firewall and selinux also disabled. for uac the firewall also disabled in the router. (dlink dir-615). what else do i need to check? pls adv.. -- Regards, MingHon ___ 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
