From rupa at rupa.com Sun Aug 1 00:18:09 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Sun, 1 Aug 2010 02:18:09 -0500 Subject: [Freeswitch-users] nibble_bill and last git In-Reply-To: <1343AFBD885444C59A21EF9852978BBA@MOBILEE1705> References: <1343AFBD885444C59A21EF9852978BBA@MOBILEE1705> Message-ID: Never mind, I misunderstood what you were asking about lcr_rate. Ok, custom sql. I'll re-review your other email on that. On Sun, Aug 1, 2010 at 12:44 AM, Madovsky wrote: > why a Jira for this ? it's not a bug.... > yes I made it working but without the custom > sql example on lcr page. > > > ----- Original Message ----- > *From:* Rupa Schomaker > *To:* FreeSWITCH Users Help > *Sent:* Sunday, August 01, 2010 1:10 AM > *Subject:* Re: [Freeswitch-users] nibble_bill and last git > > can you open a jira on this? > > Also, I guess this means you have it working? (nothing outstanding on > lcr/nibblebill) > > On Sat, Jul 31, 2010 at 11:44 PM, Madovsky wrote: > >> lcr difference between SVN and GIT version : >> the var lcr_rate_1 becomes lcr_rate_fieldl >> >> F >> >> ----- Original Message ----- >> *From:* Madovsky >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Saturday, July 31, 2010 11:25 PM >> *Subject:* Re: nibble_bill and last git >> >> my fault I forgot a var, also I need to change my lcr custom.sql >> >> myabe ti would be nice to make a coercition with ldc + nibble bill example >> and nibble bill table >> >> Thanks >> >> Franck >> >> ----- Original Message ----- >> *From:* Madovsky >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Saturday, July 31, 2010 6:49 PM >> *Subject:* nibble_bill and last git >> >> Nibble bill stopped to debit user accounts since I udpated last git. >> I didn't change anything in my settings, the console says well >> nibbleBill Beginning new billing bllbabla, >> >> but the cash field iss not debited, even with hearbeat set. >> >> Any ? >> >> Thanks >> >> Franck >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100801/da5836cc/attachment.html From infos at madovsky.org Sun Aug 1 01:05:44 2010 From: infos at madovsky.org (Madovsky) Date: Sun, 1 Aug 2010 04:05:44 -0400 Subject: [Freeswitch-users] nibble_bill and last git References: <1343AFBD885444C59A21EF9852978BBA@MOBILEE1705> Message-ID: <76AA465ADBD14309A69BFC55CC944E2C@MOBILEE1705> noprob... I go to sleep.... ----- Original Message ----- From: Rupa Schomaker To: FreeSWITCH Users Help Sent: Sunday, August 01, 2010 3:18 AM Subject: Re: [Freeswitch-users] nibble_bill and last git Never mind, I misunderstood what you were asking about lcr_rate. Ok, custom sql. I'll re-review your other email on that. On Sun, Aug 1, 2010 at 12:44 AM, Madovsky wrote: why a Jira for this ? it's not a bug.... yes I made it working but without the custom sql example on lcr page. ----- Original Message ----- From: Rupa Schomaker To: FreeSWITCH Users Help Sent: Sunday, August 01, 2010 1:10 AM Subject: Re: [Freeswitch-users] nibble_bill and last git can you open a jira on this? Also, I guess this means you have it working? (nothing outstanding on lcr/nibblebill) On Sat, Jul 31, 2010 at 11:44 PM, Madovsky wrote: lcr difference between SVN and GIT version : the var lcr_rate_1 becomes lcr_rate_fieldl F ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Saturday, July 31, 2010 11:25 PM Subject: Re: nibble_bill and last git my fault I forgot a var, also I need to change my lcr custom.sql myabe ti would be nice to make a coercition with ldc + nibble bill example and nibble bill table Thanks Franck ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Saturday, July 31, 2010 6:49 PM Subject: nibble_bill and last git Nibble bill stopped to debit user accounts since I udpated last git. I didn't change anything in my settings, the console says well nibbleBill Beginning new billing bllbabla, but the cash field iss not debited, even with hearbeat set. Any ? Thanks Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100801/d30d589b/attachment-0001.html From Nabble at slickdeals.endjunk.com Sun Aug 1 05:52:20 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Sun, 1 Aug 2010 05:52:20 -0700 (PDT) Subject: [Freeswitch-users] Asterisk equivalent Bridge() function implementation In-Reply-To: <1280449670950-5353341.post@n2.nabble.com> References: <1279302815443-5303089.post@n2.nabble.com> <1279369424513-5305762.post@n2.nabble.com> <1279931619913-5331714.post@n2.nabble.com> <1280241961671-5342450.post@n2.nabble.com> <1280449670950-5353341.post@n2.nabble.com> Message-ID: <1280667140871-5360856.post@n2.nabble.com> mercutioviz wrote: > If you know the uuid's of the two channels then you can do > whatever you want... Exactly. Unfortunately, I don't know how to find out the uuid of the two channels, especially through a dialplan context. If anyone knows and would like to shed some light on this, it sure will help. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Asterisk-equivalent-Bridge-function-implementation-tp5303089p5360856.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveu at coppice.org Sun Aug 1 06:09:42 2010 From: steveu at coppice.org (Steve Underwood) Date: Sun, 01 Aug 2010 21:09:42 +0800 Subject: [Freeswitch-users] T.38_gateway & switch_core_media_bug : increase "timeout" In-Reply-To: References: Message-ID: <4C557216.20002@coppice.org> On 07/30/2010 02:41 AM, Weera Suriya wrote: > Thanks Anthony, > It works! > > I did 2 basic tests with FS T.38 gateway and CallWeaver T.38 gateway. > > Test 1 (PSTN to T.38 ATA) > -------------------------------------- > 60 voice (concurrent) calls & 30 FAX (concurrent) calls > success rate; > for FreeSwitch : 97.3% with 10,000 faxes(single A4 text document) > for CallWeaver : 83.2% with 10,000 faxes(single A4 text document) > > > Test 2 (PSTN to T.38 ATA) > -------------------------------------- > 30 FAX (concurrent) calls > success rate; > for FreeSwitch : 97.8% with 10,000 faxes(single A4 text document) > for CallWeaver : 99.4% with 10,000 faxes(single A4 text document) > I'd like to find out why the numbers for Freeswitch are so poor. The better overall design of Freeswitch probably explains the more consistent results at the two load levels. However, if Callweaver can achieve 99.4% when lightly loaded, it would appear your ATA is capable of that level of success. Why, then, does Freeswitch only achieve 97.x%? Do you have any logs that give information about the failed calls? Steve From brokendash at gmail.com Sun Aug 1 09:55:47 2010 From: brokendash at gmail.com (broken dash) Date: Sun, 1 Aug 2010 11:55:47 -0500 Subject: [Freeswitch-users] Errors building the ESL libs... Message-ID: Im using a fresh git pull and have configured,made,installed and when trying to build the ESL stuffs it's spitting this out... any ideas? Cheers, B root at ivr-dev:/usr/src/freeswitch/libs/esl# make everymod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C perl make[1]: Entering directory `/usr/src/freeswitch/libs/esl/perl' make[1]: Nothing to be done for `all'. make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/perl' make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/src/freeswitch/libs/esl/php' g++ -I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o cc1plus: warnings being treated as errors esl_wrap.cpp:2583: error: deprecated conversion from string constant to ?char*? make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 From brokendash at gmail.com Sun Aug 1 09:55:47 2010 From: brokendash at gmail.com (broken dash) Date: Sun, 1 Aug 2010 11:55:47 -0500 Subject: [Freeswitch-users] Errors building the ESL libs... Message-ID: Im using a fresh git pull and have configured,made,installed and when trying to build the ESL stuffs it's spitting this out... any ideas? Cheers, B root at ivr-dev:/usr/src/freeswitch/libs/esl# make everymod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C perl make[1]: Entering directory `/usr/src/freeswitch/libs/esl/perl' make[1]: Nothing to be done for `all'. make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/perl' make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/src/freeswitch/libs/esl/php' g++ -I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o cc1plus: warnings being treated as errors esl_wrap.cpp:2583: error: deprecated conversion from string constant to ?char*? make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 From brokendash at gmail.com Sun Aug 1 09:55:47 2010 From: brokendash at gmail.com (broken dash) Date: Sun, 1 Aug 2010 11:55:47 -0500 Subject: [Freeswitch-users] Errors building the ESL libs... Message-ID: Im using a fresh git pull and have configured,made,installed and when trying to build the ESL stuffs it's spitting this out... any ideas? Cheers, B root at ivr-dev:/usr/src/freeswitch/libs/esl# make everymod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C perl make[1]: Entering directory `/usr/src/freeswitch/libs/esl/perl' make[1]: Nothing to be done for `all'. make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/perl' make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/src/freeswitch/libs/esl/php' g++ -I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o cc1plus: warnings being treated as errors esl_wrap.cpp:2583: error: deprecated conversion from string constant to ?char*? make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 From jim at k4gvo.com Sun Aug 1 12:11:42 2010 From: jim at k4gvo.com (Jim) Date: Sun, 01 Aug 2010 15:11:42 -0400 Subject: [Freeswitch-users] Module build problem with wanpipe Message-ID: <4C55C6EE.7010609@k4gvo.com> I'm installing the latest wanpipe drivers and it's not working. I'm wondering where he's getting the linux source name in this warning from? There is no directory in /usr/src/linux or /lib/moduels that matches it. In fact, locate doesn't match it either. There are no strings in the wanpipe source tree that match either. Where's it coming from? WARNING: Module installation dir mismatch! Linux source name = 2.6.31.12 Current image name = 2.6.31-21-generic From freeswitch-list at puzzled.xs4all.nl Sun Aug 1 13:42:44 2010 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sun, 01 Aug 2010 22:42:44 +0200 Subject: [Freeswitch-users] Errors building the ESL libs... In-Reply-To: References: Message-ID: <4C55DC44.2070805@puzzled.xs4all.nl> On 08/01/2010 06:55 PM, broken dash wrote: > Im using a fresh git pull and have configured,made,installed and when > trying to build the ESL stuffs it's spitting this out... any ideas? Posting to the list only once really is enough... Iirc the problem is caused by a typo in the php headers. Something with a missing "}" or "," in a php header file. Don't recall which one. File a php bug at your distro's bugzilla and hope they fix it. Please note that afaik this is not a FreeSWITCH problem. Regards, Patrick From moises.silva at gmail.com Sun Aug 1 13:47:45 2010 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 1 Aug 2010 16:47:45 -0400 Subject: [Freeswitch-users] Module build problem with wanpipe In-Reply-To: <4C55C6EE.7010609@k4gvo.com> References: <4C55C6EE.7010609@k4gvo.com> Message-ID: unless specified otherwise, the file at /lib/modules/`uname -r`/build/Makefile will be used to determine the kernel version. Inside that Makefile there is some variables at the top defining the version. If you need more help you need to tell us what distro you are using and the output of: "uname -a" "ls -la /lib/modules" "ls -la lib/modules/`uname -r`" "cat /lib/modules/`uname -r`/build/Makefile" Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com On Sun, Aug 1, 2010 at 3:11 PM, Jim wrote: > I'm installing the latest wanpipe drivers and it's not working. I'm > wondering where he's getting the linux source name in this warning > from? There is no directory in /usr/src/linux or /lib/moduels that > matches it. In fact, locate doesn't match it either. There are no > strings in the wanpipe source tree that match either. Where's it coming > from? > > WARNING: Module installation dir mismatch! > Linux source name = 2.6.31.12 > Current image name = 2.6.31-21-generic > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100801/42219457/attachment-0001.html From dome at tel.co.th Sun Aug 1 20:55:11 2010 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 2 Aug 2010 10:55:11 +0700 Subject: [Freeswitch-users] Proxy Media and Codec Prefer Message-ID: Dear All, I'm not sure it's bug or not ? I try to test codec negotate. with 3 FS box FS1 send PCMA to FS2 and FS2 send G729 to FS3 FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) My dialplan It's work fine When i try Call fail. because FS2 ignore absolute_codec_string='G729' and send only PCMU (I think by pass from FS1) It's bug or not ? Dome C. From steveayre at gmail.com Sun Aug 1 23:44:02 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 2 Aug 2010 07:44:02 +0100 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: Message-ID: In proxy media mode FS will pass media straight through without any processing. It won't do any transcoding or any SDP modifcation, so it can't transcode PCMU <-> G729 and absolute_codec_string will be ignored - it'll just pass the codec settings straight through from the A leg to the B leg. So it's expected behaviour. -Steve On 2 August 2010 04:55, Dome Charoenyost wrote: > Dear All, > I'm not sure it's bug or not ? > I try to test codec negotate. with 3 FS box > FS1 send PCMA to FS2 and FS2 send G729 to FS3 > > FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) > My dialplan > data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080"/> > It's work fine > When i try > > data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080"/> > Call fail. because FS2 ignore absolute_codec_string='G729' and send > only PCMU (I think by pass from FS1) > > It's bug or not ? > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/069f5304/attachment.html From mnhassan at usa.net Mon Aug 2 00:23:30 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 2 Aug 2010 13:23:30 +0600 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: Message-ID: Thank you for the explanation Steven. However, if we see that there is a common codec, say G729, and we want to make sure that the call gets connected on G729, is there any way to enforce this? Regards HASSAN On Mon, Aug 2, 2010 at 12:44, Steven Ayre wrote: > In proxy media mode FS will pass media straight through without any > processing. It won't do any transcoding or any SDP modifcation, so it can't > transcode PCMU <-> G729 and absolute_codec_string will be ignored - it'll > just pass the codec settings straight through from the A leg to the B leg. > > So it's expected behaviour. > > -Steve > > > > > On 2 August 2010 04:55, Dome Charoenyost wrote: > >> Dear All, >> I'm not sure it's bug or not ? >> I try to test codec negotate. with 3 FS box >> FS1 send PCMA to FS2 and FS2 send G729 to FS3 >> >> FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) >> My dialplan >> > data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080 >> "/> >> It's work fine >> When i try >> >> > data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080 >> "/> >> Call fail. because FS2 ignore absolute_codec_string='G729' and send >> only PCMU (I think by pass from FS1) >> >> It's bug or not ? >> >> Dome C. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/32af2bb2/attachment.html From steveayre at gmail.com Mon Aug 2 01:01:49 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 2 Aug 2010 09:01:49 +0100 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: Message-ID: Yes, don't use proxy media. Use absolute_codec_string as you already were to enforce it. There is no way if you're using proxy media. -Steve On 2 August 2010 08:23, Nyamul Hassan wrote: > Thank you for the explanation Steven. However, if we see that there is a > common codec, say G729, and we want to make sure that the call gets > connected on G729, is there any way to enforce this? > > Regards > HASSAN > > > > On Mon, Aug 2, 2010 at 12:44, Steven Ayre wrote: > >> In proxy media mode FS will pass media straight through without any >> processing. It won't do any transcoding or any SDP modifcation, so it can't >> transcode PCMU <-> G729 and absolute_codec_string will be ignored - it'll >> just pass the codec settings straight through from the A leg to the B leg. >> >> So it's expected behaviour. >> >> -Steve >> >> >> >> >> On 2 August 2010 04:55, Dome Charoenyost wrote: >> >>> Dear All, >>> I'm not sure it's bug or not ? >>> I try to test codec negotate. with 3 FS box >>> FS1 send PCMA to FS2 and FS2 send G729 to FS3 >>> >>> FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) >>> My dialplan >>> >> data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080 >>> "/> >>> It's work fine >>> When i try >>> >>> >> data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080 >>> "/> >>> Call fail. because FS2 ignore absolute_codec_string='G729' and send >>> only PCMU (I think by pass from FS1) >>> >>> It's bug or not ? >>> >>> Dome C. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/969558bd/attachment.html From mnhassan at usa.net Mon Aug 2 01:40:03 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 2 Aug 2010 14:40:03 +0600 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: Message-ID: Interesting. In proxy media, FS is in the middle of the media transfer, and one would think that such was possible in such situations. In contrast, when you are not acting as a media proxy, there is little on what media the end points can negotiate among themselves. There must be some logical reasoning that I fail to understand here. Regards HASSAN On Mon, Aug 2, 2010 at 14:01, Steven Ayre wrote: > Yes, don't use proxy media. Use absolute_codec_string as you already were > to enforce it. > > There is no way if you're using proxy media. > > -Steve > > > > On 2 August 2010 08:23, Nyamul Hassan wrote: > >> Thank you for the explanation Steven. However, if we see that there is a >> common codec, say G729, and we want to make sure that the call gets >> connected on G729, is there any way to enforce this? >> >> Regards >> HASSAN >> >> >> >> On Mon, Aug 2, 2010 at 12:44, Steven Ayre wrote: >> >>> In proxy media mode FS will pass media straight through without any >>> processing. It won't do any transcoding or any SDP modifcation, so it can't >>> transcode PCMU <-> G729 and absolute_codec_string will be ignored - it'll >>> just pass the codec settings straight through from the A leg to the B leg. >>> >>> So it's expected behaviour. >>> >>> -Steve >>> >>> >>> >>> >>> On 2 August 2010 04:55, Dome Charoenyost wrote: >>> >>>> Dear All, >>>> I'm not sure it's bug or not ? >>>> I try to test codec negotate. with 3 FS box >>>> FS1 send PCMA to FS2 and FS2 send G729 to FS3 >>>> >>>> FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) >>>> My dialplan >>>> >>> data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080 >>>> "/> >>>> It's work fine >>>> When i try >>>> >>>> >>> data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080 >>>> "/> >>>> Call fail. because FS2 ignore absolute_codec_string='G729' and send >>>> only PCMU (I think by pass from FS1) >>>> >>>> It's bug or not ? >>>> >>>> Dome C. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/473f3e19/attachment-0001.html From dome at tel.co.th Mon Aug 2 01:47:18 2010 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 2 Aug 2010 15:47:18 +0700 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: Message-ID: Use when leg A and Leg B use same codec but need to proxy Right ? Dome C. 2010/8/2 Steven Ayre : > Yes, don't use proxy media. Use absolute_codec_string as you already were to > enforce it. > > There is no way if you're using proxy media. > > -Steve > > > On 2 August 2010 08:23, Nyamul Hassan wrote: >> >> Thank you for the explanation Steven. ?However, if we see that there is a >> common codec, say G729, and we want to make sure that the call gets >> connected on G729, is there any way to enforce this? >> Regards >> HASSAN >> >> >> On Mon, Aug 2, 2010 at 12:44, Steven Ayre wrote: >>> >>> In proxy media mode FS will pass media straight through without any >>> processing. It won't do any transcoding or any SDP modifcation, so it can't >>> transcode PCMU <-> G729 and absolute_codec_string will be ignored - it'll >>> just pass the codec settings straight through from the A leg to the B leg. >>> >>> So it's expected behaviour. >>> >>> -Steve >>> >>> >>> >>> On 2 August 2010 04:55, Dome Charoenyost wrote: >>>> >>>> Dear All, >>>> ? ? ? ?I'm not sure it's bug or not ? >>>> I try to test codec ?negotate. with 3 FS box >>>> FS1 send PCMA to FS2 and FS2 send G729 to FS3 >>>> >>>> FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) >>>> My dialplan >>>> >>> >>>> data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080"/> >>>> It's work fine >>>> When i try >>>> >>>> >>> >>>> data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:5080"/> >>>> Call fail. because FS2 ignore absolute_codec_string='G729' and send >>>> only ?PCMU ?(I think by pass from FS1) >>>> >>>> It's bug or not ? >>>> >>>> Dome C. >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sos at sokhapkin.dyndns.org Mon Aug 2 03:25:37 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 2 Aug 2010 06:25:37 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: Message-ID: <201008020625.37601.sos@sokhapkin.dyndns.org> FS has 3 ways (not 2!) to handle media: 1. bypass_media=true - media path is set to run audio directly between leg a and b endpoints. absolute_codec_string variable is ignored in this mode. 2. proxy_media=true - media is proxied by FS, but without any transcoding etc, frame in - frame out. absolute_codec_string variable is ignored in this mode. 3. Neither bypass_media, no proxy_media variables are set. FS does full media handling/transcoding, absolute_codec_string variable is honored. On Monday 02 August 2010, Nyamul Hassan wrote: > Interesting. In proxy media, FS is in the middle of the media transfer, > and one would think that such was possible in such situations. > > In contrast, when you are not acting as a media proxy, there is little on > what media the end points can negotiate among themselves. > > There must be some logical reasoning that I fail to understand here. > > Regards > HASSAN > > On Mon, Aug 2, 2010 at 14:01, Steven Ayre wrote: > > Yes, don't use proxy media. Use absolute_codec_string as you already were > > to enforce it. > > > > There is no way if you're using proxy media. > > > > -Steve > > > > On 2 August 2010 08:23, Nyamul Hassan wrote: > >> Thank you for the explanation Steven. However, if we see that there is > >> a common codec, say G729, and we want to make sure that the call gets > >> connected on G729, is there any way to enforce this? > >> > >> Regards > >> HASSAN > >> > >> On Mon, Aug 2, 2010 at 12:44, Steven Ayre wrote: > >>> In proxy media mode FS will pass media straight through without any > >>> processing. It won't do any transcoding or any SDP modifcation, so it > >>> can't transcode PCMU <-> G729 and absolute_codec_string will be ignored > >>> - it'll just pass the codec settings straight through from the A leg to > >>> the B leg. > >>> > >>> So it's expected behaviour. > >>> > >>> -Steve > >>> > >>> On 2 August 2010 04:55, Dome Charoenyost wrote: > >>>> Dear All, > >>>> I'm not sure it's bug or not ? > >>>> I try to test codec negotate. with 3 FS box > >>>> FS1 send PCMA to FS2 and FS2 send G729 to FS3 > >>>> > >>>> FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) > >>>> My dialplan > >>>> >>>> data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:508 > >>>>0 "/> > >>>> It's work fine > >>>> When i try > >>>> > >>>> >>>> data="{absolute_codec_string='G729'}sofia/internal/2222 at 10.20.2.49:508 > >>>>0 "/> > >>>> Call fail. because FS2 ignore absolute_codec_string='G729' and send > >>>> only PCMU (I think by pass from FS1) > >>>> > >>>> It's bug or not ? > >>>> > >>>> Dome C. > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > >>>s http://www.freeswitch.org > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From juanito1982 at gmail.com Mon Aug 2 03:55:31 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Mon, 2 Aug 2010 12:55:31 +0200 Subject: [Freeswitch-users] Different versions, different cpu load Message-ID: Hello! I was been some test using one FS tarball version downloaded some weeks ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems with odbc connections managemens. Now I'm using a git version (FreeSWITCH Version 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve this problem but I noticed it has a high cpu load comparing with svn version. While I could manage more than 200 calls with a 75% CPU load into a dual core server using svn version, now, 50 calls consume this 75% cpu. I can see same modules are loaded (except new hash module needed in git version) and same scenario is used. I am not be able to find why now it uses more cpu than before. Any idea? Each testing call is a simple bridge to an external sip provider. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/b68fbc3c/attachment.html From steveayre at gmail.com Mon Aug 2 04:45:12 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 2 Aug 2010 12:45:12 +0100 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: Message-ID: When neitther bypass media nor proxy media is set, FS still is in the media path. The difference between 'normal' and 'proxy' is that FS doesn't do anything with the media, just passes it straight through. You seem to be thinking that bypass media is the default - it isn't. -Steve On 2 August 2010 09:40, Nyamul Hassan wrote: > Interesting. In proxy media, FS is in the middle of the media transfer, > and one would think that such was possible in such situations. > > In contrast, when you are not acting as a media proxy, there is little on > what media the end points can negotiate among themselves. > > There must be some logical reasoning that I fail to understand here. > > Regards > HASSAN > > > > On Mon, Aug 2, 2010 at 14:01, Steven Ayre wrote: > >> Yes, don't use proxy media. Use absolute_codec_string as you already were >> to enforce it. >> >> There is no way if you're using proxy media. >> >> -Steve >> >> >> >> On 2 August 2010 08:23, Nyamul Hassan wrote: >> >>> Thank you for the explanation Steven. However, if we see that there is a >>> common codec, say G729, and we want to make sure that the call gets >>> connected on G729, is there any way to enforce this? >>> >>> Regards >>> HASSAN >>> >>> >>> >>> On Mon, Aug 2, 2010 at 12:44, Steven Ayre wrote: >>> >>>> In proxy media mode FS will pass media straight through without any >>>> processing. It won't do any transcoding or any SDP modifcation, so it can't >>>> transcode PCMU <-> G729 and absolute_codec_string will be ignored - it'll >>>> just pass the codec settings straight through from the A leg to the B leg. >>>> >>>> So it's expected behaviour. >>>> >>>> -Steve >>>> >>>> >>>> >>>> >>>> On 2 August 2010 04:55, Dome Charoenyost wrote: >>>> >>>>> Dear All, >>>>> I'm not sure it's bug or not ? >>>>> I try to test codec negotate. with 3 FS box >>>>> FS1 send PCMA to FS2 and FS2 send G729 to FS3 >>>>> >>>>> FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) >>>>> My dialplan >>>>> >>>> data="{absolute_codec_string='G729'}sofia/internal/ >>>>> 2222 at 10.20.2.49:5080"/> >>>>> It's work fine >>>>> When i try >>>>> >>>>> >>>> data="{absolute_codec_string='G729'}sofia/internal/ >>>>> 2222 at 10.20.2.49:5080"/> >>>>> Call fail. because FS2 ignore absolute_codec_string='G729' and send >>>>> only PCMU (I think by pass from FS1) >>>>> >>>>> It's bug or not ? >>>>> >>>>> Dome C. >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/7913129e/attachment.html From bottleman at icf.org.ru Mon Aug 2 05:36:38 2010 From: bottleman at icf.org.ru (Georgiewskiy Yuriy) Date: Mon, 2 Aug 2010 16:36:38 +0400 (MSD) Subject: [Freeswitch-users] Mod_h323 AliasAddress Error In-Reply-To: References: Message-ID: On 2010-07-31 13:27 +0200, Saeed Ahmed wrote freeswitch-users at lists.freeswi...: SA>Hi, SA> SA>I moved forward, module is successfully loaded but, SA> SA>when i send call from my xlite and try to bridge it with another FS using SA>h323 (also running with mod_h323) and 2nd FS plays the MOH. but that doesn't SA>work SA> SA>i also tried to bridge it from 2nd FS to a H323 gate keeper but no success. SA> SA>when i make a first call then there is alot of deug output and then it SA>doesn't take next calls and continue to produce some debug infos. give debug info please, and tell what versions of ptlib and h323plus you are use. SA>maybe something is wrong with my h323.conf SA> SA>here it is: SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA> SA>PS: i also tried to register with Eikga, it got register but doesn't SA>originate the calls, do you guys know any other h323 softphone? mod_h323 cannot register endpoints himself. SA> SA>many thanks. SA>2010/7/7 Saeed Ahmed SA> SA>> Thanks Georgiewskiy, SA>> * SA>> * SA>> *I'll test it and will also update it on Wiki. SA>> * SA>> 2010/7/7 Georgiewskiy Yuriy SA>> SA>>> On 2010-07-07 16:02 +0200, Saeed Ahmed wrote SA>>> freeswitch-users at lists.freeswi...: SA>>> SA>>> SA>Dear All, SA>>> SA> SA>>> SA>I am trying to configure mod_h323 with freeswitch, while loading module SA>>> i SA>>> SA>receive that error: SA>>> SA> SA>>> SA> SA>>> SA>2010-07-07 16:01:02.027642 [CONSOLE] switch_loadable_module.c:944 SA>>> SA>Successfully Loaded [mod_h26x] SA>>> SA>2010-07-07 16:01:02.304187 [CONSOLE] mod_h323.cpp:74 Starting loading SA>>> SA>mod_h323 SA>>> SA>Assertion fail: Must have non-empty string in AliasAddress!, file SA>>> SA>h323ep.cxx, line 3586 SA>>> SA> SA>>> SA>bort, ore dump? SA>>> SA>>> wiki is outdated, add ?? SA>>> h323.conf.xml SA>>> SA>>> SA> SA>>> SA>I am following that wiki : SA>>> SA> SA>>> SA>http://wiki.freeswitch.org/wiki/Mod_h323 SA>>> SA> SA>>> SA>my h323.conf.xml is: SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA> SA>>> SA>I would appreciate any help. SA>>> SA> SA>>> SA>Thanks SA>>> SA> SA>>> SA>>> C ????????? With Best Regards SA>>> ???????????? ????. Georgiewskiy Yuriy SA>>> +7 4872 711666 +7 4872 711666 SA>>> ???? +7 4872 711143 fax +7 4872 711143 SA>>> ???????? ??? "?? ?? ??????" IT Service Ltd SA>>> http://nkoort.ru http://nkoort.ru SA>>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru SA>>> YG129-RIPE YG129-RIPE SA>>> _______________________________________________ SA>>> FreeSWITCH-users mailing list SA>>> FreeSWITCH-users at lists.freeswitch.org SA>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users SA>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users SA>>> http://www.freeswitch.org SA>>> SA>>> SA>> SA> C ????????? With Best Regards ???????????? ????. Georgiewskiy Yuriy +7 4872 711666 +7 4872 711666 ???? +7 4872 711143 fax +7 4872 711143 ???????? ??? "?? ?? ??????" IT Service Ltd http://nkoort.ru http://nkoort.ru JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE From chaitanya at vivainfomedia.com Mon Aug 2 05:48:20 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Mon, 2 Aug 2010 18:18:20 +0530 Subject: [Freeswitch-users] getDigits() over ESL Message-ID: Hey I want to get DTMF digits over ESL. I got function "uuid_recv_dtmf" in freeswitch api document, but i am not getting how to retrieve data from this function. Can someone guide me about API command or function of get DTMF? Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/46592823/attachment.html From daniel.neubert at solomo.de Mon Aug 2 06:17:13 2010 From: daniel.neubert at solomo.de (Daniel Neubert) Date: Mon, 2 Aug 2010 15:17:13 +0200 Subject: [Freeswitch-users] getDigits() over ESL In-Reply-To: References: Message-ID: <4C56C559.8020801@solomo.de> Hi, did you execute start_dtmf in dialplan? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Best regards / Mit freundlichen Gr??en, Daniel Neubert On 02.08.2010 14:48, Chaitanya Bhatt // Viva wrote: > Hey > > I want to get DTMF digits over ESL. I got function "uuid_recv_dtmf" in > freeswitch api document, but i am not getting how to retrieve data > from this function. > Can someone guide me about API command or function of get DTMF? > > Incase of any further queries, Please feel free to mail me or contact > me on the numbers provided below. > > Thanks & Regards, > Chaitanya Bhatt > Software Engineer. > > Viva Infomedia Pvt. Ltd. > 242, Oshiwara Industrial Centre, > New Link Road, Opp. Oshiwara Bus Depot, > Goregaon West, Mumbai 400104. > > Direct: +91.22.40310356 > Board: +91.22.40310310 > Email : chaitanya at vivainfomedia.com > > Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging > India Awards 2009 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/06514514/attachment.html From chaitanya at vivainfomedia.com Mon Aug 2 06:46:11 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Mon, 2 Aug 2010 19:16:11 +0530 Subject: [Freeswitch-users] getDigits() over ESL In-Reply-To: <4C56C559.8020801@solomo.de> References: <4C56C559.8020801@solomo.de> Message-ID: Hey Daniel, Thanks for your response. I don't want to specify commands in dialplan, i want to give commands to Freeswitch from my Perl script over ESL. I want to get DTMF input. I tried "READ" command but that needs sound file, but i want plain DTMF input without playing any sound file. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 On Mon, Aug 2, 2010 at 6:47 PM, Daniel Neubert wrote: > Hi, > > did you execute start_dtmf in dialplan? > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > > Best regards / Mit freundlichen Gr??en, > Daniel Neubert > > On 02.08.2010 14:48, Chaitanya Bhatt // Viva wrote: > > Hey > > I want to get DTMF digits over ESL. I got function "uuid_recv_dtmf" in > freeswitch api document, but i am not getting how to retrieve data from this > function. > Can someone guide me about API command or function of get DTMF? > > Incase of any further queries, Please feel free to mail me or contact me on > the numbers provided below. > > Thanks & Regards, > Chaitanya Bhatt > Software Engineer. > > Viva Infomedia Pvt. Ltd. > 242, Oshiwara Industrial Centre, > New Link Road, Opp. Oshiwara Bus Depot, > Goregaon West, Mumbai 400104. > > Direct: +91.22.40310356 > Board: +91.22.40310310 > Email : chaitanya at vivainfomedia.com > > Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India > Awards 2009 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/c6fe08da/attachment.html From daniel.neubert at solomo.de Mon Aug 2 07:04:30 2010 From: daniel.neubert at solomo.de (Daniel Neubert) Date: Mon, 2 Aug 2010 16:04:30 +0200 Subject: [Freeswitch-users] getDigits() over ESL In-Reply-To: References: <4C56C559.8020801@solomo.de> Message-ID: <4C56D06E.7010707@solomo.de> Hi, but in order to receive DTMF-Events you need to issue start_dtmf first: You can use start_dtmf in a dialplan to enable in-band DTMF detection (i.e. the detection of DTMF tones on a channel). You should do this when you want to be able to identify DTMF tones on a channel that doesn't otherwise support another signaling method (like RFC2833 or INFO). After start_dtmf is executed FreeSWITCH will detect DTMF an generate events for them so you can read the DTMF-Digit via ESL. Best regards / Mit freundlichen Gr??en, Daniel Neubert On 02.08.2010 15:46, Chaitanya Bhatt // Viva wrote: > Hey Daniel, > > Thanks for your response. > > I don't want to specify commands in dialplan, i want to give commands > to Freeswitch from my Perl script over ESL. > I want to get DTMF input. > I tried "READ" command but that needs sound file, but i want plain > DTMF input without playing any sound file. > > Thanks & Regards, > Chaitanya Bhatt > Software Engineer. > > Viva Infomedia Pvt. Ltd. > 242, Oshiwara Industrial Centre, > New Link Road, Opp. Oshiwara Bus Depot, > Goregaon West, Mumbai 400104. > > Direct: +91.22.40310356 > Board: +91.22.40310310 > Email : chaitanya at vivainfomedia.com > > Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging > India Awards 2009 > > > > On Mon, Aug 2, 2010 at 6:47 PM, Daniel Neubert > > wrote: > > Hi, > > did you execute start_dtmf in dialplan? > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > > Best regards / Mit freundlichen Gr??en, > Daniel Neubert > > > On 02.08.2010 14:48, Chaitanya Bhatt // Viva wrote: >> Hey >> >> I want to get DTMF digits over ESL. I got function >> "uuid_recv_dtmf" in freeswitch api document, but i am not getting >> how to retrieve data from this function. >> Can someone guide me about API command or function of get DTMF? >> >> Incase of any further queries, Please feel free to mail me or >> contact me on the numbers provided below. >> >> Thanks & Regards, >> Chaitanya Bhatt >> Software Engineer. >> >> Viva Infomedia Pvt. Ltd. >> 242, Oshiwara Industrial Centre, >> New Link Road, Opp. Oshiwara Bus Depot, >> Goregaon West, Mumbai 400104. >> >> Direct: +91.22.40310356 >> Board: +91.22.40310310 >> Email : chaitanya at vivainfomedia.com >> >> >> Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging >> India Awards 2009 >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/91dfb813/attachment.html From chaitanya at vivainfomedia.com Mon Aug 2 07:15:46 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Mon, 2 Aug 2010 19:45:46 +0530 Subject: [Freeswitch-users] getDigits() over ESL In-Reply-To: <4C56D06E.7010707@solomo.de> References: <4C56C559.8020801@solomo.de> <4C56D06E.7010707@solomo.de> Message-ID: Dear Daniel, I know this method to get DTMF digits from headers of events. But is there any direct API call to get digits besides "Read" command which requires to play sound file ? Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 On Mon, Aug 2, 2010 at 7:34 PM, Daniel Neubert wrote: > Hi, > > but in order to receive DTMF-Events you need to issue start_dtmf first: > > You can use start_dtmf in a dialplan to enable in-band DTMF detection (i.e. > the detection of DTMF tones on a channel). You should do this when you want > to be able to identify DTMF tones on a channel that doesn't otherwise > support another signaling method (like RFC2833 or INFO). > > After start_dtmf is executed FreeSWITCH will detect DTMF an generate events > for them so you can read the DTMF-Digit via ESL. > > Best regards / Mit freundlichen Gr??en, > Daniel Neubert > > > On 02.08.2010 15:46, Chaitanya Bhatt // Viva wrote: > > Hey Daniel, > > Thanks for your response. > > I don't want to specify commands in dialplan, i want to give commands to > Freeswitch from my Perl script over ESL. > I want to get DTMF input. > I tried "READ" command but that needs sound file, but i want plain DTMF > input without playing any sound file. > > Thanks & Regards, > Chaitanya Bhatt > Software Engineer. > > Viva Infomedia Pvt. Ltd. > 242, Oshiwara Industrial Centre, > New Link Road, Opp. Oshiwara Bus Depot, > Goregaon West, Mumbai 400104. > > Direct: +91.22.40310356 > Board: +91.22.40310310 > Email : chaitanya at vivainfomedia.com > > Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India > Awards 2009 > > > > On Mon, Aug 2, 2010 at 6:47 PM, Daniel Neubert wrote: > >> Hi, >> >> did you execute start_dtmf in dialplan? >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf >> >> Best regards / Mit freundlichen Gr??en, >> Daniel Neubert >> >> >> On 02.08.2010 14:48, Chaitanya Bhatt // Viva wrote: >> >> Hey >> >> I want to get DTMF digits over ESL. I got function "uuid_recv_dtmf" in >> freeswitch api document, but i am not getting how to retrieve data from this >> function. >> Can someone guide me about API command or function of get DTMF? >> >> Incase of any further queries, Please feel free to mail me or contact me >> on the numbers provided below. >> >> Thanks & Regards, >> Chaitanya Bhatt >> Software Engineer. >> >> Viva Infomedia Pvt. Ltd. >> 242, Oshiwara Industrial Centre, >> New Link Road, Opp. Oshiwara Bus Depot, >> Goregaon West, Mumbai 400104. >> >> Direct: +91.22.40310356 >> Board: +91.22.40310310 >> Email : chaitanya at vivainfomedia.com >> >> Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India >> Awards 2009 >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/48368c58/attachment-0001.html From null at invalid.name Mon Aug 2 08:10:41 2010 From: null at invalid.name (Dan Lane) Date: Mon, 2 Aug 2010 16:10:41 +0100 Subject: [Freeswitch-users] Distributing SIP registrations using ODBC Message-ID: Hi Guys, I have a cluster of FS boxes all sharing SIP registrations using ODBC and under certain circumstances I can register Client-A on FS-A and Client-B on FS-B and make calls between Client-A and Client-B without issue as the relevant FS box knows how to get a call to the client from the shared registrations table. However when the callee client is behind NAT (or using a SIP client configured to only allow calls from a server it has registered with) the call gets dropped by the NAT router (or client) I can fix this easily with some LUA but before I do I wanted to check if there was an advisable or recognised FreeSWITCH way to work around this. Has anyone else solved this issue already? Regards, Dan From jim at k4gvo.com Mon Aug 2 08:42:05 2010 From: jim at k4gvo.com (Jim) Date: Mon, 02 Aug 2010 11:42:05 -0400 Subject: [Freeswitch-users] Module build problem with wanpipe In-Reply-To: References: <4C55C6EE.7010609@k4gvo.com> Message-ID: <4C56E74D.1010604@k4gvo.com> On 08/01/2010 04:47 PM, Moises Silva wrote: > "uname -a" > "ls -la /lib/modules" > "ls -la lib/modules/`uname -r`" > "cat /lib/modules/`uname -r`/build/Makefile" > Hi, Moises, Thanks for the response. I tried numerous things yesterday and nothing seemed to work. I rebooted this morning and for some reason things are back to working better. I still have problems but I think they are unrelated to the new drivers. I'll have to get back to you. At least the A200 is now being configured by freeswitch. Thanks, Jim. From msc at freeswitch.org Mon Aug 2 08:47:30 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Aug 2010 08:47:30 -0700 Subject: [Freeswitch-users] getDigits() over ESL In-Reply-To: References: <4C56C559.8020801@solomo.de> <4C56D06E.7010707@solomo.de> Message-ID: On Mon, Aug 2, 2010 at 7:15 AM, Chaitanya Bhatt // Viva < chaitanya at vivainfomedia.com> wrote: > Dear Daniel, > > I know this method to get DTMF digits from headers of events. But is there > any direct API call to get digits besides "Read" command which requires to > play sound file ? > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits You can play silence if you wish. The app will collect digits during playback unlike read which plays the file prior to playback. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/f05a261a/attachment.html From steveayre at gmail.com Mon Aug 2 09:10:04 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 2 Aug 2010 17:10:04 +0100 Subject: [Freeswitch-users] Distributing SIP registrations using ODBC In-Reply-To: References: Message-ID: I haven't tried this myself, but routing the call via the FS server they registered too might work since it'll be the IP the NAT router is expecting traffic from. -Steve On 2 August 2010 16:10, Dan Lane wrote: > Hi Guys, > > I have a cluster of FS boxes all sharing SIP registrations using ODBC > and under certain circumstances I can register Client-A on FS-A and > Client-B on FS-B and make calls between Client-A and Client-B without > issue as the relevant FS box knows how to get a call to the client > from the shared registrations table. > > However when the callee client is behind NAT (or using a SIP client > configured to only allow calls from a server it has registered with) > the call gets dropped by the NAT router (or client) > > I can fix this easily with some LUA but before I do I wanted to check > if there was an advisable or recognised FreeSWITCH way to work around > this. > ro > Has anyone else solved this issue already? > > Regards, > Dan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/127235b5/attachment.html From null at invalid.name Mon Aug 2 09:41:17 2010 From: null at invalid.name (Dan Lane) Date: Mon, 2 Aug 2010 17:41:17 +0100 Subject: [Freeswitch-users] Distributing SIP registrations using ODBC In-Reply-To: References: Message-ID: Yup, that was how I plan to fix it using LUA to figure out where to route the call (possibly proxying it via the other FS server). Just wanted to check I wasn't re-inventing the wheel before I go ahead and do it as it seems like the sort of issue other people would have dealt with. On Mon, Aug 2, 2010 at 5:10 PM, Steven Ayre wrote: > I haven't tried this myself, but routing the call via the FS server they > registered too might work since it'll be the IP the NAT router is expecting > traffic from. > > -Steve > > > On 2 August 2010 16:10, Dan Lane wrote: >> >> Hi Guys, >> >> I have a cluster of FS boxes all sharing SIP registrations using ODBC >> and under certain circumstances I can register Client-A on FS-A and >> Client-B on FS-B and make calls between Client-A and Client-B without >> issue as the relevant FS box knows how to get a call to the client >> from the shared registrations table. >> >> However when the callee client is behind NAT (or using a SIP client >> configured to only allow calls from a server it has registered with) >> the call gets dropped by the NAT router (or client) >> >> I can fix this easily with some LUA but before I do I wanted to check >> if there was an advisable or recognised FreeSWITCH way to work around >> this. >> ro >> Has anyone else solved this issue already? >> >> Regards, >> Dan >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dswardstrom at remotelink.com Mon Aug 2 09:54:07 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Mon, 2 Aug 2010 11:54:07 -0500 (CDT) Subject: [Freeswitch-users] Executing a JavaScript application during startup In-Reply-To: <645705269.31.1280765773016.JavaMail.root@srvr12.remotelinkml.com> Message-ID: <1758996237.33.1280768047063.JavaMail.root@srvr12.remotelinkml.com> Is there a way to cause a JavaScript script to be executed during FreeSWITCH initialization? Actually, what I want to do is to add a table to the SQLite DB and populate it with some data that I will use later in a JavaScript application. I could do this some other ways, but the ability to initialize some system wide data during startup using JavaScript would be useful. I do know some techniques that I could use which I don't really want to use: * The JavaScript application that I am using could try to determine if this is the first time, and do any initialization. Sounds reasonable, but could encounter a race condition with a 2nd call. * I do plan to have an Erlang program running which uses mod_erlang_event. I could use it, but there is a possibility that a call could come in before the Erlang program starts or some other race condition. I could write a special module to do this, but wonder if there is a way to do this. Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom From dswardstrom at remotelink.com Mon Aug 2 12:31:48 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Mon, 2 Aug 2010 12:31:48 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch Crashed by ODBC32.dll In-Reply-To: <1280500144711-5355214.post@n2.nabble.com> References: <1280500144711-5355214.post@n2.nabble.com> Message-ID: <1280777508148-5365527.post@n2.nabble.com> More information on the problem with ODBC from Barrry: Note: This is using a Postgresql driver, but there is a chance that similar problematic code could be in other ODBC drivers. > Here's a description of the issue we had. > > Recently, while load testing the FreeSWITCH system we discovered a crash > at an approximate load of 560 simultaneous calls. This problem turned > out to be difficult to find because the stack was being corrupted so we > could not trace the crash reliably. > > Eventually, we got lucky and trapped a crash with gdb active and a > mildly corrupted stack. The problem turned out to be an issue with a > tcp/ip select() call. The default fdset variable used by the select() > call only allowed 1024 file descriptors to be used by the entire > system. Since FreeSWITCH uses two file descriptors (SIP) per call we > exceeded that number quite quickly. Technically, we were corrupting > the stack at the 500 call level, but the stack wasn't getting corrupted > 'enough' to crash until we approached 560 simultaneous calls. > > The fdset variable/select() call that was causing the issue was in the > Postgresql v8.4 0200 Driver code. We contacted the Postgresql > developers list and almost immediately received a patch from them. The > patch converted the select() call to a Linux poll call. Problem > solved. We were able to test the system to 2500 simultaneous calls. > > Barry Nicholson > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Crashed-by-ODBC32-dll-tp5354417p5365527.html Sent from the freeswitch-users mailing list archive at Nabble.com. From drcubi at gmail.com Mon Aug 2 13:11:32 2010 From: drcubi at gmail.com (D Cubi) Date: Mon, 2 Aug 2010 13:11:32 -0700 Subject: [Freeswitch-users] new install on freebsd 8.1 Message-ID: Followed these instructions. All seemed to have installed properly, but can't find bin folder : # pwd /usr/local/freeswitch # ls -l total 8 drwxr-xr-x 11 root wheel 512 Aug 2 11:24 conf drwxr-xr-x 2 root wheel 512 Aug 2 11:24 htdocs drwxr-xr-x 2 root wheel 512 Aug 2 12:44 lib drwxr-xr-x 4 root wheel 512 Aug 2 12:12 sounds Does anyone have any idea where I went wrong? Thnx Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/5a8d49f3/attachment.html From drcubi at gmail.com Mon Aug 2 13:30:27 2010 From: drcubi at gmail.com (D Cubi) Date: Mon, 2 Aug 2010 13:30:27 -0700 Subject: [Freeswitch-users] new install on freebsd 8.1 Message-ID: Forgot to add URL of instructions followed in original posting to list :( Followed these instructions. http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD All seemed to have installed properly, but can't find bin folder : # pwd /usr/local/freeswitch # ls -l total 8 drwxr-xr-x 11 root wheel 512 Aug 2 11:24 conf drwxr-xr-x 2 root wheel 512 Aug 2 11:24 htdocs drwxr-xr-x 2 root wheel 512 Aug 2 12:44 lib drwxr-xr-x 4 root wheel 512 Aug 2 12:12 sounds Does anyone have any idea where I went wrong? Thnx Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100802/26a6f438/attachment-0001.html From juanito1982 at gmail.com Tue Aug 3 03:41:12 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Tue, 3 Aug 2010 12:41:12 +0200 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: No one experiment this same issue? I also noticed a high cpu consume on call hangup. Regards 2010/8/2 Juan Antonio Iba?ez Santorum > Hello! > > I was been some test using one FS tarball version downloaded some weeks ago > (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems with odbc > connections managemens. Now I'm using a git version (FreeSWITCH Version > 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve this > problem but I noticed it has a high cpu load comparing with svn version. > While I could manage more than 200 calls with a 75% CPU load into a dual > core server using svn version, now, 50 calls consume this 75% cpu. I can see > same modules are loaded (except new hash module needed in git version) and > same scenario is used. I am not be able to find why now it uses more cpu > than before. Any idea? > Each testing call is a simple bridge to an external sip provider. > > Regards > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/fd92f95c/attachment.html From saeedahmad1981 at gmail.com Tue Aug 3 04:45:12 2010 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Tue, 3 Aug 2010 13:45:12 +0200 Subject: [Freeswitch-users] Mod_h323 AliasAddress Error In-Reply-To: References: Message-ID: Hi, When i changed the value to "false" then it started working. Even its working but i still see following error: 2010-08-02 16:08:42.088627 [ERR] mod_h323.cpp:1981 h323/8112# 982183863666 at 67.18.148.250 Unsupported ptime of 6 on write Audio codec G.729A/B{sw} for connection [0x2aaab80c2dc0] Thanks. 2010/8/2 Georgiewskiy Yuriy > On 2010-07-31 13:27 +0200, Saeed Ahmed wrote > freeswitch-users at lists.freeswi...: > > SA>Hi, > SA> > SA>I moved forward, module is successfully loaded but, > SA> > SA>when i send call from my xlite and try to bridge it with another FS > using > SA>h323 (also running with mod_h323) and 2nd FS plays the MOH. but that > doesn't > SA>work > SA> > SA>i also tried to bridge it from 2nd FS to a H323 gate keeper but no > success. > SA> > SA>when i make a first call then there is alot of deug output and then it > SA>doesn't take next calls and continue to produce some debug infos. > > give debug info please, and tell what versions of ptlib and h323plus you > are use. > > > SA>maybe something is wrong with my h323.conf > SA> > SA>here it is: > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA> > SA>PS: i also tried to register with Eikga, it got register but doesn't > SA>originate the calls, do you guys know any other h323 softphone? > > mod_h323 cannot register endpoints himself. > > SA> > SA>many thanks. > SA>2010/7/7 Saeed Ahmed > SA> > SA>> Thanks Georgiewskiy, > SA>> * > SA>> * > SA>> *I'll test it and will also update it on Wiki. > SA>> * > SA>> 2010/7/7 Georgiewskiy Yuriy > SA>> > SA>>> On 2010-07-07 16:02 +0200, Saeed Ahmed wrote > SA>>> freeswitch-users at lists.freeswi...: > SA>>> > SA>>> SA>Dear All, > SA>>> SA> > SA>>> SA>I am trying to configure mod_h323 with freeswitch, while loading > module > SA>>> i > SA>>> SA>receive that error: > SA>>> SA> > SA>>> SA> > SA>>> SA>2010-07-07 16:01:02.027642 [CONSOLE] switch_loadable_module.c:944 > SA>>> SA>Successfully Loaded [mod_h26x] > SA>>> SA>2010-07-07 16:01:02.304187 [CONSOLE] mod_h323.cpp:74 Starting > loading > SA>>> SA>mod_h323 > SA>>> SA>Assertion fail: Must have non-empty string in AliasAddress!, file > SA>>> SA>h323ep.cxx, line 3586 > SA>>> SA> > SA>>> SA>bort, ore dump? > SA>>> > SA>>> wiki is outdated, add ?? > SA>>> h323.conf.xml > SA>>> > SA>>> SA> > SA>>> SA>I am following that wiki : > SA>>> SA> > SA>>> SA>http://wiki.freeswitch.org/wiki/Mod_h323 > SA>>> SA> > SA>>> SA>my h323.conf.xml is: > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA> > SA>>> SA>I would appreciate any help. > SA>>> SA> > SA>>> SA>Thanks > SA>>> SA> > SA>>> > SA>>> C ????????? With Best Regards > SA>>> ???????????? ????. Georgiewskiy Yuriy > SA>>> +7 4872 711666 +7 4872 711666 > SA>>> ???? +7 4872 711143 fax +7 4872 711143 > SA>>> ???????? ??? "?? ?? ??????" IT Service Ltd > SA>>> http://nkoort.ru http://nkoort.ru > SA>>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru > SA>>> YG129-RIPE YG129-RIPE > SA>>> _______________________________________________ > SA>>> FreeSWITCH-users mailing list > SA>>> FreeSWITCH-users at lists.freeswitch.org > SA>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > SA>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > SA>>> http://www.freeswitch.org > SA>>> > SA>>> > SA>> > SA> > > C ????????? With Best Regards > ???????????? ????. Georgiewskiy Yuriy > +7 4872 711666 +7 4872 711666 > ???? +7 4872 711143 fax +7 4872 711143 > ???????? ??? "?? ?? ??????" IT Service Ltd > http://nkoort.ru http://nkoort.ru > JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru > YG129-RIPE YG129-RIPE > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/36c90072/attachment.html From bottleman at icf.org.ru Tue Aug 3 04:59:49 2010 From: bottleman at icf.org.ru (Georgiewskiy Yuriy) Date: Tue, 3 Aug 2010 15:59:49 +0400 (MSD) Subject: [Freeswitch-users] Mod_h323 AliasAddress Error In-Reply-To: References: Message-ID: On 2010-08-03 13:45 +0200, Saeed Ahmed wrote FreeSWITCH Users Help: SA>Hi, SA> SA>When i changed the value to "false" then it started working. SA> SA> SA> SA> SA>Even its working but i still see following error: SA> SA> SA>2010-08-02 16:08:42.088627 [ERR] mod_h323.cpp:1981 h323/8112# SA>982183863666 at 67.18.148.250 Unsupported ptime of 6 on write Audio codec SA>G.729A/B{sw} for connection [0x2aaab80c2dc0] it's not related to mod_h323, it's codec specific. SA>Thanks. SA> SA>2010/8/2 Georgiewskiy Yuriy SA> SA>> On 2010-07-31 13:27 +0200, Saeed Ahmed wrote SA>> freeswitch-users at lists.freeswi...: SA>> SA>> SA>Hi, SA>> SA> SA>> SA>I moved forward, module is successfully loaded but, SA>> SA> SA>> SA>when i send call from my xlite and try to bridge it with another FS SA>> using SA>> SA>h323 (also running with mod_h323) and 2nd FS plays the MOH. but that SA>> doesn't SA>> SA>work SA>> SA> SA>> SA>i also tried to bridge it from 2nd FS to a H323 gate keeper but no SA>> success. SA>> SA> SA>> SA>when i make a first call then there is alot of deug output and then it SA>> SA>doesn't take next calls and continue to produce some debug infos. SA>> SA>> give debug info please, and tell what versions of ptlib and h323plus you SA>> are use. SA>> SA>> SA>> SA>maybe something is wrong with my h323.conf SA>> SA> SA>> SA>here it is: SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA> SA>> SA>PS: i also tried to register with Eikga, it got register but doesn't SA>> SA>originate the calls, do you guys know any other h323 softphone? SA>> SA>> mod_h323 cannot register endpoints himself. SA>> SA>> SA> SA>> SA>many thanks. SA>> SA>2010/7/7 Saeed Ahmed SA>> SA> SA>> SA>> Thanks Georgiewskiy, SA>> SA>> * SA>> SA>> * SA>> SA>> *I'll test it and will also update it on Wiki. SA>> SA>> * SA>> SA>> 2010/7/7 Georgiewskiy Yuriy SA>> SA>> SA>> SA>>> On 2010-07-07 16:02 +0200, Saeed Ahmed wrote SA>> SA>>> freeswitch-users at lists.freeswi...: SA>> SA>>> SA>> SA>>> SA>Dear All, SA>> SA>>> SA> SA>> SA>>> SA>I am trying to configure mod_h323 with freeswitch, while loading SA>> module SA>> SA>>> i SA>> SA>>> SA>receive that error: SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA>2010-07-07 16:01:02.027642 [CONSOLE] switch_loadable_module.c:944 SA>> SA>>> SA>Successfully Loaded [mod_h26x] SA>> SA>>> SA>2010-07-07 16:01:02.304187 [CONSOLE] mod_h323.cpp:74 Starting SA>> loading SA>> SA>>> SA>mod_h323 SA>> SA>>> SA>Assertion fail: Must have non-empty string in AliasAddress!, file SA>> SA>>> SA>h323ep.cxx, line 3586 SA>> SA>>> SA> SA>> SA>>> SA>bort, ore dump? SA>> SA>>> SA>> SA>>> wiki is outdated, add ?? SA>> SA>>> h323.conf.xml SA>> SA>>> SA>> SA>>> SA> SA>> SA>>> SA>I am following that wiki : SA>> SA>>> SA> SA>> SA>>> SA>http://wiki.freeswitch.org/wiki/Mod_h323 SA>> SA>>> SA> SA>> SA>>> SA>my h323.conf.xml is: SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA> SA>> SA>>> SA>I would appreciate any help. SA>> SA>>> SA> SA>> SA>>> SA>Thanks SA>> SA>>> SA> SA>> SA>>> SA>> SA>>> C ????????? With Best Regards SA>> SA>>> ???????????? ????. Georgiewskiy Yuriy SA>> SA>>> +7 4872 711666 +7 4872 711666 SA>> SA>>> ???? +7 4872 711143 fax +7 4872 711143 SA>> SA>>> ???????? ??? "?? ?? ??????" IT Service Ltd SA>> SA>>> http://nkoort.ru http://nkoort.ru SA>> SA>>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru SA>> SA>>> YG129-RIPE YG129-RIPE SA>> SA>>> _______________________________________________ SA>> SA>>> FreeSWITCH-users mailing list SA>> SA>>> FreeSWITCH-users at lists.freeswitch.org SA>> SA>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users SA>> SA>>> UNSUBSCRIBE: SA>> http://lists.freeswitch.org/mailman/options/freeswitch-users SA>> SA>>> http://www.freeswitch.org SA>> SA>>> SA>> SA>>> SA>> SA>> SA>> SA> SA>> SA>> C ????????? With Best Regards SA>> ???????????? ????. Georgiewskiy Yuriy SA>> +7 4872 711666 +7 4872 711666 SA>> ???? +7 4872 711143 fax +7 4872 711143 SA>> ???????? ??? "?? ?? ??????" IT Service Ltd SA>> http://nkoort.ru http://nkoort.ru SA>> JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru SA>> YG129-RIPE YG129-RIPE SA>> SA>> _______________________________________________ SA>> FreeSWITCH-users mailing list SA>> FreeSWITCH-users at lists.freeswitch.org SA>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users SA>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users SA>> http://www.freeswitch.org SA>> SA>> SA> C ????????? With Best Regards ???????????? ????. Georgiewskiy Yuriy +7 4872 711666 +7 4872 711666 ???? +7 4872 711143 fax +7 4872 711143 ???????? ??? "?? ?? ??????" IT Service Ltd http://nkoort.ru http://nkoort.ru JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE From telteclistas at gmail.com Tue Aug 3 05:46:57 2010 From: telteclistas at gmail.com (leonardo alves) Date: Tue, 3 Aug 2010 08:46:57 -0400 Subject: [Freeswitch-users] Executing a JavaScript application during startup In-Reply-To: <1758996237.33.1280768047063.JavaMail.root@srvr12.remotelinkml.com> References: <645705269.31.1280765773016.JavaMail.root@srvr12.remotelinkml.com> <1758996237.33.1280768047063.JavaMail.root@srvr12.remotelinkml.com> Message-ID: I know that with perl and lua is possible to specy a script to execute in the freeswitch startup. But with javascript I dont know if there is this option also. To do with lua. Open the lua.conf in the autoload configs of freeswitch and there is this option there. Leo Is there a way to cause a JavaScript script to be executed > during FreeSWITCH initialization? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/0579c328/attachment.html From jprsa at yahoo.com Mon Aug 2 20:39:07 2010 From: jprsa at yahoo.com (jprsa) Date: Mon, 2 Aug 2010 20:39:07 -0700 (PDT) Subject: [Freeswitch-users] Setting up a calling station in Addis Ababa Message-ID: <29331391.post@talk.nabble.com> Hello, I am travelling to Addis Ababa in a few weeks and i would like to know if anyone has used freeswitch services from ethiopia. I set up my freeswitch servers in the US and in HK and I would like some advice re: connecting to my servers using softphones or ATAs while I am in Addis Ababa. I am not sure if the ISPs in Addis Ababa will let my packets through. Thanks jprsa -- View this message in context: http://old.nabble.com/Setting-up-a-calling-station-in-Addis-Ababa-tp29331391p29331391.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jprsa at yahoo.com Mon Aug 2 20:39:37 2010 From: jprsa at yahoo.com (jprsa) Date: Mon, 2 Aug 2010 20:39:37 -0700 (PDT) Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database Message-ID: <29331375.post@talk.nabble.com> Hello Sky1975, Were you able to solve your problem re: nibblebill not cutting off the call when the credits are empty? Hi Ram, I can't seem to access the site you pointed out http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/ would you happen to know how I can get to the said pre-paid samples? Thanks jprsa sky1975 wrote: > > Dear Ram, > > Thank you for the reply. To work with your code I hope that Mod cdr > should be there. But wiki says that its not functional. What should I > do. > > Thanks > > Senaka > > > > On Thu, Dec 17, 2009 at 7:29 PM, ram wrote: >> Hi >> >> Look at Contrib of source >> >> http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/ >> >> some pre-paid examples >> >> Ram >> >> On Wed, Dec 16, 2009 at 12:27 AM, Senaka Amarakeerthi >> >> wrote: >>> >>> Dear Sir, >>> >>> I have successfully installed freeSWITCH and it works fine in >>> passthrough >>> mode. I installed nibblebill and it deduct money from the accounts >>> database >>> and it works fine. but I have two problems. >>> >>> 1. Calls can be initiated even though there is a minus value in accounts >>> database >>> >>> 2. Calls doesn't hangup when it goes to minus values. >>> >>> Any answers are greatly appreciated. >>> >>> This is my dialplan: >>> >>> >>> >>> >>> ? >>> ? ? >>> ? ? >>> ? >>> >>> ? >>> >>> >>> >>> >>> >>> >> data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> >>> >>> >>> >>> >>> >>> This is the configuration file; >>> >>> >>> ? >>> ? ? >>> >>> ? ? >>> >>> >>> >>> >>> ? ? >>> >>> >>> ? ? >>> >>> >>> ? ? >>> >>> >>> >>> ? ? >>> >>> >>> ? ? >>> >>> >>> >>> ? ? >>> >>> >>> >>> ? ? >>> >>> >>> >>> ? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp26807854p29331375.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Mon Aug 2 22:59:45 2010 From: jason at jasonjgw.net (Jason White) Date: Tue, 3 Aug 2010 15:59:45 +1000 Subject: [Freeswitch-users] Solved: problem making outbound calls to service provider Message-ID: <20100803055945.GA9383@jdc.jasonjgw.net> First, apologies are due for not continuing the original thread; I was interacting with the mailing list via gmane.org and the recent messages, including mine, don't appear to have propagated to the newsgroup. I solved my difficulty with making outbound calls to my ISP's SIP service. The problem turned out to be the MTU setting of my ppp0 interface: reducing it to 1496 bytes allowed the FreeSWITCH invite packet (with the md5 authentication) to be fragmented and received by the remote side. From dftoro at yahoo.com Tue Aug 3 07:02:56 2010 From: dftoro at yahoo.com (Diego Toro) Date: Tue, 3 Aug 2010 07:02:56 -0700 (PDT) Subject: [Freeswitch-users] mod_unimrcp on Windows Message-ID: <451653.59657.qm@web33508.mail.mud.yahoo.com> Greetings I'm interested in using mod_unimrcp to integrate Loquendo TTS/ASR (LSS - Loquendo's mrcp server) on Windows, my question is, mod_unimrcp is ready for Windows ? I receive suggestions Thank you Diego Toro http://lacarretade.blogspot.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/ef2a0561/attachment.html From peter.olsson at visionutveckling.se Tue Aug 3 07:14:33 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 3 Aug 2010 16:14:33 +0200 Subject: [Freeswitch-users] mod_unimrcp on Windows In-Reply-To: <451653.59657.qm@web33508.mail.mud.yahoo.com> References: <451653.59657.qm@web33508.mail.mud.yahoo.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC078373@cooper> Well, it does compile :) It should work ok, but I haven't tried it yet. It's on my TODO-list yet... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Diego Toro Skickat: den 3 augusti 2010 16:03 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] mod_unimrcp on Windows Greetings I'm interested in using mod_unimrcp to integrate Loquendo TTS/ASR (LSS - Loquendo's mrcp server) on Windows, my question is, mod_unimrcp is ready for Windows ? I receive suggestions Thank you Diego Toro http://lacarretade.blogspot.com/ !DSPAM:4c58234d32932002426335! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/84c8ecbc/attachment.html From chaitanya at vivainfomedia.com Tue Aug 3 12:06:44 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Tue, 3 Aug 2010 12:06:44 -0700 Subject: [Freeswitch-users] getDigits() over ESL In-Reply-To: References: <4C56C559.8020801@solomo.de> <4C56D06E.7010707@solomo.de> Message-ID: Thanks Michael for your reply. I will play silence while taking DTMF. Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 On Mon, Aug 2, 2010 at 8:47 AM, Michael Collins wrote: > > > On Mon, Aug 2, 2010 at 7:15 AM, Chaitanya Bhatt // Viva < > chaitanya at vivainfomedia.com> wrote: > >> Dear Daniel, >> >> I know this method to get DTMF digits from headers of events. But is there >> any direct API call to get digits besides "Read" command which requires to >> play sound file ? >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits > > You can play silence if you wish. The app will collect digits during > playback unlike read which plays the file prior to playback. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/b3b12575/attachment-0001.html From infos at madovsky.org Tue Aug 3 14:17:10 2010 From: infos at madovsky.org (Madovsky) Date: Tue, 3 Aug 2010 17:17:10 -0400 Subject: [Freeswitch-users] ODBC postgresql network and FS Message-ID: Hi Clue guys, hope you enjoy at Cluecon ! Is there any setings in FS to restart it in case of idle DB connection ? exemple : if the DB on node1 fails and node2 starts DB in failover with the same shared IP, FS continues to believe that node1 is yet the right node to contact.... so all SIP communicatio with DB are stalled maybe a restart or other would be nice or anything else to unblock the status. Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/5b6a91e6/attachment.html From infos at madovsky.org Tue Aug 3 15:50:56 2010 From: infos at madovsky.org (Madovsky) Date: Tue, 3 Aug 2010 18:50:56 -0400 Subject: [Freeswitch-users] answer machine Message-ID: <17996754D275472B859921D9824E5233@MOBILEE1705> Is there a way to set up the volume of answer machine ? Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/05372530/attachment.html From infos at madovsky.org Tue Aug 3 15:53:49 2010 From: infos at madovsky.org (Madovsky) Date: Tue, 3 Aug 2010 18:53:49 -0400 Subject: [Freeswitch-users] answer machine Message-ID: <6986F8B220B943F6A7BB6ED578B84440@MOBILEE1705> found this thread http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg17343.html but is it relevant yet ? Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, August 03, 2010 6:50 PM Subject: answer machine Is there a way to set up the volume of answer machine ? Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100803/06f98e2b/attachment.html From dujinfang at gmail.com Wed Aug 4 00:19:58 2010 From: dujinfang at gmail.com (Seven Du) Date: Wed, 4 Aug 2010 15:19:58 +0800 Subject: [Freeswitch-users] sips dial string question Message-ID: Hi, I'm trying tls between 2 freeswitches, one as a server and one as a softphone. I think I successfully configured the server side, and copied the pem files to my client side. server: port 53 for sip and 54 for sips. client: external profile port 5080 for sip and 5081 for sips. I can successfully call the server from client: bgapi originate sofia/external/sips:xxxx at my-server.com:54 &echo questions: on client side, how to setup 1) register a gateway to server with sips. I tried realm=my-server.com:54 but doesn't seemed to work, is there any other param I should set ? I can register with sip(port 53). 2) then, can I use dial string like sofia/gateway/gw/xxxx with sips ? Thanks. -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From brian at microcomaustralia.com.au Wed Aug 4 04:35:55 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Wed, 4 Aug 2010 21:35:55 +1000 Subject: [Freeswitch-users] database corruption errors Message-ID: When starting freeswitch I get a long list of errors, starting from: 2010-08-04 21:24:01.608710 [INFO] switch_core_sqldb.c:1275 Opening DB 2010-08-04 21:24:01.611895 [ERR] switch_core_sqldb.c:404 SQL ERR [database disk image is malformed] drop table channels 2010-08-04 21:24:01.615011 [ERR] switch_core_sqldb.c:404 SQL ERR [database disk image is malformed] drop table calls 2010-08-04 21:24:01.617641 [ERR] switch_core_sqldb.c:404 SQL ERR [database disk image is malformed] drop table interfaces 2010-08-04 21:24:01.620330 [ERR] switch_core_sqldb.c:404 SQL ERR [database disk image is malformed] drop table tasks 2010-08-04 21:24:01.623370 [ERR] switch_core_sqldb.c:404 SQL ERR [database disk image is malformed] PRAGMA synchronous=OFF; 2010-08-04 21:24:01.626318 [ERR] switch_core_sqldb.c:404 SQL ERR [database disk image is malformed] PRAGMA cache_size=8000 2010-08-04 21:24:01.628830 [DEBUG] switch_core_sqldb.c:765 SQL ERR [database disk image is malformed] [select hostname from complete] Auto Generating Table! 2010-08-04 21:24:01.632166 [DEBUG] switch_core_sqldb.c:772 SQL ERR [database disk image is malformed] [CREATE TABLE complete ( sticky INTEGER, a1 VARCHAR(128), a2 VARCHAR(128), a3 VARCHAR(128), a4 VARCHAR(128), a5 VARCHAR(128), a6 VARCHAR(128), a7 VARCHAR(128), a8 VARCHAR(128), a9 VARCHAR(128), a10 VARCHAR(128), hostname VARCHAR(256) ); ] 2010-08-04 21:24:01.643766 [DEBUG] switch_core_sqldb.c:778 SQL ERR [database disk image is malformed] [CREATE TABLE complete ( sticky INTEGER, a1 VARCHAR(128), a2 VARCHAR(128), a3 VARCHAR(128), a4 VARCHAR(128), a5 VARCHAR(128), a6 VARCHAR(128), a7 VARCHAR(128), a8 VARCHAR(128), a9 VARCHAR(128), a10 VARCHAR(128), hostname VARCHAR(256) ); ] 2010-08-04 21:24:01.653300 [DEBUG] switch_core_sqldb.c:765 SQL ERR [database disk image is malformed] [select hostname from aliases] Auto Generating Table! [...] Which leads to two concerns: 1. Why is it trying to drop everything when the database is corrupt? (note: this is an older version now) 2. Which database is corrupt? Is there any command line tool I can use to verify it is corrupt? Or repair it even? Also not sure how it got to be corrupt in the first place. Might be something to do with a power failure. Freeswitch seemed OK, but I guess I should have checked the logs files at the time, the errors started about the same time. An automatic way of testing these files would be good, so I can ensure I am alerted if something goes wrong again. What is the best way to recover? -- Brian May From brian at microcomaustralia.com.au Wed Aug 4 04:42:05 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Wed, 4 Aug 2010 21:42:05 +1000 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: On 4 August 2010 21:35, Brian May wrote: > What is the best way to recover? For now, to solve the immediate problem of "my phone doesn't work" I renamed my db directory to db.old, and let it recreate the files from scratch. Not sure, what if anything I have lost as a result - I guess the main concern would be the voicemail greeting messages - will they be ok, are are they referenced from the db files? Also still curious if there is a better way to recover, and/or find out which files in the db directory were corrupt, other then trial and error. -- Brian May From steveayre at gmail.com Wed Aug 4 05:28:16 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 4 Aug 2010 13:28:16 +0100 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: Well, you probably only have one file - core.db You shouldn't have lost anything important in there - it mostly just stored information on current calls, sip registrations that kind of thing. As for how to do sqlite data recovery, try googling. e.g. http://www.mail-archive.com/sqlite-users at sqlite.org/msg17538.html -Steve On 4 August 2010 12:42, Brian May wrote: > On 4 August 2010 21:35, Brian May wrote: > > What is the best way to recover? > > For now, to solve the immediate problem of "my phone doesn't work" I > renamed my db directory to db.old, and let it recreate the files from > scratch. > > Not sure, what if anything I have lost as a result - I guess the main > concern would be the voicemail greeting messages - will they be ok, > are are they referenced from the db files? > > Also still curious if there is a better way to recover, and/or find > out which files in the db directory were corrupt, other then trial and > error. > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/f4130ca5/attachment.html From brian at freeswitch.org Wed Aug 4 07:07:31 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Aug 2010 09:07:31 -0500 Subject: [Freeswitch-users] sips dial string question In-Reply-To: References: Message-ID: <61EEF09E-8A60-49C7-B253-97A696464267@freeswitch.org> do this sofia/blah at blah;transport=tls /b On Aug 4, 2010, at 2:19 AM, Seven Du wrote: > Hi, > > I'm trying tls between 2 freeswitches, one as a server and one as a softphone. > > I think I successfully configured the server side, and copied the pem > files to my client side. > > > server: port 53 for sip and 54 for sips. > client: external profile port 5080 for sip and 5081 for sips. > > I can successfully call the server from client: > > bgapi originate sofia/external/sips:xxxx at my-server.com:54 &echo > > questions: > > on client side, how to setup > > 1) register a gateway to server with sips. I tried > realm=my-server.com:54 but doesn't seemed to work, is there any other > param I should set ? > I can register with sip(port 53). > > 2) then, can I use dial string like sofia/gateway/gw/xxxx with sips ? > > Thanks. > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vkozak at abisoft.spb.ru Wed Aug 4 07:46:33 2010 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Wed, 4 Aug 2010 18:46:33 +0400 Subject: [Freeswitch-users] Event queue is full Message-ID: Hi everybody. Can you tell me, please, what is the reason and how to avoid the error "Event queue is full!" FS log: 2010-08-03 14:16:48.213324 [DEBUG] sofia.c:4139 Channel sofia/external-subnet-10/sipp at 10.122.246.14:5061 entering state [completed][200] 2010-08-03 14:16:48.213324 [DEBUG] switch_core_session.c:638 Send signal sofia/external-subnet-10/sipp at 10.122.246.14:5061 [BREAK] 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! 2010-08-03 14:16:48.213324 [NOTICE] mod_dptools.c:717 Channel [sofia/external-subnet-10/sipp at 10.122.246.14:5061] has been answered 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! EXECUTE sofia/external-subnet-10/sipp at 10.122.246.14:5061 event(Event-App-Type=BP-STARTED,SYSTEM-ID=starpound,variable_sip_from_uri=sipp at 10.122.246.14:5061,variable_sip_to_uri=6783257895 at 10.122.246.112:5260,variable_sip_req_user=6783257895) 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! 2010-08-03 14:16:48.213324 [WARNING] switch_event.c:1169 queued event at a lower priority 2/0! EXECUTE sofia/external-subnet-10/sipp at 10.122.246.14:5061 park() 2010-08-03 14:16:48.213324 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.213324 [DEBUG] sofia.c:4139 Channel sofia/external-subnet-10/sipp at 10.122.246.14:5061 entering state [ready][200] 2010-08-03 14:16:48.313194 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.414213 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.514290 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.614241 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.714309 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.774320 [DEBUG] sofia.c:4139 Channel sofia/external-subnet-10/sipp at 10.122.246.14:5061 entering state [terminated][408] 2010-08-03 14:16:48.774320 [NOTICE] sofia.c:4775 Hangup sofia/external-subnet-10/sipp at 10.122.246.14:5061 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] 2010-08-03 14:16:48.774320 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.784288 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.814294 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.874289 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.884292 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.924295 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.987259 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:48.987259 [CRIT] switch_event.c:1175 Event queue is full! 2010-08-03 14:16:49.024356 [CRIT] switch_event.c:1175 Event queue is full! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/385bcf55/attachment-0001.html From kris at kriskinc.com Wed Aug 4 08:24:35 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 4 Aug 2010 11:24:35 -0400 Subject: [Freeswitch-users] sips dial string question Message-ID: Does freeswitch support the sips uri scheme? -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Wed Aug 04 10:07:31 2010 Subject: Re: [Freeswitch-users] sips dial string question do this sofia/blah at blah;transport=tls /b On Aug 4, 2010, at 2:19 AM, Seven Du wrote: > Hi, > > I'm trying tls between 2 freeswitches, one as a server and one as a > softphone. > > I think I successfully configured the server side, and copied the pem > files to my client side. > > > server: port 53 for sip and 54 for sips. > client: external profile port 5080 for sip and 5081 for sips. > > I can successfully call the server from client: > > bgapi originate sofia/external/sips:xxxx at my-server.com:54 &echo > > questions: > > on client side, how to setup > > 1) register a gateway to server with sips. I tried > realm=my-server.com:54 but doesn't seemed to work, is there any other > param I should set ? > I can register with sip(port 53). > > 2) then, can I use dial string like sofia/gateway/gw/xxxx with sips ? > > Thanks. > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Aug 4 08:29:46 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Aug 2010 10:29:46 -0500 Subject: [Freeswitch-users] Event queue is full In-Reply-To: References: Message-ID: <1C73EC5E-7E78-49E8-8171-7FFA7FDE94A1@freeswitch.org> I'm going to guess you're attached to event socket and you're not reading the events in. /b On Aug 4, 2010, at 9:46 AM, Kozak Vladimir wrote: > Can you tell me, please, what is the reason and how to avoid the error "Event queue is full!" > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/4904b089/attachment.html From gmaruzz at celliax.org Wed Aug 4 09:18:36 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 4 Aug 2010 18:18:36 +0200 Subject: [Freeswitch-users] Event queue is full In-Reply-To: References: Message-ID: 2010/8/4 Kozak Vladimir : > Hi everybody. > > > Can you tell me, please, what is the reason and how to avoid the error > "Event queue is full!" > If you're using an external program to connect to the event system, then you have to consume (read) the events, or they will choke the queue. Maybe you just connected via ESL using some programming language, but you've not yet implemented the reading loop? -giovanni From neil.burgess at redmatter.com Wed Aug 4 10:28:29 2010 From: neil.burgess at redmatter.com (Neil Burgess) Date: Wed, 4 Aug 2010 18:28:29 +0100 Subject: [Freeswitch-users] Custom NOTIFY via a proxy Message-ID: <787302A89ACCE24DA8F56DA101E77C842B38B429D0@THHS2E12BE1X.hostedservice2.net> Hi, I am trying to send CUSTOM NOTIFY events to devices, using the ESL interface with a PHP script of the following form. I am having partial success, but coming up against problems when using with an outbound proxy:- $e = new ESLevent("NOTIFY"); $e->addHeader("profile", "internal"); $e->addHeader("from", "1020"); $e->addHeader("to-uri", "sip:1019 at 192.168.2.3:5063;fs_path=sip:82.24.214.226:5063"); $e->addHeader("from-uri", "sip:1020 at pbx.rm.com"); $e->addHeader("host", "pbx.rm.com"); $e->addHeader("event-string", "check-sync"); $e->addHeader("content-type", "application/simple-message-summary"); $e->addBody("ok"); $res = $sockFSServerCommand->sendEvent($e); In our setup, we have several FreeSwitch boxes sitting behind two OpenSIPS load balancers. These Load Balancers handle registrations, etc. When we originate calls from FreeSwitch, we use an originate URL of the form (sofia/internal/1001 at sip.redmatter.com;fs_path=sip:84.45.30.2), and use fs_path to push the call via the right load balancer. This works perfectly. So, with the custom NOTIFY's I also need to make them follow the same path. As mentioned, we already have the contact string in our hand from the OpenSIPS registration server, but can't see how I can use that plus force the outbound path for the Notify. Was thinking I could use fs_path in the uri (as above) but doesn't seem to work, i.e. the wrong IP is used, and NOTIFY is messed up somewhat! Maybe another combination I am missing! Example packet capture follows: NOTIFY sip:1019 at 192.168.2.3:5063;fs_path=sip:82.24.214.22:5063 SIP/2.0 Via: SIP/2.0/UDP 212.85.24.245;rport;branch=z9hG4bKv268QHcy539yc Max-Forwards: 70 From: ;tag=0rer8BS2etmmc To: ;fs_path=sip:82.24.214.22:5063 Call-ID: 5a08d5c6-1a8e-122e-3885-0024e86c401a CSeq: 134921 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100211-0400-16602 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Event: check-sync Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=timeout Content-Type: application/simple-message-summary Content-Length: 2 ok Any thoughts gratefully appreciated. Rgds, Neil From stephen at mymessage.us Wed Aug 4 10:52:44 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Wed, 4 Aug 2010 13:52:44 -0400 Subject: [Freeswitch-users] Freeswitch Implementation Message-ID: Is there anyone in the US that would be interested in doing a basic Freeswitch implementation for us? Please contact me at stephen at fitnyc.edu Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/7b0d7ce4/attachment.html From mnhassan at usa.net Wed Aug 4 10:52:15 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Wed, 4 Aug 2010 23:52:15 +0600 Subject: [Freeswitch-users] Help on reading the log better Message-ID: Hi, I've finished the book, and was able to send a call through to another switch that we use. There was an initial codec mismatch (G729), but after reading the debug logs (fs_cli /log 7), it was identified and after fixing that, the call went through just fine. Today, I was trying some more changes on the default config, and the call will not go through. While that is not a problem as I can always go back to the default conf, what was bothering me is that I could not find out the right cause by reading the logs. The call also did not hit the other switch. Perhaps I am not reading something right. Can someone please help me identify what part of the logs below are showing where the calls are failing? The log is given below my signature. I changed my IPs and dialed number. Sorry for the inconvenience. Regards HASSAN 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected by acl "domains". Falling back to Digest auth. 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected by acl "domains". Falling back to Digest auth. 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/hassan at a.b.c.d) State NEW 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on nat.auto 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel sofia/internal/hassan at a.b.c.d entering state [received][100] 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: v=0 o=- 9 2 IN IP4 192.168.254.10 s=CounterPath Bria c=IN IP4 192.168.254.10 t=0 0 m=audio 63242 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf send/recv payload to 101 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/hassan at a.b.c.d) State INIT 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 sofia/internal/hassan at a.b.c.d SOFIA INIT 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/hassan at a.b.c.d) State INIT going to sleep 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/hassan at a.b.c.d) State ROUTING 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 sofia/internal/hassan at a.b.c.d SOFIA ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 sofia/internal/hassan at a.b.c.d Standard ROUTING 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing hassan->0111234567890 in context sky189 Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] continue=false Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false Dialplan: sofia/internal/hassan at a.b.c.d Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/hassan at a.b.c.d Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/hassan at a.b.c.d Action bridge(sofia/external/1234567890 at sky189) 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/hassan at a.b.c.d) State EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 sofia/internal/hassan at a.b.c.d SOFIA EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 sofia/internal/hassan at a.b.c.d Standard EXECUTE EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] EXECUTE sofia/internal/hassan at a.b.c.dbridge(sofia/external/1234567890 at sky189 ) 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_INIT 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1234567890 at sky189) State INIT 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 sofia/external/1234567890 at sky189 SOFIA INIT 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1234567890 at sky189) State INIT going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1234567890 at sky189) State ROUTING 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 sofia/external/1234567890 at sky189 SOFIA ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1234567890 at sky189) State ROUTING going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 (sofia/external/1234567890 at sky189) State CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [calling][0] 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [terminated][503] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal sofia/external/1234567890 at sky189 [KILL] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1234567890 at sky189) State HANGUP 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from the other leg 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel sofia/external/1234567890 at sky189 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 sofia/external/1234567890 at sky189 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1234567890 at sky189) State HANGUP going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> CS_REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 (sofia/external/1234567890 at sky189) State REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 sofia/external/1234567890 at sky189 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 (sofia/external/1234567890 at sky189) State REPORTING going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> CS_DESTROY 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal sofia/internal/hassan at a.b.c.d [KILL] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 sofia/internal/hassan at a.b.c.d skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/hassan at a.b.c.d) State HANGUP 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the other leg 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel sofia/internal/hassan at a.b.c.d hanging up, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 (sofia/external/1234567890 at sky189) Locked, Waiting on external entities 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 (sofia/external/1234567890 at sky189) Ended 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/1234567890 at sky189 [CS_DESTROY] 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 (sofia/external/1234567890 at sky189) State DESTROY 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 sofia/external/1234567890 at sky189 SOFIA DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 sofia/external/1234567890 at sky189 Standard DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 (sofia/external/1234567890 at sky189) State DESTROY going to sleep 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 503 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/hassan at a.b.c.d) State REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 (sofia/internal/hassan at a.b.c.d) Ended 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/hassan at a.b.c.d) State DESTROY 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 sofia/internal/hassan at a.b.c.d SOFIA DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 sofia/internal/hassan at a.b.c.d Standard DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/2d9ff797/attachment-0001.html From m.krivushin at imarto.net Wed Aug 4 12:09:55 2010 From: m.krivushin at imarto.net (Mikhail Krivushin) Date: Wed, 4 Aug 2010 23:09:55 +0400 Subject: [Freeswitch-users] 200 OK in tcpdump, but not anything in FS Message-ID: Hello, everyone! We have problem - leg B answer with 200 OK, but not anything appear in log and appear in tcpdump. -- ? ?????????, ???????? ?????? ?. ????? ???. +7 927 728 6799 icq: 218 744 127 xmpp: KrivushinME at jabber.ru skype: mkrivushin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/c4601403/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ngrgp.dump Type: application/octet-stream Size: 18142 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/c4601403/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: fs.dump Type: application/octet-stream Size: 54307 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/c4601403/attachment-0003.obj From shamun.toha at gmail.com Wed Aug 4 08:07:30 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Wed, 4 Aug 2010 15:07:30 +0000 (UTC) Subject: [Freeswitch-users] Callback - A made call to B by web Message-ID: Dear Sir, How can i make CallBack feature? Scenario GUI: +----------------------------------------+ | FreeSwitch - Web/Mobile/Desktop portal | |========================================| | | | Call from: __+3216000000___________ | | | | Call To: __+4414000000___________ | | | | [ CALL ] [ Call at 8AM ] | | | +----------------------------------------+ * Call from == my mobile number * Call to == where i want to call Question. How can we do this using FreeSwitch? Also to have A-leg and B-leg billing to be 2 seperate Thanks Regards From jaybinks at gmail.com Wed Aug 4 13:15:39 2010 From: jaybinks at gmail.com (jay binks) Date: Thu, 5 Aug 2010 06:15:39 +1000 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: Guys, further to this interesting discussion, please check out this wiki page http://wiki.freeswitch.org/wiki/Freeswitch_HA there was an announcement at cluecon that improves Freeswitches ability to do HA-clustering. please can you update this wiki page with the information you have learned doing the pacemaker side of this. also keep the conversation going now, with this new information :) Jay 2010/7/12 Steven Ayre > "As server FS work with share db, there is an impression that at switching > Cluster-IP with active on > a passive, passive server can pick up session begun on an active server > (the necessary data about > current sessions takes from the share db). Whether So it? > Testing has shown that at switching Cluster-IP on reserve FS, that does not > process session which > have begun on first FS :(" > > Sorry, but that's not currently possible as the call state information is > held in memory, not the shared database, and has execution threads started > for processing the signalling/media which aren't duplicated on the other > machine(s). > > -Steve > > > > 2010/7/9 viewpoint > > Many thanks for fast and substantial answers! >> Has established ipv4.ip_nonlocal_bind=1. >> Now there is no necessity to reboot FS or sofia profile. >> >> As a result it has turned out: >> 1) Two servers FS which kernel works with the shared db through unix ODBC >> 2) It Is realised Cluster-IP by installations Pacemaker >> (http://www.clusterlabs.org/wiki/Install#From_Source) >> >> As server FS work with share db, there is an impression that at switching >> Cluster-IP with active on >> a passive, passive server can pick up session begun on an active server >> (the necessary data about >> current sessions takes from the share db). Whether So it? >> Testing has shown that at switching Cluster-IP on reserve FS, that does >> not process session which >> have begun on first FS :( >> >> >> >> Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch < >> klejch+freeswitch at netbox.cz >: >> >> > >> > Hi >> > >> > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to non >> local >> > address on start of FS and then you don't need any restart of FS or >> > reload of sofia profile if this addres is active on the node >> > >> > or you can have this floating address on dummy iface (?or lo ) and then >> > use right settings in /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore >> > and then FS can use this address to bind on start and you don't need >> > restart of FS?or relaod of sofia profile ... >> > >> > >> > >> > Kleo >> > >> > On Thu, 8 Jul 2010, viewpoint wrote: >> > >> > > Hello. >> > > >> > > I currently have a project where I'm researching how to establish a >> > > clustered platform wieth failover time ~= some milliseconds. >> > > >> > > At present we have: 2 identical servers FS, wieth established >> Pacemaker >> > > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. On >> > > the one hand it is possible to tell that all works, as, at refusal of >> > > the first server, the second server receives cluster-ip. A problem >> that >> > > FreeSWITCH it is necessary to restart (or simply to start) that it >> began >> > > to work with the new IP-address. As the result, a switching total time >> > > makes 10-20 sec which basic part is necessary on launch FreeSwitch. >> > > >> > > any ideas? >> > > >> > > Thanks. >> > >> > >> > -- >> > klejch+freeswitch at netbox.cz >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/0a51ea5e/attachment.html From gmaruzz at celliax.org Wed Aug 4 13:32:18 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 4 Aug 2010 22:32:18 +0200 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: On Wed, Aug 4, 2010 at 10:15 PM, jay binks wrote: > Guys, > ??further to this interesting discussion, please check out this wiki page > http://wiki.freeswitch.org/wiki/Freeswitch_HA > there was an announcement at cluecon that improves Freeswitches ability to > do HA-clustering. > please can you update this wiki page with the information you have learned > doing the pacemaker side of this. Just read the page... I can only say: WOOOW! > also keep the conversation going now, with this new information :) > Jay > > 2010/7/12 Steven Ayre >> >> "As server FS work with share db, there is an impression that at switching >> Cluster-IP with active on >> a passive, passive server can pick up session begun on an active server >> (the necessary data about >> current sessions takes from the share db). Whether So it? >> Testing has shown that at switching Cluster-IP on reserve FS, that does >> not process session which >> have begun on first FS :(" >> >> Sorry, but that's not currently possible as the call state information is >> held in memory, not the shared database, and has execution threads started >> for processing the signalling/media which aren't duplicated on the other >> machine(s). >> >> -Steve >> >> >> >> 2010/7/9 viewpoint >>> >>> Many thanks for fast and substantial answers! >>> Has established ipv4.ip_nonlocal_bind=1. >>> Now there is no necessity to reboot FS or sofia profile. >>> >>> As a result it has turned out: >>> 1) Two servers FS which kernel works with the shared db through unix ODBC >>> 2) It Is realised Cluster-IP by installations Pacemaker >>> (http://www.clusterlabs.org/wiki/Install#From_Source) >>> >>> As server FS work with share db, there is an impression that at switching >>> Cluster-IP with active on >>> a passive, passive server can pick up session begun on an active server >>> (the necessary data about >>> current sessions takes from the share db). Whether So it? >>> Testing has shown that at switching Cluster-IP on reserve FS, that does >>> not process session which >>> have begun on first FS :( >>> >>> >>> >>> Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch >>> : >>> >>> > >>> > Hi >>> > >>> > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to non >>> > local >>> > address on start of FS and then you don't need any restart of FS or >>> > reload of sofia profile if this addres is active on the node >>> > >>> > or you can have this floating address on dummy iface (?or lo ) ?and >>> > then >>> > use right settings in >>> > /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore >>> > and then FS can use this address to bind on start and you don't need >>> > restart of FS?or relaod of sofia profile ... >>> > >>> > >>> > >>> > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? Kleo >>> > >>> > On Thu, 8 Jul 2010, viewpoint wrote: >>> > >>> > > Hello. >>> > > >>> > > I currently have a project where I'm researching how to establish a >>> > > clustered platform wieth failover time ~= some milliseconds. >>> > > >>> > > At present we have: 2 identical servers FS, wieth established >>> > > Pacemaker >>> > > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. On >>> > > the one hand it is possible to tell that all works, as, at refusal of >>> > > the first server, the second server receives cluster-ip. A problem >>> > > that >>> > > FreeSWITCH it is necessary to restart (or simply to start) that it >>> > > began >>> > > to work with the new IP-address. As the result, a switching total >>> > > time >>> > > makes 10-20 sec which basic part is necessary on launch FreeSwitch. >>> > > >>> > > any ideas? >>> > > >>> > > Thanks. >>> > >>> > >>> > -- >>> > ? ? ? klejch+freeswitch at netbox.cz >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely > > Jay > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From steveayre at gmail.com Wed Aug 4 14:13:04 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 4 Aug 2010 22:13:04 +0100 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: Cool! On 4 August 2010 21:15, jay binks wrote: > Guys, > further to this interesting discussion, please check out this wiki page > > http://wiki.freeswitch.org/wiki/Freeswitch_HA > > there was an announcement at cluecon that improves Freeswitches ability to > do HA-clustering. > please can you update this wiki page with the information you have learned > doing the pacemaker side of this. > > also keep the conversation going now, with this new information :) > > Jay > > > 2010/7/12 Steven Ayre > > "As server FS work with share db, there is an impression that at switching >> Cluster-IP with active on >> a passive, passive server can pick up session begun on an active server >> (the necessary data about >> current sessions takes from the share db). Whether So it? >> Testing has shown that at switching Cluster-IP on reserve FS, that does >> not process session which >> have begun on first FS :(" >> >> Sorry, but that's not currently possible as the call state information is >> held in memory, not the shared database, and has execution threads started >> for processing the signalling/media which aren't duplicated on the other >> machine(s). >> >> -Steve >> >> >> >> 2010/7/9 viewpoint >> >> Many thanks for fast and substantial answers! >>> Has established ipv4.ip_nonlocal_bind=1. >>> Now there is no necessity to reboot FS or sofia profile. >>> >>> As a result it has turned out: >>> 1) Two servers FS which kernel works with the shared db through unix ODBC >>> 2) It Is realised Cluster-IP by installations Pacemaker >>> (http://www.clusterlabs.org/wiki/Install#From_Source) >>> >>> As server FS work with share db, there is an impression that at switching >>> Cluster-IP with active on >>> a passive, passive server can pick up session begun on an active server >>> (the necessary data about >>> current sessions takes from the share db). Whether So it? >>> Testing has shown that at switching Cluster-IP on reserve FS, that does >>> not process session which >>> have begun on first FS :( >>> >>> >>> >>> Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch < >>> klejch+freeswitch at netbox.cz >: >>> >>> > >>> > Hi >>> > >>> > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to non >>> local >>> > address on start of FS and then you don't need any restart of FS or >>> > reload of sofia profile if this addres is active on the node >>> > >>> > or you can have this floating address on dummy iface (?or lo ) and >>> then >>> > use right settings in >>> /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore >>> > and then FS can use this address to bind on start and you don't need >>> > restart of FS?or relaod of sofia profile ... >>> > >>> > >>> > >>> > Kleo >>> > >>> > On Thu, 8 Jul 2010, viewpoint wrote: >>> > >>> > > Hello. >>> > > >>> > > I currently have a project where I'm researching how to establish a >>> > > clustered platform wieth failover time ~= some milliseconds. >>> > > >>> > > At present we have: 2 identical servers FS, wieth established >>> Pacemaker >>> > > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. >>> On >>> > > the one hand it is possible to tell that all works, as, at refusal of >>> > > the first server, the second server receives cluster-ip. A problem >>> that >>> > > FreeSWITCH it is necessary to restart (or simply to start) that it >>> began >>> > > to work with the new IP-address. As the result, a switching total >>> time >>> > > makes 10-20 sec which basic part is necessary on launch FreeSwitch. >>> > > >>> > > any ideas? >>> > > >>> > > Thanks. >>> > >>> > >>> > -- >>> > klejch+freeswitch at netbox.cz >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Sincerely > > Jay > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/cb45b028/attachment-0001.html From mnhassan at usa.net Wed Aug 4 14:40:05 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Thu, 5 Aug 2010 03:40:05 +0600 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: Wow Wow Wow! 2010/8/5 Steven Ayre > Cool! > > > > On 4 August 2010 21:15, jay binks wrote: > >> Guys, >> further to this interesting discussion, please check out this wiki page >> >> http://wiki.freeswitch.org/wiki/Freeswitch_HA >> >> there was an announcement at cluecon that improves Freeswitches ability to >> do HA-clustering. >> please can you update this wiki page with the information you have learned >> doing the pacemaker side of this. >> >> also keep the conversation going now, with this new information :) >> >> Jay >> >> >> 2010/7/12 Steven Ayre >> >> "As server FS work with share db, there is an impression that at switching >>> Cluster-IP with active on >>> a passive, passive server can pick up session begun on an active server >>> (the necessary data about >>> current sessions takes from the share db). Whether So it? >>> Testing has shown that at switching Cluster-IP on reserve FS, that does >>> not process session which >>> have begun on first FS :(" >>> >>> Sorry, but that's not currently possible as the call state information is >>> held in memory, not the shared database, and has execution threads started >>> for processing the signalling/media which aren't duplicated on the other >>> machine(s). >>> >>> -Steve >>> >>> >>> >>> 2010/7/9 viewpoint >>> >>> Many thanks for fast and substantial answers! >>>> Has established ipv4.ip_nonlocal_bind=1. >>>> Now there is no necessity to reboot FS or sofia profile. >>>> >>>> As a result it has turned out: >>>> 1) Two servers FS which kernel works with the shared db through unix >>>> ODBC >>>> 2) It Is realised Cluster-IP by installations Pacemaker >>>> (http://www.clusterlabs.org/wiki/Install#From_Source) >>>> >>>> As server FS work with share db, there is an impression that at >>>> switching Cluster-IP with active on >>>> a passive, passive server can pick up session begun on an active server >>>> (the necessary data about >>>> current sessions takes from the share db). Whether So it? >>>> Testing has shown that at switching Cluster-IP on reserve FS, that does >>>> not process session which >>>> have begun on first FS :( >>>> >>>> >>>> >>>> Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch < >>>> klejch+freeswitch at netbox.cz >: >>>> >>>> > >>>> > Hi >>>> > >>>> > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to >>>> non local >>>> > address on start of FS and then you don't need any restart of FS or >>>> > reload of sofia profile if this addres is active on the node >>>> > >>>> > or you can have this floating address on dummy iface (?or lo ) and >>>> then >>>> > use right settings in >>>> /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore >>>> > and then FS can use this address to bind on start and you don't need >>>> > restart of FS?or relaod of sofia profile ... >>>> > >>>> > >>>> > >>>> > Kleo >>>> > >>>> > On Thu, 8 Jul 2010, viewpoint wrote: >>>> > >>>> > > Hello. >>>> > > >>>> > > I currently have a project where I'm researching how to establish a >>>> > > clustered platform wieth failover time ~= some milliseconds. >>>> > > >>>> > > At present we have: 2 identical servers FS, wieth established >>>> Pacemaker >>>> > > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. >>>> On >>>> > > the one hand it is possible to tell that all works, as, at refusal >>>> of >>>> > > the first server, the second server receives cluster-ip. A problem >>>> that >>>> > > FreeSWITCH it is necessary to restart (or simply to start) that it >>>> began >>>> > > to work with the new IP-address. As the result, a switching total >>>> time >>>> > > makes 10-20 sec which basic part is necessary on launch FreeSwitch. >>>> > > >>>> > > any ideas? >>>> > > >>>> > > Thanks. >>>> > >>>> > >>>> > -- >>>> > klejch+freeswitch at netbox.cz >>>> > >>>> > _______________________________________________ >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> >>>> >>>> ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Sincerely >> >> Jay >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/a7d228b4/attachment.html From dujinfang at gmail.com Wed Aug 4 17:18:45 2010 From: dujinfang at gmail.com (Seven Du) Date: Thu, 5 Aug 2010 08:18:45 +0800 Subject: [Freeswitch-users] sips dial string question In-Reply-To: <61EEF09E-8A60-49C7-B253-97A696464267@freeswitch.org> References: <61EEF09E-8A60-49C7-B253-97A696464267@freeswitch.org> Message-ID: Thanks, but how would I use digest auth? Is sip_auth_username and sip_auth_password in the dialstr the only way? 2010/8/4 Brian West : > do this sofia/blah at blah;transport=tls > > /b > > On Aug 4, 2010, at 2:19 AM, Seven Du wrote: > >> Hi, >> >> I'm trying tls between 2 freeswitches, one as a server and one as a softphone. >> >> I think I successfully configured the server side, and copied the pem >> files to my client side. >> >> >> server: ?port 53 for sip and 54 for sips. >> client: external profile port 5080 for sip and 5081 for sips. >> >> I can successfully call the server from client: >> >> bgapi originate sofia/external/sips:xxxx at my-server.com:54 &echo >> >> questions: >> >> on client side, how to setup >> >> 1) register a gateway to server with sips. ? I tried >> realm=my-server.com:54 but doesn't seemed to work, is there any other >> param I should set ? >> I can register with sip(port 53). >> >> 2) then, can I use dial string like ? sofia/gateway/gw/xxxx ? ?with sips ? >> >> Thanks. >> >> -- >> Blog: http://www.dujinfang.com >> Proj: ?http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From jmesquita at freeswitch.org Wed Aug 4 17:49:01 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 4 Aug 2010 21:49:01 -0300 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: What if you have more then a switch on the cluster? How does the migration gets decided? I am confused on how this works. I guess I need to read the code... Jo?o Mesquita 2010/8/4 Nyamul Hassan > Wow Wow Wow! > > > > 2010/8/5 Steven Ayre > > Cool! >> >> >> >> On 4 August 2010 21:15, jay binks wrote: >> >>> Guys, >>> further to this interesting discussion, please check out this wiki >>> page >>> >>> http://wiki.freeswitch.org/wiki/Freeswitch_HA >>> >>> there was an announcement at cluecon that improves Freeswitches ability >>> to do HA-clustering. >>> please can you update this wiki page with the information you have >>> learned doing the pacemaker side of this. >>> >>> also keep the conversation going now, with this new information :) >>> >>> Jay >>> >>> >>> 2010/7/12 Steven Ayre >>> >>> "As server FS work with share db, there is an impression that at >>>> switching Cluster-IP with active on >>>> a passive, passive server can pick up session begun on an active server >>>> (the necessary data about >>>> current sessions takes from the share db). Whether So it? >>>> Testing has shown that at switching Cluster-IP on reserve FS, that does >>>> not process session which >>>> have begun on first FS :(" >>>> >>>> Sorry, but that's not currently possible as the call state information >>>> is held in memory, not the shared database, and has execution threads >>>> started for processing the signalling/media which aren't duplicated on the >>>> other machine(s). >>>> >>>> -Steve >>>> >>>> >>>> >>>> 2010/7/9 viewpoint >>>> >>>> Many thanks for fast and substantial answers! >>>>> Has established ipv4.ip_nonlocal_bind=1. >>>>> Now there is no necessity to reboot FS or sofia profile. >>>>> >>>>> As a result it has turned out: >>>>> 1) Two servers FS which kernel works with the shared db through unix >>>>> ODBC >>>>> 2) It Is realised Cluster-IP by installations Pacemaker >>>>> (http://www.clusterlabs.org/wiki/Install#From_Source) >>>>> >>>>> As server FS work with share db, there is an impression that at >>>>> switching Cluster-IP with active on >>>>> a passive, passive server can pick up session begun on an active server >>>>> (the necessary data about >>>>> current sessions takes from the share db). Whether So it? >>>>> Testing has shown that at switching Cluster-IP on reserve FS, that does >>>>> not process session which >>>>> have begun on first FS :( >>>>> >>>>> >>>>> >>>>> Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch < >>>>> klejch+freeswitch at netbox.cz >: >>>>> >>>>> > >>>>> > Hi >>>>> > >>>>> > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to >>>>> non local >>>>> > address on start of FS and then you don't need any restart of FS or >>>>> > reload of sofia profile if this addres is active on the node >>>>> > >>>>> > or you can have this floating address on dummy iface (?or lo ) and >>>>> then >>>>> > use right settings in >>>>> /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore >>>>> > and then FS can use this address to bind on start and you don't need >>>>> > restart of FS?or relaod of sofia profile ... >>>>> > >>>>> > >>>>> > >>>>> > Kleo >>>>> > >>>>> > On Thu, 8 Jul 2010, viewpoint wrote: >>>>> > >>>>> > > Hello. >>>>> > > >>>>> > > I currently have a project where I'm researching how to establish a >>>>> > > clustered platform wieth failover time ~= some milliseconds. >>>>> > > >>>>> > > At present we have: 2 identical servers FS, wieth established >>>>> Pacemaker >>>>> > > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. >>>>> On >>>>> > > the one hand it is possible to tell that all works, as, at refusal >>>>> of >>>>> > > the first server, the second server receives cluster-ip. A problem >>>>> that >>>>> > > FreeSWITCH it is necessary to restart (or simply to start) that it >>>>> began >>>>> > > to work with the new IP-address. As the result, a switching total >>>>> time >>>>> > > makes 10-20 sec which basic part is necessary on launch FreeSwitch. >>>>> > > >>>>> > > any ideas? >>>>> > > >>>>> > > Thanks. >>>>> > >>>>> > >>>>> > -- >>>>> > klejch+freeswitch at netbox.cz >>>>> > >>>>> > _______________________________________________ >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> >>>>> >>>>> ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Sincerely >>> >>> Jay >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/6b6ef5e4/attachment-0001.html From infos at madovsky.org Wed Aug 4 19:21:15 2010 From: infos at madovsky.org (Madovsky) Date: Wed, 4 Aug 2010 22:21:15 -0400 Subject: [Freeswitch-users] FS HA-cluster References: Message-ID: I think it's only for failover, not clone ----- Original Message ----- From: Jo?o Mesquita To: FreeSWITCH Users Help Sent: Wednesday, August 04, 2010 8:49 PM Subject: Re: [Freeswitch-users] FS HA-cluster What if you have more then a switch on the cluster? How does the migration gets decided? I am confused on how this works. I guess I need to read the code... Jo?o Mesquita 2010/8/4 Nyamul Hassan Wow Wow Wow! 2010/8/5 Steven Ayre Cool! On 4 August 2010 21:15, jay binks wrote: Guys, further to this interesting discussion, please check out this wiki page http://wiki.freeswitch.org/wiki/Freeswitch_HA there was an announcement at cluecon that improves Freeswitches ability to do HA-clustering. please can you update this wiki page with the information you have learned doing the pacemaker side of this. also keep the conversation going now, with this new information :) Jay 2010/7/12 Steven Ayre "As server FS work with share db, there is an impression that at switching Cluster-IP with active on a passive, passive server can pick up session begun on an active server (the necessary data about current sessions takes from the share db). Whether So it? Testing has shown that at switching Cluster-IP on reserve FS, that does not process session which have begun on first FS :(" Sorry, but that's not currently possible as the call state information is held in memory, not the shared database, and has execution threads started for processing the signalling/media which aren't duplicated on the other machine(s). -Steve 2010/7/9 viewpoint Many thanks for fast and substantial answers! Has established ipv4.ip_nonlocal_bind=1. Now there is no necessity to reboot FS or sofia profile. As a result it has turned out: 1) Two servers FS which kernel works with the shared db through unix ODBC 2) It Is realised Cluster-IP by installations Pacemaker (http://www.clusterlabs.org/wiki/Install#From_Source) As server FS work with share db, there is an impression that at switching Cluster-IP with active on a passive, passive server can pick up session begun on an active server (the necessary data about current sessions takes from the share db). Whether So it? Testing has shown that at switching Cluster-IP on reserve FS, that does not process session which have begun on first FS :( Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch : > > Hi > > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to non local > address on start of FS and then you don't need any restart of FS or > reload of sofia profile if this addres is active on the node > > or you can have this floating address on dummy iface (?or lo ) and then > use right settings in /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore > and then FS can use this address to bind on start and you don't need > restart of FS?or relaod of sofia profile ... > > > > Kleo > > On Thu, 8 Jul 2010, viewpoint wrote: > > > Hello. > > > > I currently have a project where I'm researching how to establish a > > clustered platform wieth failover time ~= some milliseconds. > > > > At present we have: 2 identical servers FS, wieth established Pacemaker > > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. On > > the one hand it is possible to tell that all works, as, at refusal of > > the first server, the second server receives cluster-ip. A problem that > > FreeSWITCH it is necessary to restart (or simply to start) that it began > > to work with the new IP-address. As the result, a switching total time > > makes 10-20 sec which basic part is necessary on launch FreeSwitch. > > > > any ideas? > > > > Thanks. > > > -- > klejch+freeswitch at netbox.cz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely Jay _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100804/30046b93/attachment.html From b_ball_henry at hotmail.com Wed Aug 4 21:57:44 2010 From: b_ball_henry at hotmail.com (Henry Huang) Date: Thu, 5 Aug 2010 12:57:44 +0800 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: This is awesome Does the 1.0.6 version support the track call feature? Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me [image: Linkedin][image: Facebook] [image: Twitter] On Thu, Aug 5, 2010 at 10:21 AM, Madovsky wrote: > I think it's only for failover, not clone > > ----- Original Message ----- > *From:* Jo?o Mesquita > *To:* FreeSWITCH Users Help > *Sent:* Wednesday, August 04, 2010 8:49 PM > *Subject:* Re: [Freeswitch-users] FS HA-cluster > > What if you have more then a switch on the cluster? > > How does the migration gets decided? I am confused on how this works. I > guess I need to read the code... > > Jo?o Mesquita > > > 2010/8/4 Nyamul Hassan > >> Wow Wow Wow! >> >> >> >> 2010/8/5 Steven Ayre >> >> Cool! >>> >>> >>> >>> On 4 August 2010 21:15, jay binks wrote: >>> >>>> Guys, >>>> further to this interesting discussion, please check out this wiki >>>> page >>>> >>>> http://wiki.freeswitch.org/wiki/Freeswitch_HA >>>> >>>> there was an announcement at cluecon that improves Freeswitches ability >>>> to do HA-clustering. >>>> please can you update this wiki page with the information you have >>>> learned doing the pacemaker side of this. >>>> >>>> also keep the conversation going now, with this new information :) >>>> >>>> Jay >>>> >>>> >>>> 2010/7/12 Steven Ayre >>>> >>>> "As server FS work with share db, there is an impression that at >>>>> switching Cluster-IP with active on >>>>> a passive, passive server can pick up session begun on an active server >>>>> (the necessary data about >>>>> current sessions takes from the share db). Whether So it? >>>>> Testing has shown that at switching Cluster-IP on reserve FS, that does >>>>> not process session which >>>>> have begun on first FS :(" >>>>> >>>>> Sorry, but that's not currently possible as the call state information >>>>> is held in memory, not the shared database, and has execution threads >>>>> started for processing the signalling/media which aren't duplicated on the >>>>> other machine(s). >>>>> >>>>> -Steve >>>>> >>>>> >>>>> >>>>> 2010/7/9 viewpoint >>>>> >>>>> Many thanks for fast and substantial answers! >>>>>> Has established ipv4.ip_nonlocal_bind=1. >>>>>> Now there is no necessity to reboot FS or sofia profile. >>>>>> >>>>>> As a result it has turned out: >>>>>> 1) Two servers FS which kernel works with the shared db through unix >>>>>> ODBC >>>>>> 2) It Is realised Cluster-IP by installations Pacemaker >>>>>> (http://www.clusterlabs.org/wiki/Install#From_Source) >>>>>> >>>>>> As server FS work with share db, there is an impression that at >>>>>> switching Cluster-IP with active on >>>>>> a passive, passive server can pick up session begun on an active >>>>>> server (the necessary data about >>>>>> current sessions takes from the share db). Whether So it? >>>>>> Testing has shown that at switching Cluster-IP on reserve FS, that >>>>>> does not process session which >>>>>> have begun on first FS :( >>>>>> >>>>>> >>>>>> >>>>>> Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch < >>>>>> klejch+freeswitch at netbox.cz >: >>>>>> >>>>>> > >>>>>> > Hi >>>>>> > >>>>>> > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to >>>>>> non local >>>>>> > address on start of FS and then you don't need any restart of FS or >>>>>> > reload of sofia profile if this addres is active on the node >>>>>> > >>>>>> > or you can have this floating address on dummy iface (?or lo ) and >>>>>> then >>>>>> > use right settings in >>>>>> /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore >>>>>> > and then FS can use this address to bind on start and you don't need >>>>>> > restart of FS?or relaod of sofia profile ... >>>>>> > >>>>>> > >>>>>> > >>>>>> > Kleo >>>>>> > >>>>>> > On Thu, 8 Jul 2010, viewpoint wrote: >>>>>> > >>>>>> > > Hello. >>>>>> > > >>>>>> > > I currently have a project where I'm researching how to establish >>>>>> a >>>>>> > > clustered platform wieth failover time ~= some milliseconds. >>>>>> > > >>>>>> > > At present we have: 2 identical servers FS, wieth established >>>>>> Pacemaker >>>>>> > > (http://www.clusterlabs.org/wiki/Install#From_Source) - >>>>>> heartbeat. On >>>>>> > > the one hand it is possible to tell that all works, as, at refusal >>>>>> of >>>>>> > > the first server, the second server receives cluster-ip. A problem >>>>>> that >>>>>> > > FreeSWITCH it is necessary to restart (or simply to start) that it >>>>>> began >>>>>> > > to work with the new IP-address. As the result, a switching total >>>>>> time >>>>>> > > makes 10-20 sec which basic part is necessary on launch >>>>>> FreeSwitch. >>>>>> > > >>>>>> > > any ideas? >>>>>> > > >>>>>> > > Thanks. >>>>>> > >>>>>> > >>>>>> > -- >>>>>> > klejch+freeswitch at netbox.cz >>>>>> > >>>>>> > _______________________________________________ >>>>>> > FreeSWITCH-users mailing list >>>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> >>>>>> >>>>>> ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Sincerely >>>> >>>> Jay >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/2702f7c8/attachment-0001.html From infos at madovsky.org Wed Aug 4 22:11:29 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 01:11:29 -0400 Subject: [Freeswitch-users] FS HA-cluster References: Message-ID: I think it's the last git ----- Original Message ----- From: Henry Huang To: FreeSWITCH Users Help Sent: Thursday, August 05, 2010 12:57 AM Subject: Re: [Freeswitch-users] FS HA-cluster This is awesome Does the 1.0.6 version support the track call feature? Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me On Thu, Aug 5, 2010 at 10:21 AM, Madovsky wrote: I think it's only for failover, not clone ----- Original Message ----- From: Jo?o Mesquita To: FreeSWITCH Users Help Sent: Wednesday, August 04, 2010 8:49 PM Subject: Re: [Freeswitch-users] FS HA-cluster What if you have more then a switch on the cluster? How does the migration gets decided? I am confused on how this works. I guess I need to read the code... Jo?o Mesquita 2010/8/4 Nyamul Hassan Wow Wow Wow! 2010/8/5 Steven Ayre Cool! On 4 August 2010 21:15, jay binks wrote: Guys, further to this interesting discussion, please check out this wiki page http://wiki.freeswitch.org/wiki/Freeswitch_HA there was an announcement at cluecon that improves Freeswitches ability to do HA-clustering. please can you update this wiki page with the information you have learned doing the pacemaker side of this. also keep the conversation going now, with this new information :) Jay 2010/7/12 Steven Ayre "As server FS work with share db, there is an impression that at switching Cluster-IP with active on a passive, passive server can pick up session begun on an active server (the necessary data about current sessions takes from the share db). Whether So it? Testing has shown that at switching Cluster-IP on reserve FS, that does not process session which have begun on first FS :(" Sorry, but that's not currently possible as the call state information is held in memory, not the shared database, and has execution threads started for processing the signalling/media which aren't duplicated on the other machine(s). -Steve 2010/7/9 viewpoint Many thanks for fast and substantial answers! Has established ipv4.ip_nonlocal_bind=1. Now there is no necessity to reboot FS or sofia profile. As a result it has turned out: 1) Two servers FS which kernel works with the shared db through unix ODBC 2) It Is realised Cluster-IP by installations Pacemaker (http://www.clusterlabs.org/wiki/Install#From_Source) As server FS work with share db, there is an impression that at switching Cluster-IP with active on a passive, passive server can pick up session begun on an active server (the necessary data about current sessions takes from the share db). Whether So it? Testing has shown that at switching Cluster-IP on reserve FS, that does not process session which have begun on first FS :( Thu, 8 Jul 2010 16:21:40 +0200 (CEST) ?????? ?? Vladimir Klejch : > > Hi > > On linux you can use /proc/sys/net/ipv4/ip_nonlocal_bind to bind to non local > address on start of FS and then you don't need any restart of FS or > reload of sofia profile if this addres is active on the node > > or you can have this floating address on dummy iface (?or lo ) and then > use right settings in /proc/sys/net/ipv4/conf/eth?/arp_filter,arp_ignore > and then FS can use this address to bind on start and you don't need > restart of FS?or relaod of sofia profile ... > > > > Kleo > > On Thu, 8 Jul 2010, viewpoint wrote: > > > Hello. > > > > I currently have a project where I'm researching how to establish a > > clustered platform wieth failover time ~= some milliseconds. > > > > At present we have: 2 identical servers FS, wieth established Pacemaker > > (http://www.clusterlabs.org/wiki/Install#From_Source) - heartbeat. On > > the one hand it is possible to tell that all works, as, at refusal of > > the first server, the second server receives cluster-ip. A problem that > > FreeSWITCH it is necessary to restart (or simply to start) that it began > > to work with the new IP-address. As the result, a switching total time > > makes 10-20 sec which basic part is necessary on launch FreeSwitch. > > > > any ideas? > > > > Thanks. > > > -- > klejch+freeswitch at netbox.cz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ? ? ???? ???? - http://my.mail.ru/list/yurazilot1/ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely Jay _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/79e36726/attachment-0001.html From jaybinks at gmail.com Wed Aug 4 22:20:46 2010 From: jaybinks at gmail.com (Jay Binks) Date: Thu, 5 Aug 2010 00:20:46 -0500 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: Message-ID: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com> Sure does support it... In fact its been hidden in the codebase for a while... :) Sent from my iPad On Aug 4, 2010, at 11:57 PM, Henry Huang wrote: > This is awesome > > Does the 1.0.6 version support the track call feature? > > > > > > Henry Huang > Unified Communication System R&D Project Manager > US: +1 (626) 606-3306 > Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com > Contact Me -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/c61aac94/attachment.html From b_ball_henry at hotmail.com Wed Aug 4 22:39:00 2010 From: b_ball_henry at hotmail.com (Henry Huang) Date: Thu, 5 Aug 2010 13:39:00 +0800 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com> References: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com> Message-ID: so when I use the core_ODBC function. Does those session information getting written to the database as well? If so , I can have several FS box maintain the same session base right? For example, I have a load balancer in front of FS 1, 2, and 3. FS 1, 2, and 3 all have the share the same core DB via core_ODBC. Then when one of the FS fails , the calls will be directed to one of the live FS server and continue the conversation? (All the FS shares the same virtual IP, this is a LVS implementation) Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me [image: Linkedin][image: Facebook] [image: Twitter] On Thu, Aug 5, 2010 at 1:20 PM, Jay Binks wrote: > Sure does support it... > > In fact its been hidden in the codebase for a while... :) > > > Sent from my iPad > > On Aug 4, 2010, at 11:57 PM, Henry Huang wrote: > > This is awesome > > Does the 1.0.6 version support the track call feature? > > > > > > Henry Huang > Unified Communication System R&D Project Manager > US: +1 (626) 606-3306 > Chat Skype: unicsolution MSN: > b_ball_henry at hotmail.com > Contact Me [image: Linkedin][image: > Facebook] [image: > Twitter] > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/41fbbfe3/attachment.html From infos at madovsky.org Wed Aug 4 22:48:16 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 01:48:16 -0400 Subject: [Freeswitch-users] FS HA-cluster References: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com> Message-ID: <17A5BD02DF55446E9F06D8C8447DDABE@MOBILEE1705> > (All the FS shares the same virtual IP, this is a LVS implementation) do you use cluster IP ? ----- Original Message ----- From: Henry Huang To: FreeSWITCH Users Help Sent: Thursday, August 05, 2010 1:39 AM Subject: Re: [Freeswitch-users] FS HA-cluster so when I use the core_ODBC function. Does those session information getting written to the database as well? If so , I can have several FS box maintain the same session base right? For example, I have a load balancer in front of FS 1, 2, and 3. FS 1, 2, and 3 all have the share the same core DB via core_ODBC. Then when one of the FS fails , the calls will be directed to one of the live FS server and continue the conversation? (All the FS shares the same virtual IP, this is a LVS implementation) Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me On Thu, Aug 5, 2010 at 1:20 PM, Jay Binks wrote: Sure does support it... In fact its been hidden in the codebase for a while... :) Sent from my iPad On Aug 4, 2010, at 11:57 PM, Henry Huang wrote: This is awesome Does the 1.0.6 version support the track call feature? Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/36acd6fa/attachment-0001.html From b_ball_henry at hotmail.com Wed Aug 4 23:23:00 2010 From: b_ball_henry at hotmail.com (Henry Huang) Date: Thu, 5 Aug 2010 14:23:00 +0800 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: <17A5BD02DF55446E9F06D8C8447DDABE@MOBILEE1705> References: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com> <17A5BD02DF55446E9F06D8C8447DDABE@MOBILEE1705> Message-ID: I am not sure of the term "cluster IP" but here is the example Load balancer have the actual public IP , while all FS servers has the same IP setup on the box as virtual IP and listens on the virtual IP. So in other words, all the FS servers are listening to the same IP address through out the architecture. Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me [image: Linkedin][image: Facebook] [image: Twitter] > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/32509778/attachment.html From steveayre at gmail.com Thu Aug 5 00:40:10 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 5 Aug 2010 08:40:10 +0100 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com> <17A5BD02DF55446E9F06D8C8447DDABE@MOBILEE1705> Message-ID: If I guess how this works correctly it'll only handle failover, threads will still be needed in the FreeSWITCH architecture. I think call state is written to the database in realtime, then if FS fails a new instance of FS is started on another machine, reads call states from the database and starts threads for each call. Just a guess though. That'd mean it's read when FreeSWITCH starts up, so couldn't handle multiple servers on the same IP but would handle primary and backup servers. -Steve On 5 August 2010 07:23, Henry Huang wrote: > I am not sure of the term "cluster IP" but here is the example > > Load balancer have the actual public IP , while all FS servers has the same > IP setup on the box as virtual IP and listens on the virtual IP. So in other > words, all the FS servers are listening to the same IP address through out > the architecture. > > Henry Huang > Unified Communication System R&D Project Manager > US: +1 (626) 606-3306 > Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com > Contact Me [image: Linkedin][image: > Facebook] [image: > Twitter] > > >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/29d1e99c/attachment.html From infos at madovsky.org Thu Aug 5 00:54:06 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 03:54:06 -0400 Subject: [Freeswitch-users] FS HA-cluster References: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com><17A5BD02DF55446E9F06D8C8447DDABE@MOBILEE1705> Message-ID: <5651A30B6D2E40BE968F348C613CD629@MOBILEE1705> I will try this param tomorrow I have a cluster with ODBC and indie IPs.... ----- Original Message ----- From: Steven Ayre To: FreeSWITCH Users Help Sent: Thursday, August 05, 2010 3:40 AM Subject: Re: [Freeswitch-users] FS HA-cluster If I guess how this works correctly it'll only handle failover, threads will still be needed in the FreeSWITCH architecture. I think call state is written to the database in realtime, then if FS fails a new instance of FS is started on another machine, reads call states from the database and starts threads for each call. Just a guess though. That'd mean it's read when FreeSWITCH starts up, so couldn't handle multiple servers on the same IP but would handle primary and backup servers. -Steve On 5 August 2010 07:23, Henry Huang wrote: I am not sure of the term "cluster IP" but here is the example Load balancer have the actual public IP , while all FS servers has the same IP setup on the box as virtual IP and listens on the virtual IP. So in other words, all the FS servers are listening to the same IP address through out the architecture. Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/16f27b05/attachment.html From lists at infosecurity.ch Thu Aug 5 00:55:24 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Thu, 05 Aug 2010 09:55:24 +0200 Subject: [Freeswitch-users] On FS-HA feature: HA + Balancing ? Message-ID: <4C5A6E6C.3010905@infosecurity.ch> Hi guys, first of all my compliments for FS-HA features, something really enterprise that make the life of infrastructure architect and system administration much simpler. My simpler question is: Does FS with HA features can do "High availability AND Load Balancing" without using third party SIP Proxy? If so the FS would simplify a lot most of the IP architecture as it may be possible to just put more FS boxes into a HA+Balancing "server farm" when you need more power. Is that possible? Fabio From steveayre at gmail.com Thu Aug 5 01:46:14 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 5 Aug 2010 09:46:14 +0100 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: Message-ID: The call is getting 503 Service Unavailable. 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [calling][0] 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [terminated][503] This either comes from the remote host, or from within the SIP stack. An example of when I've seen Sofia generate 503 errors is when DNS lookups were failing. I'm guessing this is what is happening in your case as you're dialing "sofia/external/1234567890 at sky189". sky189 isn't a domain name, so it won't resolve to anything for sofia to dial (unless you have a very unusual setup). My guess is what you're actually trying to do is call through a gateway named sky189. The syntax for that is different: "sofia/gateway/sky189/1234567890" If that's not it then more debugging information should let you track down the error. If you enable siptrace for the profile you are calling out on, then the log will include the SIP messages sent. This will let you see if the 503 is a reply from the destination. If it isn't then it's probably within the sofia SIP stack. You can enable debugging for that stack to see why the error occurs (this would show the dns error for example). The wiki shows how to enable this debugging: http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP -Steve On 4 August 2010 18:52, Nyamul Hassan wrote: > Hi, > > I've finished the book, and was able to send a call through to another > switch that we use. There was an initial codec mismatch (G729), but after > reading the debug logs (fs_cli /log 7), it was identified and after fixing > that, the call went through just fine. > > Today, I was trying some more changes on the default config, and the call > will not go through. While that is not a problem as I can always go back to > the default conf, what was bothering me is that I could not find out the > right cause by reading the logs. The call also did not hit the other > switch. Perhaps I am not reading something right. Can someone please help > me identify what part of the logs below are showing where the calls are > failing? > > The log is given below my signature. I changed my IPs and dialed number. > Sorry for the inconvenience. > > Regards > HASSAN > > > 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected > by acl "domains". Falling back to Digest auth. > 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected > by acl "domains". Falling back to Digest auth. > 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel > sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] > 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW > 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 > (sofia/internal/hassan at a.b.c.d) State NEW > 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on > nat.auto > 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel > sofia/internal/hassan at a.b.c.d entering state [received][100] > 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: > v=0 > o=- 9 2 IN IP4 192.168.254.10 > s=CounterPath Bria > c=IN IP4 192.168.254.10 > t=0 0 > m=audio 63242 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 > a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 > a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 > > 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare > [G729:18:8000:20]/[G729:18:8000:20] > 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec > sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples > 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf > send/recv payload to 101 > 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 > (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/hassan at a.b.c.d) State INIT > 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 > sofia/internal/hassan at a.b.c.d SOFIA INIT > 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 > (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/hassan at a.b.c.d) State INIT going to sleep > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING > 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 > (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/hassan at a.b.c.d) State ROUTING > 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 > sofia/internal/hassan at a.b.c.d SOFIA ROUTING > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 > sofia/internal/hassan at a.b.c.d Standard ROUTING > 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing > hassan->0111234567890 in context sky189 > Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] > continue=false > Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] > destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false > Dialplan: sofia/internal/hassan at a.b.c.d Action > set(effective_caller_id_number=${outbound_caller_id_number}) > Dialplan: sofia/internal/hassan at a.b.c.d Action > set(effective_caller_id_name=${outbound_caller_id_name}) > Dialplan: sofia/internal/hassan at a.b.c.d Action > bridge(sofia/external/1234567890 at sky189) > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 > (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/hassan at a.b.c.d) State EXECUTE > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 > sofia/internal/hassan at a.b.c.d SOFIA EXECUTE > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 > sofia/internal/hassan at a.b.c.d Standard EXECUTE > EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) > 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 > sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] > EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) > 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 > sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] > EXECUTE sofia/internal/hassan at a.b.c.dbridge(sofia/external/1234567890 at sky189 > ) > 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel > sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 > (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_INIT > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 > (sofia/external/1234567890 at sky189) State INIT > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 > sofia/external/1234567890 at sky189 SOFIA INIT > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 > (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 > (sofia/external/1234567890 at sky189) State INIT going to sleep > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING > 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 > (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 > (sofia/external/1234567890 at sky189) State ROUTING > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 > sofia/external/1234567890 at sky189 SOFIA ROUTING > 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 > (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 > (sofia/external/1234567890 at sky189) State ROUTING going to sleep > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 > (sofia/external/1234567890 at sky189) State CONSUME_MEDIA > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 > (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep > 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel > sofia/external/1234567890 at sky189 entering state [calling][0] > 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel > sofia/external/1234567890 at sky189 entering state [terminated][503] > 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 > (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP > 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup > sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] > [NORMAL_TEMPORARY_FAILURE] > 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal > sofia/external/1234567890 at sky189 [KILL] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 > (sofia/external/1234567890 at sky189) State HANGUP > 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 > sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from > the other leg > 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel > sofia/external/1234567890 at sky189 hanging up, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 > sofia/external/1234567890 at sky189 Standard HANGUP, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 > (sofia/external/1234567890 at sky189) State HANGUP going to sleep > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 > (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> CS_REPORTING > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 > (sofia/external/1234567890 at sky189) State REPORTING > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 > sofia/external/1234567890 at sky189 Standard REPORTING, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 > (sofia/external/1234567890 at sky189) State REPORTING going to sleep > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 > (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> CS_DESTROY > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate > Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] > 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. > Cause: NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 > (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP > 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup > sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal > sofia/internal/hassan at a.b.c.d [KILL] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 > sofia/internal/hassan at a.b.c.d skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/hassan at a.b.c.d) State HANGUP > 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 > sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the > other leg > 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/hassan at a.b.c.d hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 > (sofia/external/1234567890 at sky189) Locked, Waiting on external entities > 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 > (sofia/external/1234567890 at sky189) Ended > 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/1234567890 at sky189 [CS_DESTROY] > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 > (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 > (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/1234567890 at sky189) State DESTROY > 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 > sofia/external/1234567890 at sky189 SOFIA DESTROY > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 > sofia/external/1234567890 at sky189 Standard DESTROY > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/1234567890 at sky189) State DESTROY going to sleep > 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE > with: 503 > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/hassan at a.b.c.d) State REPORTING > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 > (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities > 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 > (sofia/internal/hassan at a.b.c.d) Ended > 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 > (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY > 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/hassan at a.b.c.d) State DESTROY > 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 > sofia/internal/hassan at a.b.c.d SOFIA DESTROY > 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/hassan at a.b.c.d Standard DESTROY > 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/ee024eee/attachment-0001.html From mnhassan at usa.net Thu Aug 5 02:09:14 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Thu, 5 Aug 2010 15:09:14 +0600 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: Message-ID: Thank you for the detailed response Steven. You were right, I was comparing my original config to the new one, and found that I did need to use it as you have said, sofia/gateway/sky189/$1 That being said, I was also unable to make a gateway without using a "username" / "password". My providers don't need any username password, and use our IP address for authentication. We can use the "sofia/internal/$1@" format though. Regards HASSAN On Thu, Aug 5, 2010 at 14:46, Steven Ayre wrote: > The call is getting 503 Service Unavailable. > > 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel > sofia/external/1234567890 at sky189 entering state [calling][0] > 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel > sofia/external/1234567890 at sky189 entering state [terminated][503] > > This either comes from the remote host, or from within the SIP stack. > > An example of when I've seen Sofia generate 503 errors is when DNS lookups > were failing. I'm guessing this is what is happening in your case as you're > dialing "sofia/external/1234567890 at sky189". > > sky189 isn't a domain name, so it won't resolve to anything for sofia to > dial (unless you have a very unusual setup). My guess is what you're > actually trying to do is call through a gateway named sky189. The syntax for > that is different: "sofia/gateway/sky189/1234567890" > > If that's not it then more debugging information should let you track down > the error. > If you enable siptrace for the profile you are calling out on, then the log > will include the SIP messages sent. This will let you see if the 503 is a > reply from the destination. > If it isn't then it's probably within the sofia SIP stack. You can enable > debugging for that stack to see why the error occurs (this would show the > dns error for example). The wiki shows how to enable this debugging: > http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP > > -Steve > > > > On 4 August 2010 18:52, Nyamul Hassan wrote: > >> Hi, >> >> I've finished the book, and was able to send a call through to another >> switch that we use. There was an initial codec mismatch (G729), but after >> reading the debug logs (fs_cli /log 7), it was identified and after fixing >> that, the call went through just fine. >> >> Today, I was trying some more changes on the default config, and the call >> will not go through. While that is not a problem as I can always go back to >> the default conf, what was bothering me is that I could not find out the >> right cause by reading the logs. The call also did not hit the other >> switch. Perhaps I am not reading something right. Can someone please help >> me identify what part of the logs below are showing where the calls are >> failing? >> >> The log is given below my signature. I changed my IPs and dialed number. >> Sorry for the inconvenience. >> >> Regards >> HASSAN >> >> >> 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected >> by acl "domains". Falling back to Digest auth. >> 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected >> by acl "domains". Falling back to Digest auth. >> 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel >> sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] >> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW >> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 >> (sofia/internal/hassan at a.b.c.d) State NEW >> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on >> nat.auto >> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel >> sofia/internal/hassan at a.b.c.d entering state [received][100] >> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: >> v=0 >> o=- 9 2 IN IP4 192.168.254.10 >> s=CounterPath Bria >> c=IN IP4 192.168.254.10 >> t=0 0 >> m=audio 63242 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 >> a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 >> a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 >> >> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare >> [G729:18:8000:20]/[G729:18:8000:20] >> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec >> sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples >> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf >> send/recv payload to 101 >> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 >> (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/hassan at a.b.c.d) State INIT >> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 >> sofia/internal/hassan at a.b.c.d SOFIA INIT >> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 >> (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/hassan at a.b.c.d) State INIT going to sleep >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING >> 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 >> (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/hassan at a.b.c.d) State ROUTING >> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 >> sofia/internal/hassan at a.b.c.d SOFIA ROUTING >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 >> sofia/internal/hassan at a.b.c.d Standard ROUTING >> 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing >> hassan->0111234567890 in context sky189 >> Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] >> continue=false >> Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] >> destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false >> Dialplan: sofia/internal/hassan at a.b.c.d Action >> set(effective_caller_id_number=${outbound_caller_id_number}) >> Dialplan: sofia/internal/hassan at a.b.c.d Action >> set(effective_caller_id_name=${outbound_caller_id_name}) >> Dialplan: sofia/internal/hassan at a.b.c.d Action >> bridge(sofia/external/1234567890 at sky189) >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 >> (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/hassan at a.b.c.d) State EXECUTE >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 >> sofia/internal/hassan at a.b.c.d SOFIA EXECUTE >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 >> sofia/internal/hassan at a.b.c.d Standard EXECUTE >> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) >> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] >> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) >> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] >> EXECUTE sofia/internal/hassan at a.b.c.dbridge(sofia/external/1234567890 at sky189 >> ) >> 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel >> sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 >> (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_INIT >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >> (sofia/external/1234567890 at sky189) State INIT >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 >> sofia/external/1234567890 at sky189 SOFIA INIT >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 >> (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >> (sofia/external/1234567890 at sky189) State INIT going to sleep >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING >> 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 >> (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >> (sofia/external/1234567890 at sky189) State ROUTING >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 >> sofia/external/1234567890 at sky189 SOFIA ROUTING >> 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 >> (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >> (sofia/external/1234567890 at sky189) State ROUTING going to sleep >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep >> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [calling][0] >> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [terminated][503] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >> (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP >> 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup >> sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] >> [NORMAL_TEMPORARY_FAILURE] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >> sofia/external/1234567890 at sky189 [KILL] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >> (sofia/external/1234567890 at sky189) State HANGUP >> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >> sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from >> the other leg >> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >> sofia/external/1234567890 at sky189 hanging up, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 >> sofia/external/1234567890 at sky189 Standard HANGUP, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >> (sofia/external/1234567890 at sky189) State HANGUP going to sleep >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 >> (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> CS_REPORTING >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >> (sofia/external/1234567890 at sky189) State REPORTING >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 >> sofia/external/1234567890 at sky189 Standard REPORTING, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >> (sofia/external/1234567890 at sky189) State REPORTING going to sleep >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 >> (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> >> CS_DESTROY >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate >> Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] >> 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. >> Cause: NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >> (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP >> 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup >> sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >> sofia/internal/hassan at a.b.c.d [KILL] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 >> sofia/internal/hassan at a.b.c.d skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >> (sofia/internal/hassan at a.b.c.d) State HANGUP >> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >> sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the >> other leg >> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >> sofia/internal/hassan at a.b.c.d hanging up, cause: NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 >> (sofia/external/1234567890 at sky189) Locked, Waiting on external entities >> 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 >> (sofia/external/1234567890 at sky189) Ended >> 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close >> Channel sofia/external/1234567890 at sky189 [CS_DESTROY] >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 >> (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 >> (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/1234567890 at sky189) State DESTROY >> 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 >> sofia/external/1234567890 at sky189 SOFIA DESTROY >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 >> sofia/external/1234567890 at sky189 Standard DESTROY >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/1234567890 at sky189) State DESTROY going to sleep >> 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE >> with: 503 >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 >> (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >> (sofia/internal/hassan at a.b.c.d) State REPORTING >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >> (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 >> (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities >> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 >> (sofia/internal/hassan at a.b.c.d) Ended >> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close >> Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 >> (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY >> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/hassan at a.b.c.d) State DESTROY >> 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 >> sofia/internal/hassan at a.b.c.d SOFIA DESTROY >> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/hassan at a.b.c.d Standard DESTROY >> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/48172a03/attachment-0001.html From peter.olsson at visionutveckling.se Thu Aug 5 02:20:26 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 5 Aug 2010 11:20:26 +0200 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC07857C@cooper> The gateway must have username/password configured, but you can set register=false to tell you're not using it. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Nyamul Hassan Skickat: den 5 augusti 2010 11:09 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Help on reading the log better Thank you for the detailed response Steven. You were right, I was comparing my original config to the new one, and found that I did need to use it as you have said, sofia/gateway/sky189/$1 That being said, I was also unable to make a gateway without using a "username" / "password". My providers don't need any username password, and use our IP address for authentication. We can use the "sofia/internal/$1@" format though. Regards HASSAN On Thu, Aug 5, 2010 at 14:46, Steven Ayre > wrote: The call is getting 503 Service Unavailable. 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [calling][0] 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [terminated][503] This either comes from the remote host, or from within the SIP stack. An example of when I've seen Sofia generate 503 errors is when DNS lookups were failing. I'm guessing this is what is happening in your case as you're dialing "sofia/external/1234567890 at sky189". sky189 isn't a domain name, so it won't resolve to anything for sofia to dial (unless you have a very unusual setup). My guess is what you're actually trying to do is call through a gateway named sky189. The syntax for that is different: "sofia/gateway/sky189/1234567890" If that's not it then more debugging information should let you track down the error. If you enable siptrace for the profile you are calling out on, then the log will include the SIP messages sent. This will let you see if the 503 is a reply from the destination. If it isn't then it's probably within the sofia SIP stack. You can enable debugging for that stack to see why the error occurs (this would show the dns error for example). The wiki shows how to enable this debugging: http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP -Steve On 4 August 2010 18:52, Nyamul Hassan > wrote: Hi, I've finished the book, and was able to send a call through to another switch that we use. There was an initial codec mismatch (G729), but after reading the debug logs (fs_cli /log 7), it was identified and after fixing that, the call went through just fine. Today, I was trying some more changes on the default config, and the call will not go through. While that is not a problem as I can always go back to the default conf, what was bothering me is that I could not find out the right cause by reading the logs. The call also did not hit the other switch. Perhaps I am not reading something right. Can someone please help me identify what part of the logs below are showing where the calls are failing? The log is given below my signature. I changed my IPs and dialed number. Sorry for the inconvenience. Regards HASSAN 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected by acl "domains". Falling back to Digest auth. 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected by acl "domains". Falling back to Digest auth. 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/hassan at a.b.c.d) State NEW 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on nat.auto 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel sofia/internal/hassan at a.b.c.d entering state [received][100] 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: v=0 o=- 9 2 IN IP4 192.168.254.10 s=CounterPath Bria c=IN IP4 192.168.254.10 t=0 0 m=audio 63242 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf send/recv payload to 101 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/hassan at a.b.c.d) State INIT 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 sofia/internal/hassan at a.b.c.d SOFIA INIT 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/hassan at a.b.c.d) State INIT going to sleep 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/hassan at a.b.c.d) State ROUTING 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 sofia/internal/hassan at a.b.c.d SOFIA ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 sofia/internal/hassan at a.b.c.d Standard ROUTING 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing hassan->0111234567890 in context sky189 Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] continue=false Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false Dialplan: sofia/internal/hassan at a.b.c.d Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/hassan at a.b.c.d Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/hassan at a.b.c.d Action bridge(sofia/external/1234567890 at sky189) 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/hassan at a.b.c.d) State EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 sofia/internal/hassan at a.b.c.d SOFIA EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 sofia/internal/hassan at a.b.c.d Standard EXECUTE EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] EXECUTE sofia/internal/hassan at a.b.c.d bridge(sofia/external/1234567890 at sky189) 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_INIT 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1234567890 at sky189) State INIT 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 sofia/external/1234567890 at sky189 SOFIA INIT 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1234567890 at sky189) State INIT going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1234567890 at sky189) State ROUTING 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 sofia/external/1234567890 at sky189 SOFIA ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1234567890 at sky189) State ROUTING going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 (sofia/external/1234567890 at sky189) State CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [calling][0] 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [terminated][503] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal sofia/external/1234567890 at sky189 [KILL] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1234567890 at sky189) State HANGUP 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from the other leg 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel sofia/external/1234567890 at sky189 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 sofia/external/1234567890 at sky189 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1234567890 at sky189) State HANGUP going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> CS_REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 (sofia/external/1234567890 at sky189) State REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 sofia/external/1234567890 at sky189 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 (sofia/external/1234567890 at sky189) State REPORTING going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> CS_DESTROY 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal sofia/internal/hassan at a.b.c.d [KILL] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 sofia/internal/hassan at a.b.c.d skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/hassan at a.b.c.d) State HANGUP 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the other leg 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel sofia/internal/hassan at a.b.c.d hanging up, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 (sofia/external/1234567890 at sky189) Locked, Waiting on external entities 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 (sofia/external/1234567890 at sky189) Ended 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/1234567890 at sky189 [CS_DESTROY] 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 (sofia/external/1234567890 at sky189) State DESTROY 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 sofia/external/1234567890 at sky189 SOFIA DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 sofia/external/1234567890 at sky189 Standard DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 (sofia/external/1234567890 at sky189) State DESTROY going to sleep 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 503 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/hassan at a.b.c.d) State REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 (sofia/internal/hassan at a.b.c.d) Ended 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/hassan at a.b.c.d) State DESTROY 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 sofia/internal/hassan at a.b.c.d SOFIA DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 sofia/internal/hassan at a.b.c.d Standard DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c5a81c132932113517252! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/f1576390/attachment-0001.html From gmaruzz at celliax.org Thu Aug 5 02:23:19 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Aug 2010 11:23:19 +0200 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: Message-ID: The format you uses is the correct one in your case. Gateways are only the ones you authenticate with. If you don't need to authenticate, you don't need gateway. Is in the book, IIRC -giovanni On 8/5/10, Nyamul Hassan wrote: > Thank you for the detailed response Steven. You were right, I was comparing > my original config to the new one, and found that I did need to use it as > you have said, sofia/gateway/sky189/$1 > > That being said, I was also unable to make a gateway without using a > "username" / "password". My providers don't need any username password, and > use our IP address for authentication. We can use the > "sofia/internal/$1@ of provider>" format though. > > Regards > HASSAN > > > > On Thu, Aug 5, 2010 at 14:46, Steven Ayre wrote: > >> The call is getting 503 Service Unavailable. >> >> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [calling][0] >> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [terminated][503] >> >> This either comes from the remote host, or from within the SIP stack. >> >> An example of when I've seen Sofia generate 503 errors is when DNS lookups >> were failing. I'm guessing this is what is happening in your case as >> you're >> dialing "sofia/external/1234567890 at sky189". >> >> sky189 isn't a domain name, so it won't resolve to anything for sofia to >> dial (unless you have a very unusual setup). My guess is what you're >> actually trying to do is call through a gateway named sky189. The syntax >> for >> that is different: "sofia/gateway/sky189/1234567890" >> >> If that's not it then more debugging information should let you track down >> the error. >> If you enable siptrace for the profile you are calling out on, then the >> log >> will include the SIP messages sent. This will let you see if the 503 is a >> reply from the destination. >> If it isn't then it's probably within the sofia SIP stack. You can enable >> debugging for that stack to see why the error occurs (this would show the >> dns error for example). The wiki shows how to enable this debugging: >> http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP >> >> -Steve >> >> >> >> On 4 August 2010 18:52, Nyamul Hassan wrote: >> >>> Hi, >>> >>> I've finished the book, and was able to send a call through to another >>> switch that we use. There was an initial codec mismatch (G729), but >>> after >>> reading the debug logs (fs_cli /log 7), it was identified and after >>> fixing >>> that, the call went through just fine. >>> >>> Today, I was trying some more changes on the default config, and the call >>> will not go through. While that is not a problem as I can always go back >>> to >>> the default conf, what was bothering me is that I could not find out the >>> right cause by reading the logs. The call also did not hit the other >>> switch. Perhaps I am not reading something right. Can someone please >>> help >>> me identify what part of the logs below are showing where the calls are >>> failing? >>> >>> The log is given below my signature. I changed my IPs and dialed number. >>> Sorry for the inconvenience. >>> >>> Regards >>> HASSAN >>> >>> >>> 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 >>> Rejected >>> by acl "domains". Falling back to Digest auth. >>> 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 >>> Rejected >>> by acl "domains". Falling back to Digest auth. >>> 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel >>> sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] >>> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW >>> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 >>> (sofia/internal/hassan at a.b.c.d) State NEW >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on >>> nat.auto >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel >>> sofia/internal/hassan at a.b.c.d entering state [received][100] >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: >>> v=0 >>> o=- 9 2 IN IP4 192.168.254.10 >>> s=CounterPath Bria >>> c=IN IP4 192.168.254.10 >>> t=0 0 >>> m=audio 63242 RTP/AVP 18 101 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 >>> a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 >>> a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 >>> >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare >>> [G729:18:8000:20]/[G729:18:8000:20] >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec >>> sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf >>> send/recv payload to 101 >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/hassan at a.b.c.d) State INIT >>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 >>> sofia/internal/hassan at a.b.c.d SOFIA INIT >>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/hassan at a.b.c.d) State INIT going to sleep >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 >>> (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/hassan at a.b.c.d) State ROUTING >>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 >>> sofia/internal/hassan at a.b.c.d SOFIA ROUTING >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 >>> sofia/internal/hassan at a.b.c.d Standard ROUTING >>> 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing >>> hassan->0111234567890 in context sky189 >>> Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] >>> continue=false >>> Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] >>> destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false >>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>> set(effective_caller_id_number=${outbound_caller_id_number}) >>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>> set(effective_caller_id_name=${outbound_caller_id_name}) >>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>> bridge(sofia/external/1234567890 at sky189) >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/hassan at a.b.c.d) State EXECUTE >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 >>> sofia/internal/hassan at a.b.c.d SOFIA EXECUTE >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 >>> sofia/internal/hassan at a.b.c.d Standard EXECUTE >>> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >>> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] >>> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >>> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] >>> EXECUTE >>> sofia/internal/hassan at a.b.c.dbridge(sofia/external/1234567890 at sky189 >>> ) >>> 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel >>> sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 >>> (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_INIT >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/external/1234567890 at sky189) State INIT >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 >>> sofia/external/1234567890 at sky189 SOFIA INIT >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 >>> (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/external/1234567890 at sky189) State INIT going to sleep >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 >>> (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/external/1234567890 at sky189) State ROUTING >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 >>> sofia/external/1234567890 at sky189 SOFIA ROUTING >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 >>> (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> >>> CS_CONSUME_MEDIA >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/external/1234567890 at sky189) State ROUTING going to sleep >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >>> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >>> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep >>> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >>> sofia/external/1234567890 at sky189 entering state [calling][0] >>> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >>> sofia/external/1234567890 at sky189 entering state [terminated][503] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >>> (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP >>> 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup >>> sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] >>> [NORMAL_TEMPORARY_FAILURE] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >>> sofia/external/1234567890 at sky189 [KILL] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>> (sofia/external/1234567890 at sky189) State HANGUP >>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >>> sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from >>> the other leg >>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >>> sofia/external/1234567890 at sky189 hanging up, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 >>> sofia/external/1234567890 at sky189 Standard HANGUP, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>> (sofia/external/1234567890 at sky189) State HANGUP going to sleep >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> CS_REPORTING >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >>> (sofia/external/1234567890 at sky189) State REPORTING >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 >>> sofia/external/1234567890 at sky189 Standard REPORTING, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >>> (sofia/external/1234567890 at sky189) State REPORTING going to sleep >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 >>> (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> >>> CS_DESTROY >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate >>> Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] >>> 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. >>> Cause: NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >>> (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP >>> 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup >>> sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >>> sofia/internal/hassan at a.b.c.d [KILL] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 >>> sofia/internal/hassan at a.b.c.d skip receive message >>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>> (sofia/internal/hassan at a.b.c.d) State HANGUP >>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >>> sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the >>> other leg >>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >>> sofia/internal/hassan at a.b.c.d hanging up, cause: NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 >>> (sofia/external/1234567890 at sky189) Locked, Waiting on external entities >>> 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 >>> (sofia/external/1234567890 at sky189) Ended >>> 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close >>> Channel sofia/external/1234567890 at sky189 [CS_DESTROY] >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 >>> (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 >>> (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/external/1234567890 at sky189) State DESTROY >>> 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 >>> sofia/external/1234567890 at sky189 SOFIA DESTROY >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 >>> sofia/external/1234567890 at sky189 Standard DESTROY >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/external/1234567890 at sky189) State DESTROY going to sleep >>> 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE >>> with: 503 >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 >>> sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 >>> (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >>> (sofia/internal/hassan at a.b.c.d) State REPORTING >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 >>> sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >>> (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 >>> (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities >>> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 >>> (sofia/internal/hassan at a.b.c.d) Ended >>> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close >>> Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 >>> (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY >>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/internal/hassan at a.b.c.d) State DESTROY >>> 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 >>> sofia/internal/hassan at a.b.c.d SOFIA DESTROY >>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 >>> sofia/internal/hassan at a.b.c.d Standard DESTROY >>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From steveayre at gmail.com Thu Aug 5 03:16:02 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 5 Aug 2010 11:16:02 +0100 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: Message-ID: As already mentioned, one is required. But it's only used if the gateway gives a 401 Not Authorized response. If you don't need authentication, just make one up. -Steve On 5 August 2010 10:09, Nyamul Hassan wrote: > Thank you for the detailed response Steven. You were right, I was > comparing my original config to the new one, and found that I did need to > use it as you have said, sofia/gateway/sky189/$1 > > That being said, I was also unable to make a gateway without using a > "username" / "password". My providers don't need any username password, and > use our IP address for authentication. We can use the "sofia/internal/$1@ of provider>" format though. > > Regards > HASSAN > > > > On Thu, Aug 5, 2010 at 14:46, Steven Ayre wrote: > >> The call is getting 503 Service Unavailable. >> >> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [calling][0] >> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [terminated][503] >> >> This either comes from the remote host, or from within the SIP stack. >> >> An example of when I've seen Sofia generate 503 errors is when DNS lookups >> were failing. I'm guessing this is what is happening in your case as you're >> dialing "sofia/external/1234567890 at sky189". >> >> sky189 isn't a domain name, so it won't resolve to anything for sofia to >> dial (unless you have a very unusual setup). My guess is what you're >> actually trying to do is call through a gateway named sky189. The syntax for >> that is different: "sofia/gateway/sky189/1234567890" >> >> If that's not it then more debugging information should let you track down >> the error. >> If you enable siptrace for the profile you are calling out on, then the >> log will include the SIP messages sent. This will let you see if the 503 is >> a reply from the destination. >> If it isn't then it's probably within the sofia SIP stack. You can enable >> debugging for that stack to see why the error occurs (this would show the >> dns error for example). The wiki shows how to enable this debugging: >> http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP >> >> -Steve >> >> >> >> On 4 August 2010 18:52, Nyamul Hassan wrote: >> >>> Hi, >>> >>> I've finished the book, and was able to send a call through to another >>> switch that we use. There was an initial codec mismatch (G729), but after >>> reading the debug logs (fs_cli /log 7), it was identified and after fixing >>> that, the call went through just fine. >>> >>> Today, I was trying some more changes on the default config, and the call >>> will not go through. While that is not a problem as I can always go back to >>> the default conf, what was bothering me is that I could not find out the >>> right cause by reading the logs. The call also did not hit the other >>> switch. Perhaps I am not reading something right. Can someone please help >>> me identify what part of the logs below are showing where the calls are >>> failing? >>> >>> The log is given below my signature. I changed my IPs and dialed number. >>> Sorry for the inconvenience. >>> >>> Regards >>> HASSAN >>> >>> >>> 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 >>> Rejected by acl "domains". Falling back to Digest auth. >>> 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 >>> Rejected by acl "domains". Falling back to Digest auth. >>> 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel >>> sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] >>> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW >>> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 >>> (sofia/internal/hassan at a.b.c.d) State NEW >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on >>> nat.auto >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel >>> sofia/internal/hassan at a.b.c.d entering state [received][100] >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: >>> v=0 >>> o=- 9 2 IN IP4 192.168.254.10 >>> s=CounterPath Bria >>> c=IN IP4 192.168.254.10 >>> t=0 0 >>> m=audio 63242 RTP/AVP 18 101 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 >>> a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 >>> a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 >>> >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare >>> [G729:18:8000:20]/[G729:18:8000:20] >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec >>> sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf >>> send/recv payload to 101 >>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/hassan at a.b.c.d) State INIT >>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 >>> sofia/internal/hassan at a.b.c.d SOFIA INIT >>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/internal/hassan at a.b.c.d) State INIT going to sleep >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 >>> (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/hassan at a.b.c.d) State ROUTING >>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 >>> sofia/internal/hassan at a.b.c.d SOFIA ROUTING >>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 >>> sofia/internal/hassan at a.b.c.d Standard ROUTING >>> 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing >>> hassan->0111234567890 in context sky189 >>> Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] >>> continue=false >>> Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] >>> destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false >>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>> set(effective_caller_id_number=${outbound_caller_id_number}) >>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>> set(effective_caller_id_name=${outbound_caller_id_name}) >>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>> bridge(sofia/external/1234567890 at sky189) >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/hassan at a.b.c.d) State EXECUTE >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 >>> sofia/internal/hassan at a.b.c.d SOFIA EXECUTE >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 >>> sofia/internal/hassan at a.b.c.d Standard EXECUTE >>> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >>> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] >>> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >>> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] >>> EXECUTE sofia/internal/hassan at a.b.c.dbridge(sofia/external/1234567890 at sky189 >>> ) >>> 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel >>> sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 >>> (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_INIT >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/external/1234567890 at sky189) State INIT >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 >>> sofia/external/1234567890 at sky189 SOFIA INIT >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 >>> (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >>> (sofia/external/1234567890 at sky189) State INIT going to sleep >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 >>> (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/external/1234567890 at sky189) State ROUTING >>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 >>> sofia/external/1234567890 at sky189 SOFIA ROUTING >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 >>> (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> >>> CS_CONSUME_MEDIA >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>> (sofia/external/1234567890 at sky189) State ROUTING going to sleep >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >>> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA >>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >>> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep >>> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >>> sofia/external/1234567890 at sky189 entering state [calling][0] >>> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >>> sofia/external/1234567890 at sky189 entering state [terminated][503] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >>> (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP >>> 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup >>> sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] >>> [NORMAL_TEMPORARY_FAILURE] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >>> sofia/external/1234567890 at sky189 [KILL] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>> (sofia/external/1234567890 at sky189) State HANGUP >>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >>> sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from >>> the other leg >>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >>> sofia/external/1234567890 at sky189 hanging up, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 >>> sofia/external/1234567890 at sky189 Standard HANGUP, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>> (sofia/external/1234567890 at sky189) State HANGUP going to sleep >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> >>> CS_REPORTING >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >>> (sofia/external/1234567890 at sky189) State REPORTING >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 >>> sofia/external/1234567890 at sky189 Standard REPORTING, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >>> (sofia/external/1234567890 at sky189) State REPORTING going to sleep >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 >>> (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> >>> CS_DESTROY >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/external/1234567890 at sky189 [BREAK] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate >>> Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] >>> 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. >>> Cause: NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >>> (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP >>> 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup >>> sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >>> sofia/internal/hassan at a.b.c.d [KILL] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 >>> sofia/internal/hassan at a.b.c.d skip receive message >>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>> (sofia/internal/hassan at a.b.c.d) State HANGUP >>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >>> sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the >>> other leg >>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >>> sofia/internal/hassan at a.b.c.d hanging up, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 >>> (sofia/external/1234567890 at sky189) Locked, Waiting on external entities >>> 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 >>> (sofia/external/1234567890 at sky189) Ended >>> 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close >>> Channel sofia/external/1234567890 at sky189 [CS_DESTROY] >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 >>> (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 >>> (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/external/1234567890 at sky189) State DESTROY >>> 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 >>> sofia/external/1234567890 at sky189 SOFIA DESTROY >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 >>> sofia/external/1234567890 at sky189 Standard DESTROY >>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/external/1234567890 at sky189) State DESTROY going to sleep >>> 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE >>> with: 503 >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 >>> sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 >>> (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >>> (sofia/internal/hassan at a.b.c.d) State REPORTING >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 >>> sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: >>> NORMAL_TEMPORARY_FAILURE >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >>> (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 >>> (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal >>> sofia/internal/hassan at a.b.c.d [BREAK] >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 >>> (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities >>> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 >>> (sofia/internal/hassan at a.b.c.d) Ended >>> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close >>> Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 >>> (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN >>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 >>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY >>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/internal/hassan at a.b.c.d) State DESTROY >>> 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 >>> sofia/internal/hassan at a.b.c.d SOFIA DESTROY >>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 >>> sofia/internal/hassan at a.b.c.d Standard DESTROY >>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/c947c78b/attachment-0001.html From b_ball_henry at hotmail.com Thu Aug 5 04:34:28 2010 From: b_ball_henry at hotmail.com (Henry Huang) Date: Thu, 5 Aug 2010 19:34:28 +0800 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: <5651A30B6D2E40BE968F348C613CD629@MOBILEE1705> References: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com> <17A5BD02DF55446E9F06D8C8447DDABE@MOBILEE1705> <5651A30B6D2E40BE968F348C613CD629@MOBILEE1705> Message-ID: Madovsky: I think the track feature only works under the same IP. Under HA mode, the FS server will be using the exact same IP addresses. That's why I said if I have all the FS servers listening on the same IP behind a LVS load balancer, I wonder if it works. Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me [image: Linkedin][image: Facebook] [image: Twitter] On Thu, Aug 5, 2010 at 3:54 PM, Madovsky wrote: > I will try this param tomorrow I have a cluster with ODBC and indie > IPs.... > > ----- Original Message ----- > *From:* Steven Ayre > *To:* FreeSWITCH Users Help > *Sent:* Thursday, August 05, 2010 3:40 AM > *Subject:* Re: [Freeswitch-users] FS HA-cluster > > If I guess how this works correctly it'll only handle failover, threads > will still be needed in the FreeSWITCH architecture. I think call state is > written to the database in realtime, then if FS fails a new instance of FS > is started on another machine, reads call states from the database and > starts threads for each call. Just a guess though. That'd mean it's read > when FreeSWITCH starts up, so couldn't handle multiple servers on the same > IP but would handle primary and backup servers. > > -Steve > > > On 5 August 2010 07:23, Henry Huang wrote: > >> I am not sure of the term "cluster IP" but here is the example >> >> Load balancer have the actual public IP , while all FS servers has the >> same IP setup on the box as virtual IP and listens on the virtual IP. So in >> other words, all the FS servers are listening to the same IP address through >> out the architecture. >> >> Henry Huang >> Unified Communication System R&D Project Manager >> US: +1 (626) 606-3306 >> Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com >> Contact Me [image: Linkedin][image: >> Facebook] [image: >> Twitter] >> >> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/42399c79/attachment.html From brian at freeswitch.org Thu Aug 5 06:40:58 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Aug 2010 08:40:58 -0500 Subject: [Freeswitch-users] sips dial string question In-Reply-To: References: Message-ID: <772149D0-BF66-48A6-8F8D-A473CF11F579@freeswitch.org> It does but their might be cases that aren't handled properly. Since Aastra is the only phone I know of that does it like that and testing was limited. /b On Aug 4, 2010, at 10:24 AM, Kristian Kielhofner wrote: > Does freeswitch support the sips uri scheme? > > > > -- > Kristian Kielhofner > http://blog.krisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/4fbb36c7/attachment.html From mnhassan at usa.net Thu Aug 5 06:47:37 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Thu, 5 Aug 2010 19:47:37 +0600 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: Message-ID: I've set the "registration" to false, but whenever FS attempts to call the gateway, it tries to send the call using the user/pass given. For some switches, that also messes up the CallerID. I'm using the "non-gateway" format for now, and it works. Thanks to everyone for their help. Regards HASSAN On Thu, Aug 5, 2010 at 16:16, Steven Ayre wrote: > As already mentioned, one is required. > > But it's only used if the gateway gives a 401 Not Authorized response. If > you don't need authentication, just make one up. > > -Steve > > > > > On 5 August 2010 10:09, Nyamul Hassan wrote: > >> Thank you for the detailed response Steven. You were right, I was >> comparing my original config to the new one, and found that I did need to >> use it as you have said, sofia/gateway/sky189/$1 >> >> That being said, I was also unable to make a gateway without using a >> "username" / "password". My providers don't need any username password, and >> use our IP address for authentication. We can use the "sofia/internal/$1@> of provider>" format though. >> >> Regards >> HASSAN >> >> >> >> On Thu, Aug 5, 2010 at 14:46, Steven Ayre wrote: >> >>> The call is getting 503 Service Unavailable. >>> >>> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >>> sofia/external/1234567890 at sky189 entering state [calling][0] >>> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >>> sofia/external/1234567890 at sky189 entering state [terminated][503] >>> >>> This either comes from the remote host, or from within the SIP stack. >>> >>> An example of when I've seen Sofia generate 503 errors is when DNS >>> lookups were failing. I'm guessing this is what is happening in your case as >>> you're dialing "sofia/external/1234567890 at sky189". >>> >>> sky189 isn't a domain name, so it won't resolve to anything for sofia to >>> dial (unless you have a very unusual setup). My guess is what you're >>> actually trying to do is call through a gateway named sky189. The syntax for >>> that is different: "sofia/gateway/sky189/1234567890" >>> >>> If that's not it then more debugging information should let you track >>> down the error. >>> If you enable siptrace for the profile you are calling out on, then the >>> log will include the SIP messages sent. This will let you see if the 503 is >>> a reply from the destination. >>> If it isn't then it's probably within the sofia SIP stack. You can enable >>> debugging for that stack to see why the error occurs (this would show the >>> dns error for example). The wiki shows how to enable this debugging: >>> http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP >>> >>> -Steve >>> >>> >>> >>> On 4 August 2010 18:52, Nyamul Hassan wrote: >>> >>>> Hi, >>>> >>>> I've finished the book, and was able to send a call through to another >>>> switch that we use. There was an initial codec mismatch (G729), but after >>>> reading the debug logs (fs_cli /log 7), it was identified and after fixing >>>> that, the call went through just fine. >>>> >>>> Today, I was trying some more changes on the default config, and the >>>> call will not go through. While that is not a problem as I can always go >>>> back to the default conf, what was bothering me is that I could not find out >>>> the right cause by reading the logs. The call also did not hit the other >>>> switch. Perhaps I am not reading something right. Can someone please help >>>> me identify what part of the logs below are showing where the calls are >>>> failing? >>>> >>>> The log is given below my signature. I changed my IPs and dialed >>>> number. Sorry for the inconvenience. >>>> >>>> Regards >>>> HASSAN >>>> >>>> >>>> 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 >>>> Rejected by acl "domains". Falling back to Digest auth. >>>> 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 >>>> Rejected by acl "domains". Falling back to Digest auth. >>>> 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel >>>> sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] >>>> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW >>>> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 >>>> (sofia/internal/hassan at a.b.c.d) State NEW >>>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based >>>> on nat.auto >>>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel >>>> sofia/internal/hassan at a.b.c.d entering state [received][100] >>>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: >>>> v=0 >>>> o=- 9 2 IN IP4 192.168.254.10 >>>> s=CounterPath Bria >>>> c=IN IP4 192.168.254.10 >>>> t=0 0 >>>> m=audio 63242 RTP/AVP 18 101 >>>> a=rtpmap:18 G729/8000 >>>> a=fmtp:18 annexb=yes >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 >>>> a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 >>>> a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 >>>> >>>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare >>>> [G729:18:8000:20]/[G729:18:8000:20] >>>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec >>>> sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples >>>> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf >>>> send/recv payload to 101 >>>> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 >>>> (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/internal/hassan at a.b.c.d [BREAK] >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/internal/hassan at a.b.c.d) State INIT >>>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 >>>> sofia/internal/hassan at a.b.c.d SOFIA INIT >>>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 >>>> (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/internal/hassan at a.b.c.d [BREAK] >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/internal/hassan at a.b.c.d) State INIT going to sleep >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 >>>> (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/internal/hassan at a.b.c.d) State ROUTING >>>> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 >>>> sofia/internal/hassan at a.b.c.d SOFIA ROUTING >>>> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 >>>> sofia/internal/hassan at a.b.c.d Standard ROUTING >>>> 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing >>>> hassan->0111234567890 in context sky189 >>>> Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] >>>> continue=false >>>> Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] >>>> destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false >>>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>>> set(effective_caller_id_number=${outbound_caller_id_number}) >>>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>>> set(effective_caller_id_name=${outbound_caller_id_name}) >>>> Dialplan: sofia/internal/hassan at a.b.c.d Action >>>> bridge(sofia/external/1234567890 at sky189) >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 >>>> (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/internal/hassan at a.b.c.d [BREAK] >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 >>>> (sofia/internal/hassan at a.b.c.d) State EXECUTE >>>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 >>>> sofia/internal/hassan at a.b.c.d SOFIA EXECUTE >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 >>>> sofia/internal/hassan at a.b.c.d Standard EXECUTE >>>> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) >>>> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >>>> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] >>>> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) >>>> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >>>> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] >>>> EXECUTE sofia/internal/hassan at a.b.c.dbridge(sofia/external/1234567890 at sky189 >>>> ) >>>> 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel >>>> sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] >>>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 >>>> (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/external/1234567890 at sky189 [BREAK] >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/1234567890 at sky189) Running State Change CS_INIT >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/external/1234567890 at sky189) State INIT >>>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 >>>> sofia/external/1234567890 at sky189 SOFIA INIT >>>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 >>>> (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/external/1234567890 at sky189 [BREAK] >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >>>> (sofia/external/1234567890 at sky189) State INIT going to sleep >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 >>>> (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/external/1234567890 at sky189) State ROUTING >>>> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 >>>> sofia/external/1234567890 at sky189 SOFIA ROUTING >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 >>>> (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> >>>> CS_CONSUME_MEDIA >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/external/1234567890 at sky189 [BREAK] >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >>>> (sofia/external/1234567890 at sky189) State ROUTING going to sleep >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/1234567890 at sky189) Running State Change >>>> CS_CONSUME_MEDIA >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >>>> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA >>>> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >>>> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep >>>> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >>>> sofia/external/1234567890 at sky189 entering state [calling][0] >>>> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >>>> sofia/external/1234567890 at sky189 entering state [terminated][503] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >>>> (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP >>>> 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup >>>> sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] >>>> [NORMAL_TEMPORARY_FAILURE] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >>>> sofia/external/1234567890 at sky189 [KILL] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/external/1234567890 at sky189 [BREAK] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>>> (sofia/external/1234567890 at sky189) State HANGUP >>>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >>>> sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from >>>> the other leg >>>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >>>> sofia/external/1234567890 at sky189 hanging up, cause: >>>> NORMAL_TEMPORARY_FAILURE >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/external/1234567890 at sky189 Standard HANGUP, cause: >>>> NORMAL_TEMPORARY_FAILURE >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>>> (sofia/external/1234567890 at sky189) State HANGUP going to sleep >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 >>>> (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> >>>> CS_REPORTING >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/external/1234567890 at sky189 [BREAK] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >>>> (sofia/external/1234567890 at sky189) State REPORTING >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/external/1234567890 at sky189 Standard REPORTING, cause: >>>> NORMAL_TEMPORARY_FAILURE >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >>>> (sofia/external/1234567890 at sky189) State REPORTING going to sleep >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 >>>> (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> >>>> CS_DESTROY >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/external/1234567890 at sky189 [BREAK] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate >>>> Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] >>>> 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. >>>> Cause: NORMAL_TEMPORARY_FAILURE >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >>>> (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP >>>> 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup >>>> sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >>>> sofia/internal/hassan at a.b.c.d [KILL] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/internal/hassan at a.b.c.d [BREAK] >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 >>>> sofia/internal/hassan at a.b.c.d skip receive message >>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 >>>> (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >>>> (sofia/internal/hassan at a.b.c.d) State HANGUP >>>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >>>> sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from >>>> the other leg >>>> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >>>> sofia/internal/hassan at a.b.c.d hanging up, cause: >>>> NORMAL_TEMPORARY_FAILURE >>>> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 >>>> (sofia/external/1234567890 at sky189) Locked, Waiting on external entities >>>> 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session >>>> 11 (sofia/external/1234567890 at sky189) Ended >>>> 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close >>>> Channel sofia/external/1234567890 at sky189 [CS_DESTROY] >>>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 >>>> (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN >>>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 >>>> (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY >>>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >>>> (sofia/external/1234567890 at sky189) State DESTROY >>>> 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 >>>> sofia/external/1234567890 at sky189 SOFIA DESTROY >>>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/external/1234567890 at sky189 Standard DESTROY >>>> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >>>> (sofia/external/1234567890 at sky189) State DESTROY going to sleep >>>> 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE >>>> with: 503 >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: >>>> NORMAL_TEMPORARY_FAILURE >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 >>>> (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 >>>> (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/internal/hassan at a.b.c.d [BREAK] >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >>>> (sofia/internal/hassan at a.b.c.d) State REPORTING >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: >>>> NORMAL_TEMPORARY_FAILURE >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >>>> (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 >>>> (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send >>>> signal sofia/internal/hassan at a.b.c.d [BREAK] >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 >>>> (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities >>>> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session >>>> 10 (sofia/internal/hassan at a.b.c.d) Ended >>>> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close >>>> Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 >>>> (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN >>>> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 >>>> (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY >>>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >>>> (sofia/internal/hassan at a.b.c.d) State DESTROY >>>> 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 >>>> sofia/internal/hassan at a.b.c.d SOFIA DESTROY >>>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/internal/hassan at a.b.c.d Standard DESTROY >>>> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >>>> (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/9b47367c/attachment-0001.html From kris at kriskinc.com Thu Aug 5 06:52:37 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 5 Aug 2010 09:52:37 -0400 Subject: [Freeswitch-users] sips dial string question Message-ID: Heh... I love that I got a verbal response in person at almost exactly this same time yesterday. Cluecon rocks!!! -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces at lists.freeswitch.org < freeswitch-users-bounces at lists.freeswitch.org> *To*: FreeSWITCH Users Help *Sent*: Thu Aug 05 09:40:58 2010 *Subject*: Re: [Freeswitch-users] sips dial string question It does but their might be cases that aren't handled properly. Since Aastra is the only phone I know of that does it like that and testing was limited. /b On Aug 4, 2010, at 10:24 AM, Kristian Kielhofner wrote: Does freeswitch support the sips uri scheme? -- Kristian Kielhofner http://blog.krisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/35a53f34/attachment.html From peter.olsson at visionutveckling.se Thu Aug 5 07:06:23 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 5 Aug 2010 16:06:23 +0200 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: , Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC0580F1@cooper> Have you reloaded the sip profile? I'm using this syntax for lots of gateways, and they never tries to register. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Nyamul Hassan [mnhassan at usa.net] Skickat: den 5 augusti 2010 15:47 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Help on reading the log better I've set the "registration" to false, but whenever FS attempts to call the gateway, it tries to send the call using the user/pass given. For some switches, that also messes up the CallerID. I'm using the "non-gateway" format for now, and it works. Thanks to everyone for their help. Regards HASSAN On Thu, Aug 5, 2010 at 16:16, Steven Ayre > wrote: As already mentioned, one is required. But it's only used if the gateway gives a 401 Not Authorized response. If you don't need authentication, just make one up. -Steve On 5 August 2010 10:09, Nyamul Hassan > wrote: Thank you for the detailed response Steven. You were right, I was comparing my original config to the new one, and found that I did need to use it as you have said, sofia/gateway/sky189/$1 That being said, I was also unable to make a gateway without using a "username" / "password". My providers don't need any username password, and use our IP address for authentication. We can use the "sofia/internal/$1@" format though. Regards HASSAN On Thu, Aug 5, 2010 at 14:46, Steven Ayre > wrote: The call is getting 503 Service Unavailable. 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [calling][0] 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [terminated][503] This either comes from the remote host, or from within the SIP stack. An example of when I've seen Sofia generate 503 errors is when DNS lookups were failing. I'm guessing this is what is happening in your case as you're dialing "sofia/external/1234567890 at sky189". sky189 isn't a domain name, so it won't resolve to anything for sofia to dial (unless you have a very unusual setup). My guess is what you're actually trying to do is call through a gateway named sky189. The syntax for that is different: "sofia/gateway/sky189/1234567890" If that's not it then more debugging information should let you track down the error. If you enable siptrace for the profile you are calling out on, then the log will include the SIP messages sent. This will let you see if the 503 is a reply from the destination. If it isn't then it's probably within the sofia SIP stack. You can enable debugging for that stack to see why the error occurs (this would show the dns error for example). The wiki shows how to enable this debugging: http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP -Steve On 4 August 2010 18:52, Nyamul Hassan > wrote: Hi, I've finished the book, and was able to send a call through to another switch that we use. There was an initial codec mismatch (G729), but after reading the debug logs (fs_cli /log 7), it was identified and after fixing that, the call went through just fine. Today, I was trying some more changes on the default config, and the call will not go through. While that is not a problem as I can always go back to the default conf, what was bothering me is that I could not find out the right cause by reading the logs. The call also did not hit the other switch. Perhaps I am not reading something right. Can someone please help me identify what part of the logs below are showing where the calls are failing? The log is given below my signature. I changed my IPs and dialed number. Sorry for the inconvenience. Regards HASSAN 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected by acl "domains". Falling back to Digest auth. 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected by acl "domains". Falling back to Digest auth. 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/hassan at a.b.c.d) State NEW 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on nat.auto 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel sofia/internal/hassan at a.b.c.d entering state [received][100] 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: v=0 o=- 9 2 IN IP4 192.168.254.10 s=CounterPath Bria c=IN IP4 192.168.254.10 t=0 0 m=audio 63242 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf send/recv payload to 101 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/hassan at a.b.c.d) State INIT 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 sofia/internal/hassan at a.b.c.d SOFIA INIT 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/hassan at a.b.c.d) State INIT going to sleep 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/hassan at a.b.c.d) State ROUTING 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 sofia/internal/hassan at a.b.c.d SOFIA ROUTING 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 sofia/internal/hassan at a.b.c.d Standard ROUTING 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing hassan->0111234567890 in context sky189 Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] continue=false Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false Dialplan: sofia/internal/hassan at a.b.c.d Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/hassan at a.b.c.d Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/hassan at a.b.c.d Action bridge(sofia/external/1234567890 at sky189) 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/hassan at a.b.c.d) State EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 sofia/internal/hassan at a.b.c.d SOFIA EXECUTE 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 sofia/internal/hassan at a.b.c.d Standard EXECUTE EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] EXECUTE sofia/internal/hassan at a.b.c.d bridge(sofia/external/1234567890 at sky189) 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_INIT 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1234567890 at sky189) State INIT 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 sofia/external/1234567890 at sky189 SOFIA INIT 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1234567890 at sky189) State INIT going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1234567890 at sky189) State ROUTING 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 sofia/external/1234567890 at sky189 SOFIA ROUTING 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1234567890 at sky189) State ROUTING going to sleep 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 (sofia/external/1234567890 at sky189) State CONSUME_MEDIA 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [calling][0] 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel sofia/external/1234567890 at sky189 entering state [terminated][503] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal sofia/external/1234567890 at sky189 [KILL] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1234567890 at sky189) State HANGUP 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from the other leg 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel sofia/external/1234567890 at sky189 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 sofia/external/1234567890 at sky189 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/external/1234567890 at sky189) State HANGUP going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> CS_REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 (sofia/external/1234567890 at sky189) State REPORTING 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 sofia/external/1234567890 at sky189 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 (sofia/external/1234567890 at sky189) State REPORTING going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> CS_DESTROY 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/1234567890 at sky189 [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal sofia/internal/hassan at a.b.c.d [KILL] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 sofia/internal/hassan at a.b.c.d skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/hassan at a.b.c.d) State HANGUP 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the other leg 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel sofia/internal/hassan at a.b.c.d hanging up, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 (sofia/external/1234567890 at sky189) Locked, Waiting on external entities 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 (sofia/external/1234567890 at sky189) Ended 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/1234567890 at sky189 [CS_DESTROY] 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 (sofia/external/1234567890 at sky189) State DESTROY 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 sofia/external/1234567890 at sky189 SOFIA DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 sofia/external/1234567890 at sky189 Standard DESTROY 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 (sofia/external/1234567890 at sky189) State DESTROY going to sleep 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 503 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/hassan at a.b.c.d) State REPORTING 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/hassan at a.b.c.d [BREAK] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 (sofia/internal/hassan at a.b.c.d) Ended 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/hassan at a.b.c.d) State DESTROY 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 sofia/internal/hassan at a.b.c.d SOFIA DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 sofia/internal/hassan at a.b.c.d Standard DESTROY 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c5ac2e032932114784393! From brian at freeswitch.org Thu Aug 5 07:24:37 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Aug 2010 09:24:37 -0500 Subject: [Freeswitch-users] FS HA-cluster In-Reply-To: References: <85178C87-B40E-4173-A1F8-3606A9460307@gmail.com> <17A5BD02DF55446E9F06D8C8447DDABE@MOBILEE1705> <5651A30B6D2E40BE968F348C613CD629@MOBILEE1705> Message-ID: <1FD7AFBA-C0EA-4E9E-8BFB-AD065AB041DA@freeswitch.org> It might just work... why not try it. /b On Aug 5, 2010, at 6:34 AM, Henry Huang wrote: > Madovsky: > > I think the track feature only works under the same IP. Under HA mode, the FS server will be using the exact same IP addresses. That's why I said if I have all the FS servers listening on the same IP behind a LVS load balancer, I wonder if it works. From mnhassan at usa.net Thu Aug 5 07:55:51 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Thu, 5 Aug 2010 20:55:51 +0600 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57DC0580F1@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C57DC0580F1@cooper> Message-ID: I have certainly reloaded the profile. Did a full FS restart too. The problem is not about registration. Coz, FS does not register at all when "registration" is set to "false". But, when FS tries to send the call, it appends the user/pass in the call setup message. For example, we have an asterisk box doing SS7 to a telco. We just added the FS as a "peer" and it shows up in the "sip show peers" output. But, when sending calls from FS, Asterisk says cannot authenticate "user/pass@". Regards HASSAN On Thu, Aug 5, 2010 at 20:06, Peter Olsson wrote: > Have you reloaded the sip profile? I'm using this syntax for lots of > gateways, and they never tries to register. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] för Nyamul Hassan [ > mnhassan at usa.net] > Skickat: den 5 augusti 2010 15:47 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Help on reading the log better > > I've set the "registration" to false, but whenever FS attempts to call the > gateway, it tries to send the call using the user/pass given. For some > switches, that also messes up the CallerID. > > I'm using the "non-gateway" format for now, and it works. > > Thanks to everyone for their help. > > Regards > HASSAN > > > > On Thu, Aug 5, 2010 at 16:16, Steven Ayre steveayre at gmail.com>> wrote: > As already mentioned, one is required. > > But it's only used if the gateway gives a 401 Not Authorized response. If > you don't need authentication, just make one up. > > -Steve > > > > > On 5 August 2010 10:09, Nyamul Hassan mnhassan at usa.net>> wrote: > Thank you for the detailed response Steven. You were right, I was > comparing my original config to the new one, and found that I did need to > use it as you have said, sofia/gateway/sky189/$1 > > That being said, I was also unable to make a gateway without using a > "username" / "password". My providers don't need any username password, and > use our IP address for authentication. We can use the "sofia/internal/$1@ of provider>" format though. > > Regards > HASSAN > > > > On Thu, Aug 5, 2010 at 14:46, Steven Ayre steveayre at gmail.com>> wrote: > The call is getting 503 Service Unavailable. > > 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel > sofia/external/1234567890 at sky189 entering state [calling][0] > 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel > sofia/external/1234567890 at sky189 entering state [terminated][503] > > This either comes from the remote host, or from within the SIP stack. > > An example of when I've seen Sofia generate 503 errors is when DNS lookups > were failing. I'm guessing this is what is happening in your case as you're > dialing "sofia/external/1234567890 at sky189". > > sky189 isn't a domain name, so it won't resolve to anything for sofia to > dial (unless you have a very unusual setup). My guess is what you're > actually trying to do is call through a gateway named sky189. The syntax for > that is different: "sofia/gateway/sky189/1234567890" > > If that's not it then more debugging information should let you track down > the error. > If you enable siptrace for the profile you are calling out on, then the log > will include the SIP messages sent. This will let you see if the 503 is a > reply from the destination. > If it isn't then it's probably within the sofia SIP stack. You can enable > debugging for that stack to see why the error occurs (this would show the > dns error for example). The wiki shows how to enable this debugging: > http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP > > -Steve > > > > On 4 August 2010 18:52, Nyamul Hassan mnhassan at usa.net>> wrote: > Hi, > > I've finished the book, and was able to send a call through to another > switch that we use. There was an initial codec mismatch (G729), but after > reading the debug logs (fs_cli /log 7), it was identified and after fixing > that, the call went through just fine. > > Today, I was trying some more changes on the default config, and the call > will not go through. While that is not a problem as I can always go back to > the default conf, what was bothering me is that I could not find out the > right cause by reading the logs. The call also did not hit the other > switch. Perhaps I am not reading something right. Can someone please help > me identify what part of the logs below are showing where the calls are > failing? > > The log is given below my signature. I changed my IPs and dialed number. > Sorry for the inconvenience. > > Regards > HASSAN > > > 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected > by acl "domains". Falling back to Digest auth. > 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected > by acl "domains". Falling back to Digest auth. > 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel > sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] > 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW > 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 > (sofia/internal/hassan at a.b.c.d) State NEW > 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on > nat.auto > 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel > sofia/internal/hassan at a.b.c.d entering state [received][100] > 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: > v=0 > o=- 9 2 IN IP4 192.168.254.10 > s=CounterPath Bria > c=IN IP4 192.168.254.10 > t=0 0 > m=audio 63242 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=yes > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 > a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 > a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 > > 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare > [G729:18:8000:20]/[G729:18:8000:20] > 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec > sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples > 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf > send/recv payload to 101 > 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 > (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/hassan at a.b.c.d) State INIT > 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 > sofia/internal/hassan at a.b.c.d SOFIA INIT > 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 > (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/hassan at a.b.c.d) State INIT going to sleep > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING > 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 > (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/hassan at a.b.c.d) State ROUTING > 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 > sofia/internal/hassan at a.b.c.d SOFIA ROUTING > 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 > sofia/internal/hassan at a.b.c.d Standard ROUTING > 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing > hassan->0111234567890 in context sky189 > Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] > continue=false > Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] > destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false > Dialplan: sofia/internal/hassan at a.b.c.d Action > set(effective_caller_id_number=${outbound_caller_id_number}) > Dialplan: sofia/internal/hassan at a.b.c.d Action > set(effective_caller_id_name=${outbound_caller_id_name}) > Dialplan: sofia/internal/hassan at a.b.c.d Action > bridge(sofia/external/1234567890 at sky189) > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 > (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/hassan at a.b.c.d) State EXECUTE > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 > sofia/internal/hassan at a.b.c.d SOFIA EXECUTE > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 > sofia/internal/hassan at a.b.c.d Standard EXECUTE > EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) > 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 > sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] > EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) > 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 > sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] > EXECUTE sofia/internal/hassan at a.b.c.dbridge(sofia/external/1234567890 at sky189 > ) > 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel > sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 > (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_INIT > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 > (sofia/external/1234567890 at sky189) State INIT > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 > sofia/external/1234567890 at sky189 SOFIA INIT > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 > (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 > (sofia/external/1234567890 at sky189) State INIT going to sleep > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING > 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 > (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 > (sofia/external/1234567890 at sky189) State ROUTING > 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 > sofia/external/1234567890 at sky189 SOFIA ROUTING > 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 > (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 > (sofia/external/1234567890 at sky189) State ROUTING going to sleep > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 > (sofia/external/1234567890 at sky189) State CONSUME_MEDIA > 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 > (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep > 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel > sofia/external/1234567890 at sky189 entering state [calling][0] > 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel > sofia/external/1234567890 at sky189 entering state [terminated][503] > 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 > (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP > 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup > sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] > [NORMAL_TEMPORARY_FAILURE] > 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal > sofia/external/1234567890 at sky189 [KILL] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 > (sofia/external/1234567890 at sky189) State HANGUP > 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 > sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from > the other leg > 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel > sofia/external/1234567890 at sky189 hanging up, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 > sofia/external/1234567890 at sky189 Standard HANGUP, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 > (sofia/external/1234567890 at sky189) State HANGUP going to sleep > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 > (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> CS_REPORTING > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 > (sofia/external/1234567890 at sky189) State REPORTING > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 > sofia/external/1234567890 at sky189 Standard REPORTING, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 > (sofia/external/1234567890 at sky189) State REPORTING going to sleep > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 > (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> CS_DESTROY > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/1234567890 at sky189 [BREAK] > 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate > Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] > 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. > Cause: NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 > (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP > 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup > sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal > sofia/internal/hassan at a.b.c.d [KILL] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 > sofia/internal/hassan at a.b.c.d skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/hassan at a.b.c.d) State HANGUP > 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 > sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the > other leg > 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/hassan at a.b.c.d hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 > (sofia/external/1234567890 at sky189) Locked, Waiting on external entities > 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 > (sofia/external/1234567890 at sky189) Ended > 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/1234567890 at sky189 [CS_DESTROY] > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 > (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 > (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/1234567890 at sky189) State DESTROY > 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 > sofia/external/1234567890 at sky189 SOFIA DESTROY > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 > sofia/external/1234567890 at sky189 Standard DESTROY > 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/1234567890 at sky189) State DESTROY going to sleep > 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE > with: 503 > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/hassan at a.b.c.d) State REPORTING > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: > NORMAL_TEMPORARY_FAILURE > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/hassan at a.b.c.d [BREAK] > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 > (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities > 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 > (sofia/internal/hassan at a.b.c.d) Ended > 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 > (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN > 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 > (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY > 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/hassan at a.b.c.d) State DESTROY > 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 > sofia/internal/hassan at a.b.c.d SOFIA DESTROY > 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/hassan at a.b.c.d Standard DESTROY > 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > !DSPAM:4c5ac2e032932114784393! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/c2e01542/attachment-0001.html From msc at freeswitch.org Thu Aug 5 08:21:55 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Aug 2010 10:21:55 -0500 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C57DC0580F1@cooper> Message-ID: On Thu, Aug 5, 2010 at 9:55 AM, Nyamul Hassan wrote: > I have certainly reloaded the profile. Did a full FS restart too. The > problem is not about registration. Coz, FS does not register at all when > "registration" is set to "false". > > But, when FS tries to send the call, it appends the user/pass in the call > setup message. For example, we have an asterisk box doing SS7 to a telco. > We just added the FS as a "peer" and it shows up in the "sip show peers" > output. But, when sending calls from FS, Asterisk says cannot authenticate > "user/pass@". > Sounds like you found the problem: Authentication. It looks like Ast is trying to auth the call and the gateway is responding with whatever you put in for the username and password. Were you not expecting Asterisk to challenge the calls? (Don't forget: there is a difference between registration and authentication. Buy the FreeSWITCH book and read chapter four. :P ) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/e842816a/attachment.html From steveayre at gmail.com Thu Aug 5 08:26:23 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 5 Aug 2010 16:26:23 +0100 Subject: [Freeswitch-users] Help on reading the log better In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C57DC0580F1@cooper> Message-ID: Try configuring your gateway like this: The gateway has benefits over sending direct to a IP - FS monitors the gateway and stops sending traffic if the gateway is offline, resulting in your calls failing quickly (and possibly trying another route) rather than calls starting to the offline gateway but then timing out. -Steve On 5 August 2010 15:55, Nyamul Hassan wrote: > I have certainly reloaded the profile. Did a full FS restart too. The > problem is not about registration. Coz, FS does not register at all when > "registration" is set to "false". > > But, when FS tries to send the call, it appends the user/pass in the call > setup message. For example, we have an asterisk box doing SS7 to a telco. > We just added the FS as a "peer" and it shows up in the "sip show peers" > output. But, when sending calls from FS, Asterisk says cannot authenticate > "user/pass@". > > Regards > HASSAN > > > > On Thu, Aug 5, 2010 at 20:06, Peter Olsson < > peter.olsson at visionutveckling.se> wrote: > >> Have you reloaded the sip profile? I'm using this syntax for lots of >> gateways, and they never tries to register. >> >> /Peter >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >> freeswitch-users-bounces at lists.freeswitch.org] för Nyamul Hassan [ >> mnhassan at usa.net] >> Skickat: den 5 augusti 2010 15:47 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] Help on reading the log better >> >> I've set the "registration" to false, but whenever FS attempts to call the >> gateway, it tries to send the call using the user/pass given. For some >> switches, that also messes up the CallerID. >> >> I'm using the "non-gateway" format for now, and it works. >> >> Thanks to everyone for their help. >> >> Regards >> HASSAN >> >> >> >> On Thu, Aug 5, 2010 at 16:16, Steven Ayre > steveayre at gmail.com>> wrote: >> As already mentioned, one is required. >> >> But it's only used if the gateway gives a 401 Not Authorized response. If >> you don't need authentication, just make one up. >> >> -Steve >> >> >> >> >> On 5 August 2010 10:09, Nyamul Hassan > mnhassan at usa.net>> wrote: >> Thank you for the detailed response Steven. You were right, I was >> comparing my original config to the new one, and found that I did need to >> use it as you have said, sofia/gateway/sky189/$1 >> >> That being said, I was also unable to make a gateway without using a >> "username" / "password". My providers don't need any username password, and >> use our IP address for authentication. We can use the "sofia/internal/$1@> of provider>" format though. >> >> Regards >> HASSAN >> >> >> >> On Thu, Aug 5, 2010 at 14:46, Steven Ayre > steveayre at gmail.com>> wrote: >> The call is getting 503 Service Unavailable. >> >> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [calling][0] >> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [terminated][503] >> >> This either comes from the remote host, or from within the SIP stack. >> >> An example of when I've seen Sofia generate 503 errors is when DNS lookups >> were failing. I'm guessing this is what is happening in your case as you're >> dialing "sofia/external/1234567890 at sky189". >> >> sky189 isn't a domain name, so it won't resolve to anything for sofia to >> dial (unless you have a very unusual setup). My guess is what you're >> actually trying to do is call through a gateway named sky189. The syntax for >> that is different: "sofia/gateway/sky189/1234567890" >> >> If that's not it then more debugging information should let you track down >> the error. >> If you enable siptrace for the profile you are calling out on, then the >> log will include the SIP messages sent. This will let you see if the 503 is >> a reply from the destination. >> If it isn't then it's probably within the sofia SIP stack. You can enable >> debugging for that stack to see why the error occurs (this would show the >> dns error for example). The wiki shows how to enable this debugging: >> http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP >> >> -Steve >> >> >> >> On 4 August 2010 18:52, Nyamul Hassan > mnhassan at usa.net>> wrote: >> Hi, >> >> I've finished the book, and was able to send a call through to another >> switch that we use. There was an initial codec mismatch (G729), but after >> reading the debug logs (fs_cli /log 7), it was identified and after fixing >> that, the call went through just fine. >> >> Today, I was trying some more changes on the default config, and the call >> will not go through. While that is not a problem as I can always go back to >> the default conf, what was bothering me is that I could not find out the >> right cause by reading the logs. The call also did not hit the other >> switch. Perhaps I am not reading something right. Can someone please help >> me identify what part of the logs below are showing where the calls are >> failing? >> >> The log is given below my signature. I changed my IPs and dialed number. >> Sorry for the inconvenience. >> >> Regards >> HASSAN >> >> >> 2010-08-04 13:06:47.459004 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected >> by acl "domains". Falling back to Digest auth. >> 2010-08-04 13:06:48.853933 [DEBUG] sofia.c:6000 IP 192.168.254.10 Rejected >> by acl "domains". Falling back to Digest auth. >> 2010-08-04 13:06:48.855161 [NOTICE] switch_channel.c:779 New Channel >> sofia/internal/hassan at a.b.c.d [3b01cb99-20dd-4a9d-8084-2f2e16a5aee6] >> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_NEW >> 2010-08-04 13:06:48.855161 [DEBUG] switch_core_state_machine.c:320 >> (sofia/internal/hassan at a.b.c.d) State NEW >> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:6823 Setting NAT mode based on >> nat.auto >> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4318 Channel >> sofia/internal/hassan at a.b.c.d entering state [received][100] >> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4329 Remote SDP: >> v=0 >> o=- 9 2 IN IP4 192.168.254.10 >> s=CounterPath Bria >> c=IN IP4 192.168.254.10 >> t=0 0 >> m=audio 63242 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=yes >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=alt:1 2 : wsPFY2UY ge45TD7O a1.b1.c1.d1 63242 >> a=alt:2 1 : mCzsWRNJ nH/GgDGG 192.168.254.10 63242 >> a=x-rtp-session-id:C59D45BA9189497CB4C041AC656702E2 >> >> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3845 Audio Codec Compare >> [G729:18:8000:20]/[G729:18:8000:20] >> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:2442 Set Codec >> sofia/internal/hassan at a.b.c.d G729/8000 20 ms 160 samples >> 2010-08-04 13:06:48.856895 [DEBUG] sofia_glue.c:3941 Set 2833 dtmf >> send/recv payload to 101 >> 2010-08-04 13:06:48.856895 [DEBUG] sofia.c:4476 >> (sofia/internal/hassan at a.b.c.d) State Change CS_NEW -> CS_INIT >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_INIT >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/hassan at a.b.c.d) State INIT >> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:83 >> sofia/internal/hassan at a.b.c.d SOFIA INIT >> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:119 >> (sofia/internal/hassan at a.b.c.d) State Change CS_INIT -> CS_ROUTING >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/hassan at a.b.c.d) State INIT going to sleep >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_ROUTING >> 2010-08-04 13:06:48.856895 [DEBUG] switch_channel.c:1512 >> (sofia/internal/hassan at a.b.c.d) Callstate Change DOWN -> RINGING >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/hassan at a.b.c.d) State ROUTING >> 2010-08-04 13:06:48.856895 [DEBUG] mod_sofia.c:142 >> sofia/internal/hassan at a.b.c.d SOFIA ROUTING >> 2010-08-04 13:06:48.856895 [DEBUG] switch_core_state_machine.c:77 >> sofia/internal/hassan at a.b.c.d Standard ROUTING >> 2010-08-04 13:06:48.856895 [INFO] mod_dialplan_xml.c:331 Processing >> hassan->0111234567890 in context sky189 >> Dialplan: sofia/internal/hassan at a.b.c.d parsing [sky189->dial_sky189] >> continue=false >> Dialplan: sofia/internal/hassan at a.b.c.d Regex (PASS) [dial_sky189] >> destination_number(0111234567890) =~ /^011(\d+)$/ break=on-false >> Dialplan: sofia/internal/hassan at a.b.c.d Action >> set(effective_caller_id_number=${outbound_caller_id_number}) >> Dialplan: sofia/internal/hassan at a.b.c.d Action >> set(effective_caller_id_name=${outbound_caller_id_name}) >> Dialplan: sofia/internal/hassan at a.b.c.d Action >> bridge(sofia/external/1234567890 at sky189) >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:119 >> (sofia/internal/hassan at a.b.c.d) State Change CS_ROUTING -> CS_EXECUTE >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/hassan at a.b.c.d) State ROUTING going to sleep >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_EXECUTE >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/hassan at a.b.c.d) State EXECUTE >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:235 >> sofia/internal/hassan at a.b.c.d SOFIA EXECUTE >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:157 >> sofia/internal/hassan at a.b.c.d Standard EXECUTE >> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_number=) >> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_number]=[UNDEF] >> EXECUTE sofia/internal/hassan at a.b.c.d set(effective_caller_id_name=) >> 2010-08-04 13:06:48.858888 [DEBUG] mod_dptools.c:854 >> sofia/internal/hassan at a.b.c.d SET [effective_caller_id_name]=[UNDEF] >> EXECUTE sofia/internal/hassan at a.b.c.dbridge(sofia/external/1234567890 at sky189 >> ) >> 2010-08-04 13:06:48.858888 [NOTICE] switch_channel.c:779 New Channel >> sofia/external/1234567890 at sky189 [cfbd2d4f-87fc-4c59-b320-2290a5a076b3] >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:3892 >> (sofia/external/1234567890 at sky189) State Change CS_NEW -> CS_INIT >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_INIT >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >> (sofia/external/1234567890 at sky189) State INIT >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:83 >> sofia/external/1234567890 at sky189 SOFIA INIT >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:119 >> (sofia/external/1234567890 at sky189) State Change CS_INIT -> CS_ROUTING >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:338 >> (sofia/external/1234567890 at sky189) State INIT going to sleep >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_ROUTING >> 2010-08-04 13:06:48.858888 [DEBUG] switch_channel.c:1512 >> (sofia/external/1234567890 at sky189) Callstate Change DOWN -> RINGING >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >> (sofia/external/1234567890 at sky189) State ROUTING >> 2010-08-04 13:06:48.858888 [DEBUG] mod_sofia.c:142 >> sofia/external/1234567890 at sky189 SOFIA ROUTING >> 2010-08-04 13:06:48.858888 [DEBUG] switch_ivr_originate.c:66 >> (sofia/external/1234567890 at sky189) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:341 >> (sofia/external/1234567890 at sky189) State ROUTING going to sleep >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_CONSUME_MEDIA >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA >> 2010-08-04 13:06:48.858888 [DEBUG] switch_core_state_machine.c:360 >> (sofia/external/1234567890 at sky189) State CONSUME_MEDIA going to sleep >> 2010-08-04 13:06:48.858888 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [calling][0] >> 2010-08-04 13:06:48.861065 [DEBUG] sofia.c:4318 Channel >> sofia/external/1234567890 at sky189 entering state [terminated][503] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >> (sofia/external/1234567890 at sky189) Callstate Change RINGING -> HANGUP >> 2010-08-04 13:06:48.861065 [NOTICE] sofia.c:4932 Hangup >> sofia/external/1234567890 at sky189 [CS_CONSUME_MEDIA] >> [NORMAL_TEMPORARY_FAILURE] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >> sofia/external/1234567890 at sky189 [KILL] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_HANGUP >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >> (sofia/external/1234567890 at sky189) State HANGUP >> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >> sofia/external/1234567890 at sky189 Overriding SIP cause 503 with 503 from >> the other leg >> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >> sofia/external/1234567890 at sky189 hanging up, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:46 >> sofia/external/1234567890 at sky189 Standard HANGUP, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >> (sofia/external/1234567890 at sky189) State HANGUP going to sleep >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:333 >> (sofia/external/1234567890 at sky189) State Change CS_HANGUP -> CS_REPORTING >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/1234567890 at sky189) Running State Change CS_REPORTING >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >> (sofia/external/1234567890 at sky189) State REPORTING >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:53 >> sofia/external/1234567890 at sky189 Standard REPORTING, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:595 >> (sofia/external/1234567890 at sky189) State REPORTING going to sleep >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:327 >> (sofia/external/1234567890 at sky189) State Change CS_REPORTING -> >> CS_DESTROY >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/external/1234567890 at sky189 [BREAK] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_ivr_originate.c:3431 Originate >> Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] >> 2010-08-04 13:06:48.861065 [INFO] mod_dptools.c:2393 Originate Failed. >> Cause: NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2309 >> (sofia/internal/hassan at a.b.c.d) Callstate Change RINGING -> HANGUP >> 2010-08-04 13:06:48.861065 [NOTICE] mod_dptools.c:2456 Hangup >> sofia/internal/hassan at a.b.c.d [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_channel.c:2325 Send signal >> sofia/internal/hassan at a.b.c.d [KILL] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1905 >> sofia/internal/hassan at a.b.c.d skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:348 >> (sofia/internal/hassan at a.b.c.d) State EXECUTE going to sleep >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_HANGUP >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_state_machine.c:535 >> (sofia/internal/hassan at a.b.c.d) State HANGUP >> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:447 >> sofia/internal/hassan at a.b.c.d Overriding SIP cause 503 with 503 from the >> other leg >> 2010-08-04 13:06:48.861065 [DEBUG] mod_sofia.c:453 Channel >> sofia/internal/hassan at a.b.c.d hanging up, cause: NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.861065 [DEBUG] switch_core_session.c:1202 Session 11 >> (sofia/external/1234567890 at sky189) Locked, Waiting on external entities >> 2010-08-04 13:06:48.861065 [NOTICE] switch_core_session.c:1220 Session 11 >> (sofia/external/1234567890 at sky189) Ended >> 2010-08-04 13:06:48.862912 [NOTICE] switch_core_session.c:1222 Close >> Channel sofia/external/1234567890 at sky189 [CS_DESTROY] >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:427 >> (sofia/external/1234567890 at sky189) Callstate Change HANGUP -> DOWN >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:430 >> (sofia/external/1234567890 at sky189) Running State Change CS_DESTROY >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/1234567890 at sky189) State DESTROY >> 2010-08-04 13:06:48.862912 [DEBUG] mod_sofia.c:358 >> sofia/external/1234567890 at sky189 SOFIA DESTROY >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:60 >> sofia/external/1234567890 at sky189 Standard DESTROY >> 2010-08-04 13:06:48.862912 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/1234567890 at sky189) State DESTROY going to sleep >> 2010-08-04 13:06:48.864893 [DEBUG] mod_sofia.c:515 Responding to INVITE >> with: 503 >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/hassan at a.b.c.d Standard HANGUP, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:535 >> (sofia/internal/hassan at a.b.c.d) State HANGUP going to sleep >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/hassan at a.b.c.d) State Change CS_HANGUP -> CS_REPORTING >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_REPORTING >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >> (sofia/internal/hassan at a.b.c.d) State REPORTING >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/hassan at a.b.c.d Standard REPORTING, cause: >> NORMAL_TEMPORARY_FAILURE >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:595 >> (sofia/internal/hassan at a.b.c.d) State REPORTING going to sleep >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/hassan at a.b.c.d) State Change CS_REPORTING -> CS_DESTROY >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/hassan at a.b.c.d [BREAK] >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_session.c:1202 Session 10 >> (sofia/internal/hassan at a.b.c.d) Locked, Waiting on external entities >> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1220 Session 10 >> (sofia/internal/hassan at a.b.c.d) Ended >> 2010-08-04 13:06:48.864893 [NOTICE] switch_core_session.c:1222 Close >> Channel sofia/internal/hassan at a.b.c.d [CS_DESTROY] >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:427 >> (sofia/internal/hassan at a.b.c.d) Callstate Change HANGUP -> DOWN >> 2010-08-04 13:06:48.864893 [DEBUG] switch_core_state_machine.c:430 >> (sofia/internal/hassan at a.b.c.d) Running State Change CS_DESTROY >> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/hassan at a.b.c.d) State DESTROY >> 2010-08-04 13:06:48.866921 [DEBUG] mod_sofia.c:358 >> sofia/internal/hassan at a.b.c.d SOFIA DESTROY >> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/hassan at a.b.c.d Standard DESTROY >> 2010-08-04 13:06:48.866921 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/hassan at a.b.c.d) State DESTROY going to sleep >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org> FreeSWITCH-users at lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> !DSPAM:4c5ac2e032932114784393! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/8c55ae7a/attachment-0001.html From egable+freeswitch at gmail.com Thu Aug 5 08:37:12 2010 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Thu, 5 Aug 2010 11:37:12 -0400 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: My most recent testing as of about two weeks ago showed a 25% performance drop between the old SVN version I was running and the new GIT version. Initially, it seemed closer to your reported performance drop, but after moving the database to a ramdisk, it went to 25%. I am unsure of whether my initial testing was using the db on a ramdisk, so the drop could be higher if my previous testing did not have the db on a ramdisk. 2010/8/3 Juan Antonio Iba?ez Santorum : > No one experiment this same issue? > > I also noticed a high cpu consume on call hangup. > > Regards > > 2010/8/2 Juan Antonio Iba?ez Santorum >> >> Hello! >> >> I was been some test using one FS tarball version downloaded some weeks >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems with >> odbc connections managemens. Now I'm using a git version (FreeSWITCH Version >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve this >> problem but I noticed it has a high cpu load comparing with svn version. >> While I could manage more than 200 calls with a 75% CPU load into a dual >> core server using svn version, now, 50 calls consume this 75% cpu. I can see >> same modules are loaded (except new hash module needed in git version) and >> same scenario is used. I am not be able to find why now it uses more cpu >> than before. Any idea? >> Each testing call is a simple bridge to an external sip provider. >> >> Regards > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From jmesquita at freeswitch.org Thu Aug 5 08:56:58 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 5 Aug 2010 12:56:58 -0300 Subject: [Freeswitch-users] Freeswitch Implementation In-Reply-To: References: Message-ID: Have you tried consulting at freeswitch.org? FreeSWITCH? Solutions provides all sorts of consulting on FreeSWITCH? and it's team is composed by most (if not all) FS developers. Regards, Jo?o Mesquita On Wed, Aug 4, 2010 at 2:52 PM, Stephen Cattaneo wrote: > Is there anyone in the US that would be interested in doing a basic > Freeswitch implementation for us? Please contact me at stephen at fitnyc.edu > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/e57f35c5/attachment.html From stephen at stephenjc.com Thu Aug 5 09:01:18 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Thu, 5 Aug 2010 12:01:18 -0400 Subject: [Freeswitch-users] Freeswitch Implementation In-Reply-To: References: Message-ID: yes i have. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print 2010/8/5 Jo?o Mesquita : > Have you tried consulting at freeswitch.org? > FreeSWITCH? Solutions provides all sorts of consulting on FreeSWITCH? and > it's team is composed by most (if not all) FS developers. > Regards, > Jo?o Mesquita > > > On Wed, Aug 4, 2010 at 2:52 PM, Stephen Cattaneo > wrote: >> >> Is there anyone in the US that would be interested in doing a basic >> Freeswitch implementation for us? Please contact me at stephen at fitnyc.edu >> >> >> Thanks, >> Stephen C >> -All of my email addresses go to the same place >> -Save Paper, think before you print >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ovvenkatesan at gmail.com Thu Aug 5 09:12:34 2010 From: ovvenkatesan at gmail.com (ovvenkat) Date: Thu, 5 Aug 2010 21:42:34 +0530 Subject: [Freeswitch-users] Error occurring while compiling ASR Grammar Message-ID: hi to all, I have followed the this wiki page http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts As said in the wiki page, I have created a file called *directory.sent* inside the director * /usr/local/freeswitch/grammar*. After that, I have issued the following command to compile (my current directory is *director /usr/local/freeswitch/grammar* ) *make /usr/local/freeswitch/grammar directory* ( or ) *make /usr/local/freeswitch/grammar directory.sent *I am getting the following error message make: Nothing to be done for `/usr/local/freeswitch/grammar'. make: *** No rule to make target `directory'. Stop.* * Can you anyone please correct me, where I m dong wrong? Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/9f689c34/attachment.html From juanito1982 at gmail.com Thu Aug 5 10:16:41 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Thu, 5 Aug 2010 19:16:41 +0200 Subject: [Freeswitch-users] Freeswitch Implementation In-Reply-To: References: Message-ID: Hello Stephen, What is what you need? Regards 2010/8/4 Stephen Cattaneo > Is there anyone in the US that would be interested in doing a basic > Freeswitch implementation for us? Please contact me at stephen at fitnyc.edu > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/94e02c97/attachment.html From infos at madovsky.org Thu Aug 5 10:27:00 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 13:27:00 -0400 Subject: [Freeswitch-users] On FS-HA feature: HA + Balancing ? References: <4C5A6E6C.3010905@infosecurity.ch> Message-ID: For now I use NAPTR SRV for load balancing (it's round robin but it avoids any proxy) for HA you have LVS, Pacemaker, Linix-HA, IBM cluster etc... ----- Original Message ----- From: "Fabio Pietrosanti (naif)" To: "FreeSWITCH Users Help" Sent: Thursday, August 05, 2010 3:55 AM Subject: [Freeswitch-users] On FS-HA feature: HA + Balancing ? > Hi guys, > > first of all my compliments for FS-HA features, something really > enterprise that make the life of infrastructure architect and system > administration much simpler. > > My simpler question is: > > Does FS with HA features can do "High availability AND Load Balancing" > without using third party SIP Proxy? > > If so the FS would simplify a lot most of the IP architecture as it may > be possible to just put more FS boxes into a HA+Balancing "server farm" > when you need more power. > > Is that possible? > > Fabio > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From infos at madovsky.org Thu Aug 5 10:34:22 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 13:34:22 -0400 Subject: [Freeswitch-users] On FS-HA feature: HA + Balancing ? Message-ID: <03C7CC6BD82D4150BDA3C3422E0394BA@MOBILEE1705> of course you have to use a proxy if are behind a fw ----- Original Message ----- From: "Madovsky" To: "FreeSWITCH Users Help" Sent: Thursday, August 05, 2010 1:27 PM Subject: Re: [Freeswitch-users] On FS-HA feature: HA + Balancing ? > For now I use NAPTR SRV for load balancing (it's round robin but it avoids > any proxy) > for HA you have LVS, Pacemaker, Linix-HA, IBM cluster etc... > > ----- Original Message ----- > From: "Fabio Pietrosanti (naif)" > To: "FreeSWITCH Users Help" > Sent: Thursday, August 05, 2010 3:55 AM > Subject: [Freeswitch-users] On FS-HA feature: HA + Balancing ? > > >> Hi guys, >> >> first of all my compliments for FS-HA features, something really >> enterprise that make the life of infrastructure architect and system >> administration much simpler. >> >> My simpler question is: >> >> Does FS with HA features can do "High availability AND Load Balancing" >> without using third party SIP Proxy? >> >> If so the FS would simplify a lot most of the IP architecture as it may >> be possible to just put more FS boxes into a HA+Balancing "server farm" >> when you need more power. >> >> Is that possible? >> >> Fabio >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From robust.process at gmail.com Thu Aug 5 10:41:33 2010 From: robust.process at gmail.com (Robust Process) Date: Thu, 5 Aug 2010 11:41:33 -0600 Subject: [Freeswitch-users] any disaster lurking in this over all plan? Message-ID: All, when it comes to VOIP, I was born yesterday (but I staid up all night). I was about to buy a Digium TDM404E with 4 FXO (Red) Modules and eight Aastra 6755i phones when it occurred to me I probably should ask the FreeSWITCH community if what I'm about to do is sane :] 8 endpoints (planned to be aastra 6755i) 6 people using phones (four concurrent PSTN lines busy at peak usage) (55 to 60 long distance calls at a current cost of $55 to $70/mo) I hope to create a VOIP hybrid (at first) where just the long distance calls are done via SIP. Later, I would disconnect PSTN2 and PSTN3 and use the SIP DID as the roll over group/hunt group once I have the system stabilized. ---------------------------------------------------------------------------- Inbound Calls ---------------------------------------------------------------------------- PSTN1->TDM404E| PSTN2->TDM404E|->FreeSWITCH->local ethernet network->(aastra 6755i endpoints) PSTN3->TDM404E| PSTN4->TDM404E| {will later disconnect PSTN2 and PSTN3 and use SIP DID as roll over group -and hunt for outgoing} ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- Outbound Calls ---------------------------------------------------------------------------- SIP service (one DID)<-FreeSWITCH<-local ethernet network<-(long distance calls via aastra 6755i) PSTN2->TDM404E|<-FreeSWITCH<-local ethernet network<-(local calls via aastra 6755i) PSTN3->TDM404E| PSTN4->TDM404E| ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- Ethernet Configuration ---------------------------------------------------------------------------- | -TDM404E->PSTN -DSL Modem NAT---3com Ethernet Switch---FreeSWITCH| | |_aastra 6755i--Computer (this represents eight endpoints/PCs) ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- FreeSWITCH Server ---------------------------------------------------------------------------- Dell 1400sc Service Tag: JRRXJ01 **************** Hardware **************** BIOS ver A03 CPUs Two 1000MHz, fsb 133 Mhz, L2 256kb RAM 512MB ECC Video ATI Rage XL FDD CDROM Tape Drive (Archive Python 06408-xxx) Raid Controller: American Megatrends Perc2/D2 (Now LSI) bois v1p00, firmware ver 1.01 DRAM 128MB 15 scsi ids on each channel Harddisk Space: 66GB RAID 5, 3 stripes, Stripe Size 64kb, Wide Termination (one 34,578MB Seagate ST336607LW, two 34,938MB Seagate ST336807LW ) PS/2 Keyboard PS/2 Mouse Two DB 9 pin serial ports One 25 pin lpt port 2 USB ports Serverworks One onboard Intel NIC 10/100M, MAC 00B0D0D094E8 Card Slots: Slot 4, planned to place TDM404E in this slot Slot 5, Intel NIC 10/100M Slot 6, Intel SCSI ---------------------------------------------------------------------------- So, is the above sane or am I going to have to stay up all weekend? -seriously, let me know if you see looming disaster (and it'd be useful if you told me the specific disaster you see and how to avoid it). Thanks, Orin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/62d21691/attachment-0001.html From rupa at rupa.com Thu Aug 5 10:58:51 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 5 Aug 2010 12:58:51 -0500 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: sounds like a good opportunity to try git-bisect to locate where the performance change occurs. On Thu, Aug 5, 2010 at 10:37 AM, Eliot Gable > wrote: > My most recent testing as of about two weeks ago showed a 25% > performance drop between the old SVN version I was running and the new > GIT version. Initially, it seemed closer to your reported performance > drop, but after moving the database to a ramdisk, it went to 25%. I am > unsure of whether my initial testing was using the db on a ramdisk, so > the drop could be higher if my previous testing did not have the db on > a ramdisk. > > 2010/8/3 Juan Antonio Iba?ez Santorum : > > No one experiment this same issue? > > > > I also noticed a high cpu consume on call hangup. > > > > Regards > > > > 2010/8/2 Juan Antonio Iba?ez Santorum > >> > >> Hello! > >> > >> I was been some test using one FS tarball version downloaded some weeks > >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems > with > >> odbc connections managemens. Now I'm using a git version (FreeSWITCH > Version > >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve > this > >> problem but I noticed it has a high cpu load comparing with svn version. > >> While I could manage more than 200 calls with a 75% CPU load into a dual > >> core server using svn version, now, 50 calls consume this 75% cpu. I can > see > >> same modules are loaded (except new hash module needed in git version) > and > >> same scenario is used. I am not be able to find why now it uses more cpu > >> than before. Any idea? > >> Each testing call is a simple bridge to an external sip provider. > >> > >> Regards > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Eliot Gable > > "We do not inherit the Earth from our ancestors: we borrow it from our > children." ~David Brower > > "I decided the words were too conservative for me. We're not borrowing > from our children, we're stealing from them--and it's not even > considered to be a crime." ~David Brower > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > live; not live to eat.) ~Marcus Tullius Cicero > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/c7a473f6/attachment.html From infos at madovsky.org Thu Aug 5 11:00:55 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 14:00:55 -0400 Subject: [Freeswitch-users] destination_number question Message-ID: if I call from a sip phone for ex 99999999 at mydomain and use in dialplan it fails. I need to remove the << ^ >> and it works. so I there any chars in front of the number in destination_number variable if I call from SIP phone ? Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/ef203c24/attachment.html From infos at madovsky.org Thu Aug 5 11:30:35 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 14:30:35 -0400 Subject: [Freeswitch-users] destination_number question Message-ID: <5A31D6ADDA744340AC793152F70461C0@MOBILEE1705> have some doubt can I have destination_number stacked condition like blablablaa Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Thursday, August 05, 2010 2:00 PM Subject: destination_number question if I call from a sip phone for ex 99999999 at mydomain and use in dialplan it fails. I need to remove the << ^ >> and it works. so I there any chars in front of the number in destination_number variable if I call from SIP phone ? Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/6958ad55/attachment.html From brian at freeswitch.org Thu Aug 5 11:49:51 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Aug 2010 13:49:51 -0500 Subject: [Freeswitch-users] any disaster lurking in this over all plan? In-Reply-To: References: Message-ID: <8033EF35-6BFB-4BCE-A450-D7E8D1DD6B51@freeswitch.org> The only disaster I see is the Aastra in the mix. Thats just my personal preference. /b On Aug 5, 2010, at 12:41 PM, Robust Process wrote: > disaster you see and how to avoid it From brian at freeswitch.org Thu Aug 5 11:50:05 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Aug 2010 13:50:05 -0500 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: Or stop using ubuntu. /b On Aug 5, 2010, at 12:58 PM, Rupa Schomaker wrote: > sounds like a good opportunity to try git-bisect to locate where the performance change occurs. From Nabble at slickdeals.endjunk.com Thu Aug 5 11:54:40 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Thu, 5 Aug 2010 11:54:40 -0700 (PDT) Subject: [Freeswitch-users] destination_number question In-Reply-To: References: Message-ID: <1281034480298-5377836.post@n2.nabble.com> Madovsky wrote: > > if I call from a sip phone for ex 99999999 at mydomain > > and use > > > > in dialplan it fails. I need to remove the << ^ >> You have an extra } character on your dialplan. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/destination-number-question-tp5377663p5377836.html Sent from the freeswitch-users mailing list archive at Nabble.com. From infos at madovsky.org Thu Aug 5 12:08:43 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 15:08:43 -0400 Subject: [Freeswitch-users] destination_number question References: <1281034480298-5377836.post@n2.nabble.com> Message-ID: <9491D9BA936B4037AA2834B8F8A748C8@MOBILEE1705> sorry it's a typo, not in my dialplan... Anyway I resolved it thank you. my problem now is the other thread ;) ----- Original Message ----- From: "mazilo" To: Sent: Thursday, August 05, 2010 2:54 PM Subject: Re: [Freeswitch-users] destination_number question > > > Madovsky wrote: >> >> if I call from a sip phone for ex 99999999 at mydomain >> >> and use >> >> >> >> in dialplan it fails. I need to remove the << ^ >> > You have an extra } character on your dialplan. > > ----- > don't and stop are the ONLY two 4-letter words considered offensive to > men, > but not when used together. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/destination-number-question-tp5377663p5377836.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From infos at madovsky.org Thu Aug 5 12:09:22 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 15:09:22 -0400 Subject: [Freeswitch-users] Different versions, different cpu load References: Message-ID: <471CAB6FCAAB4C63B051CC7A41A23236@MOBILEE1705> ubunto is a nice linux distrib for audio station, not servers... ----- Original Message ----- From: "Brian West" To: "FreeSWITCH Users Help" Sent: Thursday, August 05, 2010 2:50 PM Subject: Re: [Freeswitch-users] Different versions, different cpu load > Or stop using ubuntu. > > /b > > On Aug 5, 2010, at 12:58 PM, Rupa Schomaker wrote: > >> sounds like a good opportunity to try git-bisect to locate where the >> performance change occurs. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From robust.process at gmail.com Thu Aug 5 12:39:58 2010 From: robust.process at gmail.com (Robust Process) Date: Thu, 5 Aug 2010 13:39:58 -0600 Subject: [Freeswitch-users] any disaster lurking in this over all plan? Message-ID: >The only disaster I see is the Aastra in the mix. Thats just my personal preference. thanks for the response Brian, since I've never had my hands on any of the IP phones, what feature, lack of feature, quality issue, or configuration problems have you had with the Aastra's compared to other IP phones? Orin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/0812ece3/attachment.html From brian at freeswitch.org Thu Aug 5 12:50:51 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Aug 2010 14:50:51 -0500 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: <471CAB6FCAAB4C63B051CC7A41A23236@MOBILEE1705> References: <471CAB6FCAAB4C63B051CC7A41A23236@MOBILEE1705> Message-ID: <8965E4A1-70D5-48E4-851C-C80A98F67C77@freeswitch.org> Ubuntu is a common theme when someone says "OMG IT USES ALL MY CPU!" /b On Aug 5, 2010, at 2:09 PM, Madovsky wrote: > ubunto is a nice linux distrib for audio station, not servers... > > ----- Original Message ----- > From: "Brian West" > To: "FreeSWITCH Users Help" > Sent: Thursday, August 05, 2010 2:50 PM > Subject: Re: [Freeswitch-users] Different versions, different cpu load > > >> Or stop using ubuntu. >> >> /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/240498d5/attachment.html From lists at infosecurity.ch Thu Aug 5 13:00:05 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Thu, 05 Aug 2010 22:00:05 +0200 Subject: [Freeswitch-users] On FS-HA feature: HA + Balancing ? In-Reply-To: References: <4C5A6E6C.3010905@infosecurity.ch> Message-ID: <4C5B1845.2000906@infosecurity.ch> mmm i am using a SIP UA client that does not support NAPTR SRV. Would be it be possible to do an infrastructure like this: sipserver.domain.com IN A 192.168.0.2 IN A 192.168.0.3 so that the client DNS request will go in roundrobin and in theory could scale up to an important number. The client user as a proxy-server configuration sipserver.domain.com . On the two machine 192.168.0.2 and 192.168.0.3 there are a couple of FS with the HA feature turned on. That way about 50% of the user will get registered to 192.168.0.2 and 50% on 192.168.0.3 . My questions are: a) Would it possible to have both FS active at the same time with HA feature? b) How users bound on different servers communicate each other? If the userA is connected to sipserver.domain.com on server 0.2 and userB is connected to sipserver.domain.com on 0.3 (and both FS does RTP relay) how can FS on 0.2 know that userB is registered on 0.3 ? Does they need to register something like a registration status to a database? I am just wondering whether FS could scale-up horizzontally with simple roundrobin (in past i had 4 mailserver in roundrobin on the same hostname) solution with implicit balancing + high availability without introducing the additional complexity of a SIP proxy. Fabio On 05/08/10 19.27, Madovsky wrote: > For now I use NAPTR SRV for load balancing (it's round robin but it avoids > any proxy) > for HA you have LVS, Pacemaker, Linix-HA, IBM cluster etc... > > ----- Original Message ----- > From: "Fabio Pietrosanti (naif)" > To: "FreeSWITCH Users Help" > Sent: Thursday, August 05, 2010 3:55 AM > Subject: [Freeswitch-users] On FS-HA feature: HA + Balancing ? > > > >> Hi guys, >> >> first of all my compliments for FS-HA features, something really >> enterprise that make the life of infrastructure architect and system >> administration much simpler. >> >> My simpler question is: >> >> Does FS with HA features can do "High availability AND Load Balancing" >> without using third party SIP Proxy? >> >> If so the FS would simplify a lot most of the IP architecture as it may >> be possible to just put more FS boxes into a HA+Balancing "server farm" >> when you need more power. >> >> Is that possible? >> >> Fabio >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From infos at madovsky.org Thu Aug 5 16:59:09 2010 From: infos at madovsky.org (Madovsky) Date: Thu, 5 Aug 2010 19:59:09 -0400 Subject: [Freeswitch-users] nibbleBill action extension Message-ID: <8DC1A7902C3541C2AA553B7D82EA7658@MOBILEE1705> Is it possible to have different extension for lowbal and nobal action ? for ex I d like to play a different wave for a leg B and leg A Thanks F -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100805/076bde0b/attachment.html From sos at sokhapkin.dyndns.org Thu Aug 5 17:12:41 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Thu, 5 Aug 2010 20:12:41 -0400 Subject: [Freeswitch-users] nibbleBill action extension In-Reply-To: <8DC1A7902C3541C2AA553B7D82EA7658@MOBILEE1705> References: <8DC1A7902C3541C2AA553B7D82EA7658@MOBILEE1705> Message-ID: <201008052012.41958.sos@sokhapkin.dyndns.org> Nope. Take a look at transfer_call() function in mod_nibblebill.c. It does transfer of both legs to the same extension. On Thursday 05 August 2010, Madovsky wrote: > Is it possible to have different extension for lowbal and nobal action ? > > for ex I d like to play a different wave for a leg B and leg A > > Thanks > > F > From infos at madovsky.org Thu Aug 5 21:28:27 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 6 Aug 2010 00:28:27 -0400 Subject: [Freeswitch-users] nibbleBill action extension References: <8DC1A7902C3541C2AA553B7D82EA7658@MOBILEE1705> <201008052012.41958.sos@sokhapkin.dyndns.org> Message-ID: <0FEBAC693FF44D8EA5091484A10EA4ED@MOBILEE1705> ok thanks S ! ----- Original Message ----- From: "Sergey Okhapkin" To: "FreeSWITCH Users Help" Sent: Thursday, August 05, 2010 8:12 PM Subject: Re: [Freeswitch-users] nibbleBill action extension > Nope. Take a look at transfer_call() function in mod_nibblebill.c. It does > transfer of both legs to the same extension. > > On Thursday 05 August 2010, Madovsky wrote: >> Is it possible to have different extension for lowbal and nobal action ? >> >> for ex I d like to play a different wave for a leg B and leg A >> >> Thanks >> >> F >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From infos at madovsky.org Thu Aug 5 22:25:11 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 6 Aug 2010 01:25:11 -0400 Subject: [Freeswitch-users] destination_number question References: <4c5b5f21.2412960a.0a41.7661@mx.google.com> Message-ID: <3926A956DBAE4E89912B51FADA3AD9A4@MOBILEE1705> I think I need to rest, drink and smoke again.... what I neded was simply an expression like this ----- Original Message ----- From: msc at freeswitch.org To: Madovsky Sent: Thursday, August 05, 2010 9:02 PM Subject: Re: [Freeswitch-users] destination_number question You could but why? In your example the blahblah would never get executed because there is no dest num that could possibly match both conditions. What are you trying to accomplish? Also, don't forget that Darren did a great job of explaining this kind of thing in chapter 8 of the FS book. -MC - Sent from my HTC on the Now Network from Sprint! ----- Reply message ----- From: "Madovsky" Date: Thu, Aug 5, 2010 2:30 pm Subject: [Freeswitch-users] destination_number question To: have some doubt can I have destination_number stacked condition like blablablaa Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Thursday, August 05, 2010 2:00 PM Subject: destination_number question if I call from a sip phone for ex 99999999 at mydomain and use in dialplan it fails. I need to remove the << ^ >> and it works. so I there any chars in front of the number in destination_number variable if I call from SIP phone ? Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/bcd62b19/attachment.html From terrymr at gmail.com Thu Aug 5 23:14:48 2010 From: terrymr at gmail.com (Terry Moore-Read) Date: Thu, 5 Aug 2010 23:14:48 -0700 Subject: [Freeswitch-users] mod_skinny issue Message-ID: When starting freeswitch, I'm seeing the following : 2010-08-05 23:08:43.665581 [ERR] mod_skinny.c:357 SQL ERR: [DELETE FROM skinny_devices] no such table: skinny_devices 2010-08-05 23:08:43.666177 [ERR] mod_skinny.c:357 SQL ERR: [DELETE FROM skinny_lines] no such table: skinny_lines 2010-08-05 23:08:43.666761 [ERR] mod_skinny.c:357 SQL ERR: [DELETE FROM skinny_buttons] no such table: skinny_buttons 2010-08-05 23:08:43.667323 [ERR] mod_skinny.c:357 SQL ERR: [DELETE FROM skinny_active_lines] no such table: skinny_active_lines It appears from mod_skinny.c that these tables should have been created by this point if they don't exist. I'm not sure why this isn't happening. FreeSWITCH Version 1.0.head (git-b60d6b3 2010-08-05 16-07-14 -0400) using sqlite From juanito1982 at gmail.com Thu Aug 5 23:33:22 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 6 Aug 2010 08:33:22 +0200 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: <8965E4A1-70D5-48E4-851C-C80A98F67C77@freeswitch.org> References: <471CAB6FCAAB4C63B051CC7A41A23236@MOBILEE1705> <8965E4A1-70D5-48E4-851C-C80A98F67C77@freeswitch.org> Message-ID: Which distro do you sugest? Regards 2010/8/5 Brian West > Ubuntu is a common theme when someone says "OMG IT USES ALL MY CPU!" > > /b > > On Aug 5, 2010, at 2:09 PM, Madovsky wrote: > > ubunto is a nice linux distrib for audio station, not servers... > > ----- Original Message ----- > From: "Brian West" > To: "FreeSWITCH Users Help" > Sent: Thursday, August 05, 2010 2:50 PM > Subject: Re: [Freeswitch-users] Different versions, different cpu load > > > Or stop using ubuntu. > > > /b > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/7a2dc3de/attachment.html From juanito1982 at gmail.com Thu Aug 5 23:36:28 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 6 Aug 2010 08:36:28 +0200 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: I've got about 75% decrease in performarce. I am using same server and same config. DB was already in ramdisk as svn version so I don't find an explanation for it... Regards 2010/8/5 Eliot Gable > > My most recent testing as of about two weeks ago showed a 25% > performance drop between the old SVN version I was running and the new > GIT version. Initially, it seemed closer to your reported performance > drop, but after moving the database to a ramdisk, it went to 25%. I am > unsure of whether my initial testing was using the db on a ramdisk, so > the drop could be higher if my previous testing did not have the db on > a ramdisk. > > 2010/8/3 Juan Antonio Iba?ez Santorum : > > No one experiment this same issue? > > > > I also noticed a high cpu consume on call hangup. > > > > Regards > > > > 2010/8/2 Juan Antonio Iba?ez Santorum > >> > >> Hello! > >> > >> I was been some test using one FS tarball version downloaded some weeks > >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems > with > >> odbc connections managemens. Now I'm using a git version (FreeSWITCH > Version > >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve > this > >> problem but I noticed it has a high cpu load comparing with svn version. > >> While I could manage more than 200 calls with a 75% CPU load into a dual > >> core server using svn version, now, 50 calls consume this 75% cpu. I can > see > >> same modules are loaded (except new hash module needed in git version) > and > >> same scenario is used. I am not be able to find why now it uses more cpu > >> than before. Any idea? > >> Each testing call is a simple bridge to an external sip provider. > >> > >> Regards > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Eliot Gable > > "We do not inherit the Earth from our ancestors: we borrow it from our > children." ~David Brower > > "I decided the words were too conservative for me. We're not borrowing > from our children, we're stealing from them--and it's not even > considered to be a crime." ~David Brower > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > live; not live to eat.) ~Marcus Tullius Cicero > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/7d2c0506/attachment.html From juanito1982 at gmail.com Thu Aug 5 23:39:43 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 6 Aug 2010 08:39:43 +0200 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: I thinkg it is very difficult to find it without a good knowledge of FS internals. Do you know if there is any way to make a profiling which could help? Regards 2010/8/5 Rupa Schomaker > sounds like a good opportunity to try git-bisect to locate where the > performance change occurs. > > > On Thu, Aug 5, 2010 at 10:37 AM, Eliot Gable > > wrote: > >> My most recent testing as of about two weeks ago showed a 25% >> performance drop between the old SVN version I was running and the new >> GIT version. Initially, it seemed closer to your reported performance >> drop, but after moving the database to a ramdisk, it went to 25%. I am >> unsure of whether my initial testing was using the db on a ramdisk, so >> the drop could be higher if my previous testing did not have the db on >> a ramdisk. >> >> 2010/8/3 Juan Antonio Iba?ez Santorum : >> > No one experiment this same issue? >> > >> > I also noticed a high cpu consume on call hangup. >> > >> > Regards >> > >> > 2010/8/2 Juan Antonio Iba?ez Santorum >> >> >> >> Hello! >> >> >> >> I was been some test using one FS tarball version downloaded some weeks >> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems >> with >> >> odbc connections managemens. Now I'm using a git version (FreeSWITCH >> Version >> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve >> this >> >> problem but I noticed it has a high cpu load comparing with svn >> version. >> >> While I could manage more than 200 calls with a 75% CPU load into a >> dual >> >> core server using svn version, now, 50 calls consume this 75% cpu. I >> can see >> >> same modules are loaded (except new hash module needed in git version) >> and >> >> same scenario is used. I am not be able to find why now it uses more >> cpu >> >> than before. Any idea? >> >> Each testing call is a simple bridge to an external sip provider. >> >> >> >> Regards >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Eliot Gable >> >> "We do not inherit the Earth from our ancestors: we borrow it from our >> children." ~David Brower >> >> "I decided the words were too conservative for me. We're not borrowing >> from our children, we're stealing from them--and it's not even >> considered to be a crime." ~David Brower >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >> live; not live to eat.) ~Marcus Tullius Cicero >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/b37a0b39/attachment.html From dujinfang at gmail.com Fri Aug 6 01:22:10 2010 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Aug 2010 16:22:10 +0800 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: Were both of your kernel running on 1000HZ ? 2010/8/6 Juan Antonio Iba?ez Santorum : > I've got about 75% decrease in performarce. I am using same server and same > config. DB was already in ramdisk as svn version so I don't find an > explanation for it... > > Regards > > 2010/8/5 Eliot Gable >> >> My most recent testing as of about two weeks ago showed a 25% >> performance drop between the old SVN version I was running and the new >> GIT version. Initially, it seemed closer to your reported performance >> drop, but after moving the database to a ramdisk, it went to 25%. I am >> unsure of whether my initial testing was using the db on a ramdisk, so >> the drop could be higher if my previous testing did not have the db on >> a ramdisk. >> >> 2010/8/3 Juan Antonio Iba?ez Santorum : >> > No one experiment this same issue? >> > >> > I also noticed a high cpu consume on call hangup. >> > >> > Regards >> > >> > 2010/8/2 Juan Antonio Iba?ez Santorum >> >> >> >> Hello! >> >> >> >> I was been some test using one FS tarball version downloaded some weeks >> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems >> >> with >> >> odbc connections managemens. Now I'm using a git version (FreeSWITCH >> >> Version >> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve >> >> this >> >> problem but I noticed it has a high cpu load comparing with svn >> >> version. >> >> While I could manage more than 200 calls with a 75% CPU load into a >> >> dual >> >> core server using svn version, now, 50 calls consume this 75% cpu. I >> >> can see >> >> same modules are loaded (except new hash module needed in git version) >> >> and >> >> same scenario is used. I am not be able to find why now it uses more >> >> cpu >> >> than before. Any idea? >> >> Each testing call is a simple bridge to an external sip provider. >> >> >> >> Regards >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Eliot Gable >> >> "We do not inherit the Earth from our ancestors: we borrow it from our >> children." ~David Brower >> >> "I decided the words were too conservative for me. We're not borrowing >> from our children, we're stealing from them--and it's not even >> considered to be a crime." ~David Brower >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >> live; not live to eat.) ~Marcus Tullius Cicero >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From b_ball_henry at hotmail.com Fri Aug 6 02:00:38 2010 From: b_ball_henry at hotmail.com (Henry Huang) Date: Fri, 6 Aug 2010 17:00:38 +0800 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: How do I tell if the kernel is running at 1000hz 2010-8-6 ??4:29? "Seven Du" ??? Were both of your kernel running on 1000HZ ? 2010/8/6 Juan Antonio Iba?ez Santorum : > I've got about 75% decrease in performarce. I am using same server and same > config. DB was alrea... Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at list... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/f72c4795/attachment.html From juanito1982 at gmail.com Fri Aug 6 02:24:09 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 6 Aug 2010 11:24:09 +0200 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: These compariso is done into the same machine. I run SVN version, make the test, stop SVN version, run GIT version and make the test again. It runs over a "Intel(R) Core(TM)2 Duo CPU E7500 @ 2.93GHz" Regards 2010/8/6 Seven Du > Were both of your kernel running on 1000HZ ? > > 2010/8/6 Juan Antonio Iba?ez Santorum : > > I've got about 75% decrease in performarce. I am using same server and > same > > config. DB was already in ramdisk as svn version so I don't find an > > explanation for it... > > > > Regards > > > > 2010/8/5 Eliot Gable > > > >> > >> My most recent testing as of about two weeks ago showed a 25% > >> performance drop between the old SVN version I was running and the new > >> GIT version. Initially, it seemed closer to your reported performance > >> drop, but after moving the database to a ramdisk, it went to 25%. I am > >> unsure of whether my initial testing was using the db on a ramdisk, so > >> the drop could be higher if my previous testing did not have the db on > >> a ramdisk. > >> > >> 2010/8/3 Juan Antonio Iba?ez Santorum : > >> > No one experiment this same issue? > >> > > >> > I also noticed a high cpu consume on call hangup. > >> > > >> > Regards > >> > > >> > 2010/8/2 Juan Antonio Iba?ez Santorum > >> >> > >> >> Hello! > >> >> > >> >> I was been some test using one FS tarball version downloaded some > weeks > >> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some > problems > >> >> with > >> >> odbc connections managemens. Now I'm using a git version (FreeSWITCH > >> >> Version > >> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve > >> >> this > >> >> problem but I noticed it has a high cpu load comparing with svn > >> >> version. > >> >> While I could manage more than 200 calls with a 75% CPU load into a > >> >> dual > >> >> core server using svn version, now, 50 calls consume this 75% cpu. I > >> >> can see > >> >> same modules are loaded (except new hash module needed in git > version) > >> >> and > >> >> same scenario is used. I am not be able to find why now it uses more > >> >> cpu > >> >> than before. Any idea? > >> >> Each testing call is a simple bridge to an external sip provider. > >> >> > >> >> Regards > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> > >> > >> -- > >> Eliot Gable > >> > >> "We do not inherit the Earth from our ancestors: we borrow it from our > >> children." ~David Brower > >> > >> "I decided the words were too conservative for me. We're not borrowing > >> from our children, we're stealing from them--and it's not even > >> considered to be a crime." ~David Brower > >> > >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > >> live; not live to eat.) ~Marcus Tullius Cicero > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/a3258c8f/attachment.html From dujinfang at gmail.com Fri Aug 6 03:03:55 2010 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Aug 2010 18:03:55 +0800 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: The default Ubuntu kernel running on 100Hz/250Hz. And there might be some great changed in timing between your two versions. I don't remember exactly. And I don't know if it's the reason causing load. I use the following command to see Hz, it should approximately increase 100 or 250 or 1000 each second. watch grep interr /proc/interrupts Also, can you try start FS with -nocal to see the difference? I still using Ubuntu because it need to schedule down time to migrate to centos in our datacenter. So I really would like to see it works ok on Ubuntu. 2010/8/6 Juan Antonio Iba?ez Santorum : > These compariso is done into the same machine. I run SVN version, make the > test, stop SVN version, run GIT version and make the test again. > > It runs over a "Intel(R) Core(TM)2 Duo CPU E7500 @ 2.93GHz" > > Regards > > 2010/8/6 Seven Du >> >> Were both of your kernel running on 1000HZ ? >> >> 2010/8/6 Juan Antonio Iba?ez Santorum : >> > I've got about 75% decrease in performarce. I am using same server and >> > same >> > config. DB was already in ramdisk as svn version so I don't find an >> > explanation for it... >> > >> > Regards >> > >> > 2010/8/5 Eliot Gable >> >> >> >> My most recent testing as of about two weeks ago showed a 25% >> >> performance drop between the old SVN version I was running and the new >> >> GIT version. Initially, it seemed closer to your reported performance >> >> drop, but after moving the database to a ramdisk, it went to 25%. I am >> >> unsure of whether my initial testing was using the db on a ramdisk, so >> >> the drop could be higher if my previous testing did not have the db on >> >> a ramdisk. >> >> >> >> 2010/8/3 Juan Antonio Iba?ez Santorum : >> >> > No one experiment this same issue? >> >> > >> >> > I also noticed a high cpu consume on call hangup. >> >> > >> >> > Regards >> >> > >> >> > 2010/8/2 Juan Antonio Iba?ez Santorum >> >> >> >> >> >> Hello! >> >> >> >> >> >> I was been some test using one FS tarball version downloaded some >> >> >> weeks >> >> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some >> >> >> problems >> >> >> with >> >> >> odbc connections managemens. Now I'm using a git version (FreeSWITCH >> >> >> Version >> >> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to >> >> >> solve >> >> >> this >> >> >> problem but I noticed it has a high cpu load comparing with svn >> >> >> version. >> >> >> While I could manage more than 200 calls with a 75% CPU load into a >> >> >> dual >> >> >> core server using svn version, now, 50 calls consume this 75% cpu. I >> >> >> can see >> >> >> same modules are loaded (except new hash module needed in git >> >> >> version) >> >> >> and >> >> >> same scenario is used. I am not be able to find why now it uses more >> >> >> cpu >> >> >> than before. Any idea? >> >> >> Each testing call is a simple bridge to an external sip provider. >> >> >> >> >> >> Regards >> >> > >> >> > >> >> > _______________________________________________ >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Eliot Gable >> >> >> >> "We do not inherit the Earth from our ancestors: we borrow it from our >> >> children." ~David Brower >> >> >> >> "I decided the words were too conservative for me. We're not borrowing >> >> from our children, we're stealing from them--and it's not even >> >> considered to be a crime." ~David Brower >> >> >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >> >> live; not live to eat.) ~Marcus Tullius Cicero >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Blog: http://www.dujinfang.com >> Proj:? http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From odermann at googlemail.com Fri Aug 6 05:01:43 2010 From: odermann at googlemail.com (Dennis) Date: Fri, 6 Aug 2010 14:01:43 +0200 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) Message-ID: hi, we are currently playing with streaming calls. with fs and icecast (over mod_shout) it works quite well. the problem is, that http-streaming to a website in conjuction with a flashplayer, a lot of overhead is produced. therefore this is no optimal solution. the best for streaming to a flashplayer embedded in a website are rtmp-streams (which are also used by webradios). red5 is an open source streaming server (written in java), which supports rtmp-streams and mp3. the problem seems to be, that fs "only" supports icecast and shoutcast. we can not get fs to work with red5. does someone have experiences with fs and red5 and can tell me, if there might be a way to get it working? or are there any technical issues, why this can't work? thanks and kind regards dennis From gustavo.espeche at upper-soft.com Fri Aug 6 05:03:21 2010 From: gustavo.espeche at upper-soft.com (Gustavo Espeche) Date: Fri, 06 Aug 2010 09:03:21 -0300 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 50, Issue 28 In-Reply-To: References: Message-ID: <1281096201.4215.4.camel@gustavo-laptop> Hi i'm suggest debian lenny, and connect to a database through mod_cul, we reach excellent result with this combination. Regards. Gustavo Espeche www.easyipcall.com EasyTrade Billing/Routing On Fri, 2010-08-06 at 01:22 -0700, freeswitch-users-request at lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: Different versions, different cpu load > (Juan Antonio Iba?ez Santorum) > 2. Re: Different versions, different cpu load > (Juan Antonio Iba?ez Santorum) > 3. Re: Different versions, different cpu load > (Juan Antonio Iba?ez Santorum) > 4. Re: Different versions, different cpu load (Seven Du) > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Juan Antonio Iba?ez Santorum > > Reply-to: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Different versions, different cpu > > load > > Date: Fri, 6 Aug 2010 08:33:22 +0200 > > > > Which distro do you sugest? > > > > Regards > > > > 2010/8/5 Brian West > > Ubuntu is a common theme when someone says "OMG IT USES ALL > > MY CPU!" > > > > > > /b > > > > On Aug 5, 2010, at 2:09 PM, Madovsky wrote: > > > > > ubunto is a nice linux distrib for audio station, not > > > servers... > > > > > > ----- Original Message ----- > > > From: "Brian West" > > > To: "FreeSWITCH Users Help" > > > > > > Sent: Thursday, August 05, 2010 2:50 PM > > > Subject: Re: [Freeswitch-users] Different versions, > > > different cpu load > > > > > > > > > > Or stop using ubuntu. > > > > > > > > /b > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Juan Antonio Iba?ez Santorum > > Reply-to: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Different versions, different cpu > > load > > Date: Fri, 6 Aug 2010 08:36:28 +0200 > > > > I've got about 75% decrease in performarce. I am using same server > > and same config. DB was already in ramdisk as svn version so I don't > > find an explanation for it... > > > > Regards > > > > 2010/8/5 Eliot Gable > > My most recent testing as of about two weeks ago showed a > > 25% > > performance drop between the old SVN version I was running > > and the new > > GIT version. Initially, it seemed closer to your reported > > performance > > drop, but after moving the database to a ramdisk, it went to > > 25%. I am > > unsure of whether my initial testing was using the db on a > > ramdisk, so > > the drop could be higher if my previous testing did not have > > the db on > > a ramdisk. > > > > 2010/8/3 Juan Antonio Iba?ez Santorum > > : > > > > > No one experiment this same issue? > > > > > > I also noticed a high cpu consume on call hangup. > > > > > > Regards > > > > > > 2010/8/2 Juan Antonio Iba?ez Santorum > > > > >> > > >> Hello! > > >> > > >> I was been some test using one FS tarball version > > downloaded some weeks > > >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has > > some problems with > > >> odbc connections managemens. Now I'm using a git version > > (FreeSWITCH Version > > >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that > > seems to solve this > > >> problem but I noticed it has a high cpu load comparing > > with svn version. > > >> While I could manage more than 200 calls with a 75% CPU > > load into a dual > > >> core server using svn version, now, 50 calls consume this > > 75% cpu. I can see > > >> same modules are loaded (except new hash module needed in > > git version) and > > >> same scenario is used. I am not be able to find why now > > it uses more cpu > > >> than before. Any idea? > > >> Each testing call is a simple bridge to an external sip > > provider. > > >> > > >> Regards > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > -- > > Eliot Gable > > > > "We do not inherit the Earth from our ancestors: we borrow > > it from our > > children." ~David Brower > > > > "I decided the words were too conservative for me. We're not > > borrowing > > from our children, we're stealing from them--and it's not > > even > > considered to be a crime." ~David Brower > > > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst > > eat to > > live; not live to eat.) ~Marcus Tullius Cicero > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Juan Antonio Iba?ez Santorum > > Reply-to: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Different versions, different cpu > > load > > Date: Fri, 6 Aug 2010 08:39:43 +0200 > > > > I thinkg it is very difficult to find it without a good knowledge of > > FS internals. Do you know if there is any way to make a profiling > > which could help? > > > > Regards > > > > 2010/8/5 Rupa Schomaker > > sounds like a good opportunity to try git-bisect to locate > > where the performance change occurs. > > > > > > > > On Thu, Aug 5, 2010 at 10:37 AM, Eliot Gable > +freeswitch at gmail.com> wrote: > > My most recent testing as of about two weeks ago > > showed a 25% > > performance drop between the old SVN version I was > > running and the new > > GIT version. Initially, it seemed closer to your > > reported performance > > drop, but after moving the database to a ramdisk, it > > went to 25%. I am > > unsure of whether my initial testing was using the > > db on a ramdisk, so > > the drop could be higher if my previous testing did > > not have the db on > > a ramdisk. > > > > 2010/8/3 Juan Antonio Iba?ez Santorum > > : > > > > > No one experiment this same issue? > > > > > > I also noticed a high cpu consume on call hangup. > > > > > > Regards > > > > > > 2010/8/2 Juan Antonio Iba?ez Santorum > > > > >> > > >> Hello! > > >> > > >> I was been some test using one FS tarball version > > downloaded some weeks > > >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but > > it has some problems with > > >> odbc connections managemens. Now I'm using a git > > version (FreeSWITCH Version > > >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) > > that seems to solve this > > >> problem but I noticed it has a high cpu load > > comparing with svn version. > > >> While I could manage more than 200 calls with a > > 75% CPU load into a dual > > >> core server using svn version, now, 50 calls > > consume this 75% cpu. I can see > > >> same modules are loaded (except new hash module > > needed in git version) and > > >> same scenario is used. I am not be able to find > > why now it uses more cpu > > >> than before. Any idea? > > >> Each testing call is a simple bridge to an > > external sip provider. > > >> > > >> Regards > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > Eliot Gable > > > > "We do not inherit the Earth from our ancestors: we > > borrow it from our > > children." ~David Brower > > > > "I decided the words were too conservative for me. > > We're not borrowing > > from our children, we're stealing from them--and > > it's not even > > considered to be a crime." ~David Brower > > > > "Esse oportet ut vivas, non vivere ut edas." (Thou > > shouldst eat to > > live; not live to eat.) ~Marcus Tullius Cicero > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > -Rupa > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Seven Du > > Reply-to: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Different versions, different cpu > > load > > Date: Fri, 6 Aug 2010 16:22:10 +0800 > > > > Were both of your kernel running on 1000HZ ? > > > > 2010/8/6 Juan Antonio Ibaez Santorum : > > > I've got about 75% decrease in performarce. I am using same server and same > > > config. DB was already in ramdisk as svn version so I don't find an > > > explanation for it... > > > > > > Regards > > > > > > 2010/8/5 Eliot Gable > > >> > > >> My most recent testing as of about two weeks ago showed a 25% > > >> performance drop between the old SVN version I was running and the new > > >> GIT version. Initially, it seemed closer to your reported performance > > >> drop, but after moving the database to a ramdisk, it went to 25%. I am > > >> unsure of whether my initial testing was using the db on a ramdisk, so > > >> the drop could be higher if my previous testing did not have the db on > > >> a ramdisk. > > >> > > >> 2010/8/3 Juan Antonio Ibaez Santorum : > > >> > No one experiment this same issue? > > >> > > > >> > I also noticed a high cpu consume on call hangup. > > >> > > > >> > Regards > > >> > > > >> > 2010/8/2 Juan Antonio Ibaez Santorum > > >> >> > > >> >> Hello! > > >> >> > > >> >> I was been some test using one FS tarball version downloaded some weeks > > >> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems > > >> >> with > > >> >> odbc connections managemens. Now I'm using a git version (FreeSWITCH > > >> >> Version > > >> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve > > >> >> this > > >> >> problem but I noticed it has a high cpu load comparing with svn > > >> >> version. > > >> >> While I could manage more than 200 calls with a 75% CPU load into a > > >> >> dual > > >> >> core server using svn version, now, 50 calls consume this 75% cpu. I > > >> >> can see > > >> >> same modules are loaded (except new hash module needed in git version) > > >> >> and > > >> >> same scenario is used. I am not be able to find why now it uses more > > >> >> cpu > > >> >> than before. Any idea? > > >> >> Each testing call is a simple bridge to an external sip provider. > > >> >> > > >> >> Regards > > >> > > > >> > > > >> > _______________________________________________ > > >> > FreeSWITCH-users mailing list > > >> > FreeSWITCH-users at lists.freeswitch.org > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> > http://www.freeswitch.org > > >> > > > >> > > > >> > > >> > > >> > > >> -- > > >> Eliot Gable > > >> > > >> "We do not inherit the Earth from our ancestors: we borrow it from our > > >> children." ~David Brower > > >> > > >> "I decided the words were too conservative for me. We're not borrowing > > >> from our children, we're stealing from them--and it's not even > > >> considered to be a crime." ~David Brower > > >> > > >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > > >> live; not live to eat.) ~Marcus Tullius Cicero > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanito1982 at gmail.com Fri Aug 6 05:13:18 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 6 Aug 2010 14:13:18 +0200 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: I cannot see updates/seconds using that command. I can see following boot params for kernel at /boot/config-2.6.32-21-server: CONFIG_NO_HZ=y CONFIG_HZ_100=y # CONFIG_HZ_250 is not set # CONFIG_HZ_300 is not set # CONFIG_HZ_1000 is not set CONFIG_HZ=100 I have been reading about this task and people recomends using 1000 HZ instead 100 ?what do you think? Centos seems use 1000 by default. Regards 2010/8/6 Seven Du > The default Ubuntu kernel running on 100Hz/250Hz. And there might be > some great changed in timing between your two versions. I don't > remember exactly. And I don't know if it's the reason causing load. > > > I use the following command to see Hz, it should approximately > increase 100 or 250 or 1000 each second. > > watch grep interr /proc/interrupts > > Also, can you try start FS with -nocal to see the difference? > > I still using Ubuntu because it need to schedule down time to migrate > to centos in our datacenter. So I really would like to see it works ok > on Ubuntu. > > 2010/8/6 Juan Antonio Iba?ez Santorum : > > These compariso is done into the same machine. I run SVN version, make > the > > test, stop SVN version, run GIT version and make the test again. > > > > It runs over a "Intel(R) Core(TM)2 Duo CPU E7500 @ 2.93GHz" > > > > Regards > > > > 2010/8/6 Seven Du > >> > >> Were both of your kernel running on 1000HZ ? > >> > >> 2010/8/6 Juan Antonio Iba?ez Santorum : > >> > I've got about 75% decrease in performarce. I am using same server and > >> > same > >> > config. DB was already in ramdisk as svn version so I don't find an > >> > explanation for it... > >> > > >> > Regards > >> > > >> > 2010/8/5 Eliot Gable > > > >> >> > >> >> My most recent testing as of about two weeks ago showed a 25% > >> >> performance drop between the old SVN version I was running and the > new > >> >> GIT version. Initially, it seemed closer to your reported performance > >> >> drop, but after moving the database to a ramdisk, it went to 25%. I > am > >> >> unsure of whether my initial testing was using the db on a ramdisk, > so > >> >> the drop could be higher if my previous testing did not have the db > on > >> >> a ramdisk. > >> >> > >> >> 2010/8/3 Juan Antonio Iba?ez Santorum : > >> >> > No one experiment this same issue? > >> >> > > >> >> > I also noticed a high cpu consume on call hangup. > >> >> > > >> >> > Regards > >> >> > > >> >> > 2010/8/2 Juan Antonio Iba?ez Santorum > >> >> >> > >> >> >> Hello! > >> >> >> > >> >> >> I was been some test using one FS tarball version downloaded some > >> >> >> weeks > >> >> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some > >> >> >> problems > >> >> >> with > >> >> >> odbc connections managemens. Now I'm using a git version > (FreeSWITCH > >> >> >> Version > >> >> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to > >> >> >> solve > >> >> >> this > >> >> >> problem but I noticed it has a high cpu load comparing with svn > >> >> >> version. > >> >> >> While I could manage more than 200 calls with a 75% CPU load into > a > >> >> >> dual > >> >> >> core server using svn version, now, 50 calls consume this 75% cpu. > I > >> >> >> can see > >> >> >> same modules are loaded (except new hash module needed in git > >> >> >> version) > >> >> >> and > >> >> >> same scenario is used. I am not be able to find why now it uses > more > >> >> >> cpu > >> >> >> than before. Any idea? > >> >> >> Each testing call is a simple bridge to an external sip provider. > >> >> >> > >> >> >> Regards > >> >> > > >> >> > > >> >> > _______________________________________________ > >> >> > FreeSWITCH-users mailing list > >> >> > FreeSWITCH-users at lists.freeswitch.org > >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > > >> >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> > http://www.freeswitch.org > >> >> > > >> >> > > >> >> > >> >> > >> >> > >> >> -- > >> >> Eliot Gable > >> >> > >> >> "We do not inherit the Earth from our ancestors: we borrow it from > our > >> >> children." ~David Brower > >> >> > >> >> "I decided the words were too conservative for me. We're not > borrowing > >> >> from our children, we're stealing from them--and it's not even > >> >> considered to be a crime." ~David Brower > >> >> > >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > >> >> live; not live to eat.) ~Marcus Tullius Cicero > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> > >> > >> -- > >> Blog: http://www.dujinfang.com > >> Proj: http://www.freeswitch.org.cn > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/8d7ceb90/attachment.html From dujinfang at gmail.com Fri Aug 6 06:05:14 2010 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Aug 2010 21:05:14 +0800 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: 2010/8/6 Juan Antonio Iba?ez Santorum : > I cannot see updates/seconds using that command. > It should show number of interrupts every second > I can see following boot params for kernel at /boot/config-2.6.32-21-server: > > CONFIG_NO_HZ=y > CONFIG_HZ_100=y > # CONFIG_HZ_250 is not set > # CONFIG_HZ_300 is not set > # CONFIG_HZ_1000 is not set > CONFIG_HZ=100 > That's it. > I have been reading about this task and people recomends using 1000 HZ > instead 100 ?what do you think? > Anthony recommends 1000 for best sound. I still use the default 250 on my box. Have you tried -nocal ? Again, I don't know if it's related with your problem. > Centos seems use 1000 by default. > > Regards > > 2010/8/6 Seven Du >> >> The default Ubuntu kernel running on 100Hz/250Hz. And there might be >> some great changed in timing between your two versions. I don't >> remember exactly. And I don't know if it's the reason causing load. >> >> >> I use the following command to see Hz, it should approximately >> increase 100 or 250 or 1000 each second. >> >> ?watch grep interr /proc/interrupts >> >> Also, can you try start FS with -nocal to see the difference? >> >> I still using Ubuntu because it need to schedule down time to migrate >> to centos in our datacenter. So I really would like to see it works ok >> on Ubuntu. >> >> 2010/8/6 Juan Antonio Iba?ez Santorum : >> > These compariso is done into the same machine. I run SVN version, make >> > the >> > test, stop SVN version, run GIT version and make the test again. >> > >> > It runs over a "Intel(R) Core(TM)2 Duo CPU E7500 @ 2.93GHz" >> > >> > Regards >> > >> > 2010/8/6 Seven Du >> >> >> >> Were both of your kernel running on 1000HZ ? >> >> >> >> 2010/8/6 Juan Antonio Iba?ez Santorum : >> >> > I've got about 75% decrease in performarce. I am using same server >> >> > and >> >> > same >> >> > config. DB was already in ramdisk as svn version so I don't find an >> >> > explanation for it... >> >> > >> >> > Regards >> >> > >> >> > 2010/8/5 Eliot Gable >> >> >> >> >> >> My most recent testing as of about two weeks ago showed a 25% >> >> >> performance drop between the old SVN version I was running and the >> >> >> new >> >> >> GIT version. Initially, it seemed closer to your reported >> >> >> performance >> >> >> drop, but after moving the database to a ramdisk, it went to 25%. I >> >> >> am >> >> >> unsure of whether my initial testing was using the db on a ramdisk, >> >> >> so >> >> >> the drop could be higher if my previous testing did not have the db >> >> >> on >> >> >> a ramdisk. >> >> >> >> >> >> 2010/8/3 Juan Antonio Iba?ez Santorum : >> >> >> > No one experiment this same issue? >> >> >> > >> >> >> > I also noticed a high cpu consume on call hangup. >> >> >> > >> >> >> > Regards >> >> >> > >> >> >> > 2010/8/2 Juan Antonio Iba?ez Santorum >> >> >> >> >> >> >> >> Hello! >> >> >> >> >> >> >> >> I was been some test using one FS tarball version downloaded some >> >> >> >> weeks >> >> >> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some >> >> >> >> problems >> >> >> >> with >> >> >> >> odbc connections managemens. Now I'm using a git version >> >> >> >> (FreeSWITCH >> >> >> >> Version >> >> >> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to >> >> >> >> solve >> >> >> >> this >> >> >> >> problem but I noticed it has a high cpu load comparing with svn >> >> >> >> version. >> >> >> >> While I could manage more than 200 calls with a 75% CPU load into >> >> >> >> a >> >> >> >> dual >> >> >> >> core server using svn version, now, 50 calls consume this 75% >> >> >> >> cpu. I >> >> >> >> can see >> >> >> >> same modules are loaded (except new hash module needed in git >> >> >> >> version) >> >> >> >> and >> >> >> >> same scenario is used. I am not be able to find why now it uses >> >> >> >> more >> >> >> >> cpu >> >> >> >> than before. Any idea? >> >> >> >> Each testing call is a simple bridge to an external sip provider. >> >> >> >> >> >> >> >> Regards >> >> >> > >> >> >> > >> >> >> > _______________________________________________ >> >> >> > FreeSWITCH-users mailing list >> >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> > >> >> >> > >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> > http://www.freeswitch.org >> >> >> > >> >> >> > >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Eliot Gable >> >> >> >> >> >> "We do not inherit the Earth from our ancestors: we borrow it from >> >> >> our >> >> >> children." ~David Brower >> >> >> >> >> >> "I decided the words were too conservative for me. We're not >> >> >> borrowing >> >> >> from our children, we're stealing from them--and it's not even >> >> >> considered to be a crime." ~David Brower >> >> >> >> >> >> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >> >> >> live; not live to eat.) ~Marcus Tullius Cicero >> >> >> >> >> >> _______________________________________________ >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > _______________________________________________ >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Blog: http://www.dujinfang.com >> >> Proj:? http://www.freeswitch.org.cn >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Blog: http://www.dujinfang.com >> Proj:? http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From dujinfang at gmail.com Fri Aug 6 06:08:47 2010 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Aug 2010 21:08:47 +0800 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: Message-ID: And another question, will this also support video? 2010/8/6 Dennis : > hi, > > we are currently playing with streaming calls. > > with fs and icecast (over mod_shout) it works quite well. the problem > is, that http-streaming to a website in conjuction with a flashplayer, > a lot of overhead is produced. therefore this is no optimal solution. > > the best for streaming to a flashplayer embedded in a website are > rtmp-streams (which are also used by webradios). red5 is an open > source streaming server (written in java), which supports rtmp-streams > and mp3. > > the problem seems to be, that fs "only" supports icecast and > shoutcast. we can not get fs to work with red5. > > does someone have experiences with fs and red5 and can tell me, if > there might be a way to get it working? or are there any technical > issues, why this can't work? > > > thanks and kind regards > dennis > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From ash at archerdrive.com Fri Aug 6 01:38:03 2010 From: ash at archerdrive.com (Ash) Date: Fri, 6 Aug 2010 18:38:03 +1000 Subject: [Freeswitch-users] No audio for 3 seconds on some calls Message-ID: <43156CB8-D7FF-4D9C-A96E-7DB6F097E475@archerdrive.com> Hi All, I am running freeswitch 1.0.6 and some users have reported that they will be on a call and the audio will go silent for 3 seconds. This doesn't appear to be happening to all users. So far, I have done the following: - Internal and External profiles set ( I was getting the RTP warning about ptime being 30 and all phones are set to 20ms) - Set the phone to use PCMA and made sure its PCMA through to the provider - We have installed a SIP Proxy at the other location so the voip phones all go via the SIP Proxy as we thought it might be a NAT issue. - Disabled VAD and CNG on the phone - The remote phones are connected via a DSL service but this appears to be rock solid and averages response times between 10ms and 40ms. I thought it might be something with VAD, is VAD disabled by default or should I add to the internal and external profiles Has anyone else come across something like this before? Maybe some other ways I can investigate what is going wrong? Any help would be much appreciated. Ash. From rupa at rupa.com Fri Aug 6 08:00:48 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 6 Aug 2010 10:00:48 -0500 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: Not really. If you have a test that shows "good/bad" we (devs) can find the reason for a problem. git handles choosing versions for testing and you handle the "good/bad" decision. The end result is "here is the git version where performance changes" which tony or whomever can then look at the patch and decide what he wants to do about it. 2010/8/6 Juan Antonio Iba?ez Santorum > I thinkg it is very difficult to find it without a good knowledge of FS > internals. Do you know if there is any way to make a profiling which could > help? > > Regards > > 2010/8/5 Rupa Schomaker > >> sounds like a good opportunity to try git-bisect to locate where the >> performance change occurs. >> >> >> On Thu, Aug 5, 2010 at 10:37 AM, Eliot Gable >> > wrote: >> >>> My most recent testing as of about two weeks ago showed a 25% >>> performance drop between the old SVN version I was running and the new >>> GIT version. Initially, it seemed closer to your reported performance >>> drop, but after moving the database to a ramdisk, it went to 25%. I am >>> unsure of whether my initial testing was using the db on a ramdisk, so >>> the drop could be higher if my previous testing did not have the db on >>> a ramdisk. >>> >>> 2010/8/3 Juan Antonio Iba?ez Santorum : >>> > No one experiment this same issue? >>> > >>> > I also noticed a high cpu consume on call hangup. >>> > >>> > Regards >>> > >>> > 2010/8/2 Juan Antonio Iba?ez Santorum >>> >> >>> >> Hello! >>> >> >>> >> I was been some test using one FS tarball version downloaded some >>> weeks >>> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems >>> with >>> >> odbc connections managemens. Now I'm using a git version (FreeSWITCH >>> Version >>> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve >>> this >>> >> problem but I noticed it has a high cpu load comparing with svn >>> version. >>> >> While I could manage more than 200 calls with a 75% CPU load into a >>> dual >>> >> core server using svn version, now, 50 calls consume this 75% cpu. I >>> can see >>> >> same modules are loaded (except new hash module needed in git version) >>> and >>> >> same scenario is used. I am not be able to find why now it uses more >>> cpu >>> >> than before. Any idea? >>> >> Each testing call is a simple bridge to an external sip provider. >>> >> >>> >> Regards >>> > >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> >>> >>> -- >>> Eliot Gable >>> >>> "We do not inherit the Earth from our ancestors: we borrow it from our >>> children." ~David Brower >>> >>> "I decided the words were too conservative for me. We're not borrowing >>> from our children, we're stealing from them--and it's not even >>> considered to be a crime." ~David Brower >>> >>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >>> live; not live to eat.) ~Marcus Tullius Cicero >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> -Rupa >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/2f062b55/attachment.html From ash at archerdrive.com Fri Aug 6 07:57:43 2010 From: ash at archerdrive.com (Ash) Date: Sat, 7 Aug 2010 00:57:43 +1000 Subject: [Freeswitch-users] No audio for 3 seconds on some calls In-Reply-To: <43156CB8-D7FF-4D9C-A96E-7DB6F097E475@archerdrive.com> References: <43156CB8-D7FF-4D9C-A96E-7DB6F097E475@archerdrive.com> Message-ID: <97E5A79A-7A70-4395-BC3F-199E0B2D82C9@archerdrive.com> I think I may have solved my own problem... I am running on Ubuntu (I know its not recommended) and my kernel timer was set to 100hz. Running time_test 1000 1000 it would avg from 997 to 1300 when I kept running it. I have now set the kernel to be 1000hz and it now avg around 1008-1014 on multiple tests. Unfortunately its not one of those always happens issues so I will have to see how it goes. On 06/08/2010, at 6:38 PM, Ash wrote: > Hi All, > > I am running freeswitch 1.0.6 and some users have reported that they will be on a call and the audio will go silent for 3 seconds. This doesn't appear to be happening to all users. So far, I have done the following: > > - Internal and External profiles set ( I was getting the RTP warning about ptime being 30 and all phones are set to 20ms) > - Set the phone to use PCMA and made sure its PCMA through to the provider > - We have installed a SIP Proxy at the other location so the voip phones all go via the SIP Proxy as we thought it might be a NAT issue. > - Disabled VAD and CNG on the phone > - The remote phones are connected via a DSL service but this appears to be rock solid and averages response times between 10ms and 40ms. > > I thought it might be something with VAD, is VAD disabled by default or should I add to the internal and external profiles > > > > Has anyone else come across something like this before? Maybe some other ways I can investigate what is going wrong? > > Any help would be much appreciated. > > Ash. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shamun.toha at gmail.com Fri Aug 6 07:57:18 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Fri, 6 Aug 2010 16:57:18 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype Message-ID: Hello, I have few questions about FS and Skype, and very confuse. Please kindly can someone give me some advise? Assume A, B, C: ============= - A to use skype messenger or mobile/land line numbers to place a call (A is a flying person, who can be in CAR/Train or office or home or restaurant) - B will always receive calls in sip phone or skype messenger (B is a static person, he is in computer, his job is operator role for every call) - C is call receiver, by B only ( C is flying person, who can be in CAR/ Train or even office waiting in Skype messenger) - A is caller, B is Operator, who decide if C will be connected, and C is waiting for a call. - A is logged in to skype messenger or not - B is logged in to skype messenger or sip phone to FS server - C is logged in to skype messenger or not - FS is installed with Skype incoming numbers + skypeID's Question: ======= Can caller A make call from his skype messenger to FS skypeID? How B will know about this call? How C can be connected by B? How the flow happens in FS and Skype? In terms of Sipphone vs Skype Messenger? Thank you Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/03b5d19a/attachment.html From mustafa.pk at gmail.com Fri Aug 6 08:37:14 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Fri, 6 Aug 2010 20:37:14 +0500 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: please keep the things simple. once you properly configure mod_skypopen it will act just like other endpoints (e.g. sip) Regards. On Fri, Aug 6, 2010 at 7:57 PM, Shamun toha md wrote: > Hello, > > I have few questions about FS and Skype, and very confuse. Please kindly can > someone give me some advise? > > Assume A, B, C: > ============= > - A to use skype messenger or mobile/land line numbers to place a call (A is > a flying person, who can be in CAR/Train or office or home or restaurant) > - B will always receive calls in sip phone or skype messenger (B is a static > person, he is in computer, his job is operator role for every call) > - C is call receiver, by B only ( C is flying person, who can be in CAR/ > Train or even office waiting in Skype messenger) > > - A is caller, B is Operator, who decide if C will be connected, and C is > waiting for a call. > - A is logged in to skype messenger or not > - B is logged in to skype messenger or sip phone to FS server > - C is logged in to skype messenger or not > > - FS is installed with Skype incoming numbers + skypeID's > > Question: > ======= > Can caller A make call from his skype messenger to FS skypeID? > How B will know about this call? > How C can be connected by B? > > How the flow happens in FS and Skype? In terms of Sipphone vs Skype > Messenger? > > > Thank you > Best regards > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com From gmaruzz at celliax.org Fri Aug 6 09:01:31 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 6 Aug 2010 18:01:31 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: On Fri, Aug 6, 2010 at 5:37 PM, Ghulam Mustafa wrote: > please keep the things simple. > > once you properly configure mod_skypopen it will act just like other > endpoints (e.g. sip) > Yes, I would concur with Ghulam. The example in your message is very hard to read, and I was not able to understand it (I mean, probably if I study it I will understand, but... :) ) If you think that can be done with any other endpoint (eg sofia), it will probably works with skypopen too. Test it, and report if you get problems. -giovanni > Regards. > > > On Fri, Aug 6, 2010 at 7:57 PM, Shamun toha md wrote: >> Hello, >> >> I have few questions about FS and Skype, and very confuse. Please kindly can >> someone give me some advise? >> >> Assume A, B, C: >> ============= >> - A to use skype messenger or mobile/land line numbers to place a call (A is >> a flying person, who can be in CAR/Train or office or home or restaurant) >> - B will always receive calls in sip phone or skype messenger (B is a static >> person, he is in computer, his job is operator role for every call) >> - C is call receiver, by B only ( C is flying person, who can be in CAR/ >> Train or even office waiting in Skype messenger) >> >> - A is caller, B is Operator, who decide if C will be connected, and C is >> waiting for a call. >> - A is logged in to skype messenger or not >> - B is logged in to skype messenger or sip phone to FS server >> - C is logged in to skype messenger or not >> >> - FS is installed with Skype incoming numbers + skypeID's >> >> Question: >> ======= >> Can caller A make call from his skype messenger to FS skypeID? >> How B will know about this call? >> How C can be connected by B? >> >> How the flow happens in FS and Skype? In terms of Sipphone vs Skype >> Messenger? >> >> >> Thank you >> Best regards >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From infos at madovsky.org Fri Aug 6 09:42:51 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 6 Aug 2010 12:42:51 -0400 Subject: [Freeswitch-users] nibbleBill action extension References: <8DC1A7902C3541C2AA553B7D82EA7658@MOBILEE1705> <201008052012.41958.sos@sokhapkin.dyndns.org> Message-ID: Also I noticed that if you set nibbileBill on leg B and his account has nobal so nibbleBill transfer the call to the nobal extension in dialplan, but (depend the trunks) his phone continues to ring. Is there any tip to avoid that ? Thank you very much Franck ----- Original Message ----- From: "Sergey Okhapkin" To: "FreeSWITCH Users Help" Sent: Thursday, August 05, 2010 8:12 PM Subject: Re: [Freeswitch-users] nibbleBill action extension > Nope. Take a look at transfer_call() function in mod_nibblebill.c. It does > transfer of both legs to the same extension. > > On Thursday 05 August 2010, Madovsky wrote: >> Is it possible to have different extension for lowbal and nobal action ? >> >> for ex I d like to play a different wave for a leg B and leg A >> >> Thanks >> >> F >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shamun.toha at gmail.com Fri Aug 6 08:53:20 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Fri, 6 Aug 2010 17:53:20 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: A need to make call using SIP or DDI or Incoming numbers? B need to receive call using SIP? C need to receive call using SIP? N.B: that means, integration of Skype inside FreeSwitch, doesn't make Skype Client software's available? Like as said, to get something you lose something!! ex: skype for mobile we have to stop dreaming, and build our own sip application for mobile? ex: skype for device we have to stop dreaming, and build our own sip application for device? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/a2d47c6d/attachment.html From shamun.toha at gmail.com Fri Aug 6 09:28:15 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Fri, 6 Aug 2010 18:28:15 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: But why is so? 1. Why should whole Skype client software need to be drop? And put 10 years of life time to build a new Sip clients for cross platforms, which is always a night mare? N.B: Skype has inbound solution, which some one will need to make? a. skype for windows b. skype of linux c. skype of mac d. skype of iphone e. skype of android f. skype of symbian g. skype of windows mobile h. skype of blackberry So, you are saying not skype client is possible? make SIP client for those platform one by one? Is there no call forwarding/call redirect to skype network from freeSwitch? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/50176092/attachment.html From sos at sokhapkin.dyndns.org Fri Aug 6 10:21:57 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 6 Aug 2010 13:21:57 -0400 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: <201008061321.57883.sos@sokhapkin.dyndns.org> I share your feelings, but can't understand what do you want to say... On Friday 06 August 2010, Shamun toha md wrote: > But why is so? > > 1. Why should whole Skype client software need to be drop? And put 10 years > of life time to build a new Sip clients for cross platforms, which is > always a night mare? > > N.B: Skype has inbound solution, which some one will need to make? > a. skype for windows > b. skype of linux > c. skype of mac > d. skype of iphone > e. skype of android > f. skype of symbian > g. skype of windows mobile > h. skype of blackberry > > > So, you are saying not skype client is possible? make SIP client for those > platform one by one? Is there no call forwarding/call redirect to skype > network from freeSwitch? > From gmaruzz at celliax.org Fri Aug 6 10:29:48 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 6 Aug 2010 19:29:48 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: Please have a read at this page, then after having *really* read it all, test it, and if you're still having problems, post here again: http://wiki.freeswitch.org/wiki/Skypopen -giovanni On Fri, Aug 6, 2010 at 4:57 PM, Shamun toha md wrote: > Hello, > > I have few questions about FS and Skype, and very confuse. Please kindly can > someone give me some advise? > > Assume A, B, C: > ============= > - A to use skype messenger or mobile/land line numbers to place a call (A is > a flying person, who can be in CAR/Train or office or home or restaurant) > - B will always receive calls in sip phone or skype messenger (B is a static > person, he is in computer, his job is operator role for every call) > - C is call receiver, by B only ( C is flying person, who can be in CAR/ > Train or even office waiting in Skype messenger) > > - A is caller, B is Operator, who decide if C will be connected, and C is > waiting for a call. > - A is logged in to skype messenger or not > - B is logged in to skype messenger or sip phone to FS server > - C is logged in to skype messenger or not > > - FS is installed with Skype incoming numbers + skypeID's > > Question: > ======= > Can caller A make call from his skype messenger to FS skypeID? > How B will know about this call? > How C can be connected by B? > > How the flow happens in FS and Skype? In terms of Sipphone vs Skype > Messenger? > > > Thank you > Best regards > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From steveayre at gmail.com Fri Aug 6 10:39:37 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 6 Aug 2010 18:39:37 +0100 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: SIP clients for all those platforms already exist, so you don't need to put 10 years of your life into writing new ones. FreeSWITCH can send calls to Skype, and can receive calls from Skype. Just like any other endpoint. All of these are possible: SIP Client -> FS -> Skype Network -> SIP Client SIP Client -> FS -> Skype Network -> Skype Client Skype Client -> FS -> Skype Network -> SIP Client Skype Client -> FS -> Skype Network -> Skype Client Does that answer your question? - Steve On 6 August 2010 17:28, Shamun toha md wrote: > But why is so? > > 1. Why should whole Skype client software need to be drop? And put 10 years > of life time to build a new Sip clients for cross platforms, which is always > a night mare? > > N.B: Skype has inbound solution, which some one will need to make? > a. skype for windows > b. skype of linux > c. skype of mac > d. skype of iphone > e. skype of android > f. skype of symbian > g. skype of windows mobile > h. skype of blackberry > > > So, you are saying not skype client is possible? make SIP client for those > platform one by one? Is there no call forwarding/call redirect to skype > network from freeSwitch? > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/6cf3279a/attachment.html From mustafa.pk at gmail.com Fri Aug 6 10:54:36 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Fri, 6 Aug 2010 22:54:36 +0500 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: AQuestion: ======= Can caller A make call from his skype messenger to FS skypeID? *(yes ofcourse) -- Caller A (On skype messenger or sip device) will be able to call FS SkypeID, Caller A's call will land into FS dialplan, and you can write your Call Routing logic in dialplan (unlimited possibilities) * How B will know about this call? *Again It's your FS dialplan login which will route calls. and is responsible for ringing destinations you define in it.* How C can be connected by B? *Again, as above mentioned. * How the flow happens in FS and Skype? In terms of Sipphone vs Skype Messenger? *SIP devices registered with your FS can only communicate to each other, * *For Landline/GSM you will need to terminate your SIP calls to some Voip Gateway or Analog/Digital Trunk line (defined in your FS configuration) ,* *To allow your SIP Devices to talk with Skype users you will need mod_skypopen (FS Skype Module)* and regarding your question to get rid of skype messenger i fully support you, but problem is skype holds 65% of voip traffic all over the world. and we need a solution which can help us reach other 65% as well. Regards. On Fri, Aug 6, 2010 at 10:29 PM, Giovanni Maruzzelli wrote: > Please have a read at this page, then after having *really* read it > all, test it, and if you're still having problems, post here again: > http://wiki.freeswitch.org/wiki/Skypopen > > -giovanni > > On Fri, Aug 6, 2010 at 4:57 PM, Shamun toha md wrote: >> Hello, >> >> I have few questions about FS and Skype, and very confuse. Please kindly can >> someone give me some advise? >> >> Assume A, B, C: >> ============= >> - A to use skype messenger or mobile/land line numbers to place a call (A is >> a flying person, who can be in CAR/Train or office or home or restaurant) >> - B will always receive calls in sip phone or skype messenger (B is a static >> person, he is in computer, his job is operator role for every call) >> - C is call receiver, by B only ( C is flying person, who can be in CAR/ >> Train or even office waiting in Skype messenger) >> >> - A is caller, B is Operator, who decide if C will be connected, and C is >> waiting for a call. >> - A is logged in to skype messenger or not >> - B is logged in to skype messenger or sip phone to FS server >> - C is logged in to skype messenger or not >> >> - FS is installed with Skype incoming numbers + skypeID's >> >> Question: >> ======= >> Can caller A make call from his skype messenger to FS skypeID? >> How B will know about this call? >> How C can be connected by B? >> >> How the flow happens in FS and Skype? In terms of Sipphone vs Skype >> Messenger? >> >> >> Thank you >> Best regards >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/061ed7ff/attachment-0001.html From 12ukwn at gmail.com Fri Aug 6 11:31:12 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 06 Aug 2010 20:31:12 +0200 Subject: [Freeswitch-users] FS performance Message-ID: <4C5C54F0.6000000@gmail.com> Hi list, How does FS performs compared to Kamailio on the same machine? (I'm just talking about pure switching, no transcoding, nor AA) JY -- From sos at sokhapkin.dyndns.org Fri Aug 6 11:45:19 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 6 Aug 2010 14:45:19 -0400 Subject: [Freeswitch-users] FS performance In-Reply-To: <4C5C54F0.6000000@gmail.com> References: <4C5C54F0.6000000@gmail.com> Message-ID: <201008061445.19345.sos@sokhapkin.dyndns.org> Don't compare apples to oranges, FS is b2bua. I'd say FS takes 30-40 times more CPU power than openser. No magic. On Friday 06 August 2010, Jean-Yves F. Barbier wrote: > Hi list, > > How does FS performs compared to Kamailio on the same machine? > (I'm just talking about pure switching, no transcoding, nor > AA) > > JY > From shamun.toha at gmail.com Fri Aug 6 11:06:33 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Fri, 6 Aug 2010 20:06:33 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: Great thanks. I got my answer. SIP Client -> FS -> Skype Network -> SIP Client SIP Client -> FS -> Skype Network -> Skype Client Skype Client -> FS -> Skype Network -> SIP Client Skype Client -> FS -> Skype Network -> Skype Client -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/56ab2586/attachment.html From 12ukwn at gmail.com Fri Aug 6 12:33:34 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 06 Aug 2010 21:33:34 +0200 Subject: [Freeswitch-users] FS performance In-Reply-To: <201008061445.19345.sos@sokhapkin.dyndns.org> References: <4C5C54F0.6000000@gmail.com> <201008061445.19345.sos@sokhapkin.dyndns.org> Message-ID: <4C5C638E.3060002@gmail.com> Le 06/08/2010 20:45, Sergey Okhapkin a ?crit : > Don't compare apples to oranges, FS is b2bua. They both have juice, seeds and fall from trees ;) > I'd say FS takes 30-40 times more CPU power than openser. No magic. I thought using it only as a switch the CPU consumption was lowered. But this answer my unformulated question which was: why using a couple made of Kamailio & FS (I'm trying to find a powerful but malleable solution that could fit either SMB & large companies) Thanks > On Friday 06 August 2010, Jean-Yves F. Barbier wrote: >> Hi list, >> >> How does FS performs compared to Kamailio on the same machine? >> (I'm just talking about pure switching, no transcoding, nor >> AA) >> >> JY >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- I'm mentally OVERDRAWN! What's that SIGNPOST up ahead? Where's ROD STERLING when you really need him? From sos at sokhapkin.dyndns.org Fri Aug 6 13:12:36 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 6 Aug 2010 16:12:36 -0400 Subject: [Freeswitch-users] FS performance In-Reply-To: <4C5C638E.3060002@gmail.com> References: <4C5C54F0.6000000@gmail.com> <201008061445.19345.sos@sokhapkin.dyndns.org> <4C5C638E.3060002@gmail.com> Message-ID: <201008061612.36416.sos@sokhapkin.dyndns.org> You can't use FS as a pure SIP router. It's b2bua. On Friday 06 August 2010, Jean-Yves F. Barbier wrote: > Le 06/08/2010 20:45, Sergey Okhapkin a ?crit : > > Don't compare apples to oranges, FS is b2bua. > > They both have juice, seeds and fall from trees ;) > > > I'd say FS takes 30-40 times more CPU power than openser. No magic. > > I thought using it only as a switch the CPU consumption was lowered. > > But this answer my unformulated question which was: why using a couple made > of Kamailio & FS (I'm trying to find a powerful but malleable solution > that could fit either SMB & large companies) > > Thanks > > > On Friday 06 August 2010, Jean-Yves F. Barbier wrote: > >> Hi list, > >> > >> How does FS performs compared to Kamailio on the same machine? > >> (I'm just talking about pure switching, no transcoding, nor > >> AA) > >> > >> JY > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From infos at madovsky.org Fri Aug 6 13:36:06 2010 From: infos at madovsky.org (Madovsky) Date: Fri, 6 Aug 2010 16:36:06 -0400 Subject: [Freeswitch-users] destination_number question Message-ID: <80535773A206468D85FC0F9959E801A6@MOBILEE1705> :) to be more precise it needs [] with OR Sent from my commodre 64 Franck ----- Original Message ----- From: msc at freeswitch.org To: Madovsky Sent: Friday, August 06, 2010 4:06 PM Subject: Re: [Freeswitch-users] destination_number question I think you have seen the light! Remember that stacking conditions results in a logical AND statement. Like you mentioned, using alternation (that is, using the pipe) is how you do logical OR statements. -MC Sent from my HTC on the Now Network from Sprint! ----- Reply message ----- From: "Madovsky" Date: Thu, Aug 5, 2010 10:25 pm Subject: [Freeswitch-users] destination_number question To: I think I need to rest, drink and smoke again.... what I neded was simply an expression like this ----- Original Message ----- From: msc at freeswitch.org To: Madovsky Sent: Thursday, August 05, 2010 9:02 PM Subject: Re: [Freeswitch-users] destination_number question You could but why? In your example the blahblah would never get executed because there is no dest num that could possibly match both conditions. What are you trying to accomplish? Also, don't forget that Darren did a great job of explaining this kind of thing in chapter 8 of the FS book. -MC - Sent from my HTC on the Now Network from Sprint! ----- Reply message ----- From: "Madovsky" Date: Thu, Aug 5, 2010 2:30 pm Subject: [Freeswitch-users] destination_number question To: have some doubt can I have destination_number stacked condition like blablablaa Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users at lists.freeswitch.org Sent: Thursday, August 05, 2010 2:00 PM Subject: destination_number question if I call from a sip phone for ex 99999999 at mydomain and use in dialplan it fails. I need to remove the << ^ >> and it works. so I there any chars in front of the number in destination_number variable if I call from SIP phone ? Thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/8b0e6b1b/attachment.html From 12ukwn at gmail.com Fri Aug 6 14:33:18 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 06 Aug 2010 23:33:18 +0200 Subject: [Freeswitch-users] FS performance In-Reply-To: <201008061612.36416.sos@sokhapkin.dyndns.org> References: <4C5C54F0.6000000@gmail.com> <201008061445.19345.sos@sokhapkin.dyndns.org> <4C5C638E.3060002@gmail.com> <201008061612.36416.sos@sokhapkin.dyndns.org> Message-ID: <4C5C7F9E.9040400@gmail.com> Le 06/08/2010 22:12, Sergey Okhapkin a ?crit : > You can't use FS as a pure SIP router. It's b2bua. Ah, I still need to study then. I miss some of the VoIP vocabulary, could you point me to a place that could be a kind of dictionnary PLS? -- The only people who make love all the time are liars. -- Louis Jordan From shamun.toha at gmail.com Fri Aug 6 11:54:36 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Fri, 6 Aug 2010 20:54:36 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: Thanks Steve!, Mustafa, those few words from both of you were really transparent and special. I appreciate it. @Mustafa: Never mind, just personal comment ========= 1. I spend 3 years to make my j2ME code works for SIP got failed, audio codec's killed me 2. I spend more time to use OS SIP got failed 3. I spend more time to use pjSIP got failed (segmentation fault), and there supports are really dead 4. I spend more time to find Alchemy+Flex sdk to make SIP user agent cross platform. ========================================================================== Sub total: 4 hard time Total amount to pay: 6 year (PAID, got nothing) I am sorry, i have to use Skype client (skype4com/skypeKit) this time, because the SIP user agent client is not for junior programmers, there is strongly no cross platform (web/mobile/desktop) solid release available yet. Many thanks for sharing. I know at least now, what i can do to forward my project. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/11a5f5e8/attachment-0001.html From tayeb.meftah at gmail.com Sat Aug 7 11:31:58 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 07 Aug 2010 20:31:58 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: <4C5DA69E.5080709@gmail.com> go hack the skype protocol and build your own skype endpoint module you need to say thanks to giovanni and apreciate his contribution to the project Le 06/08/2010 18:28, Shamun toha md a ?crit : > But why is so? > > 1. Why should whole Skype client software need to be drop? And put 10 > years of life time to build a new Sip clients for cross platforms, > which is always a night mare? > > N.B: Skype has inbound solution, which some one will need to make? > a. skype for windows > b. skype of linux > c. skype of mac > d. skype of iphone > e. skype of android > f. skype of symbian > g. skype of windows mobile > h. skype of blackberry > > > So, you are saying not skype client is possible? make SIP client for > those platform one by one? Is there no call forwarding/call redirect > to skype network from freeSwitch? > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100807/ed90542b/attachment.html From mewash at gmail.com Fri Aug 6 15:30:53 2010 From: mewash at gmail.com (milosz) Date: Fri, 6 Aug 2010 18:30:53 -0400 Subject: [Freeswitch-users] freeswitch as fax/voice router Message-ID: hi all, i am looking to use freeswitch to do uc fax/voice routing for my user did's. my users are on an ip pbx already (sipx, but the pre-freeswitch version, at least for now). i have calls coming into a pri and i'd like to use freeswitch to do fax detection on the call and do either fax-to-email or send the call to sipx. is anyone doing anything like this? is it feasible? clues for how to get started? thanks milosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/cfef6e71/attachment.html From mustafa.pk at gmail.com Fri Aug 6 15:56:45 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Sat, 7 Aug 2010 03:56:45 +0500 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: <4C5DA69E.5080709@gmail.com> References: <4C5DA69E.5080709@gmail.com> Message-ID: Hi, I believe pjsip is capable of running on many platforms (mac, iphone, windows, linux, symbion, windows mobile), and their support doesn't sucks at all. pjsua is extremely easy to embed in your application. also give a try to linphone this year. if you have spent coding 6 years then don't tell me you are still a junior level programmer/developer, there is something seriously wrong with your approach, please try to elaborate what actually you are trying to do, read the documentations carefully, try to troubleshoot problems you are facing and come-up with to the point questions at mailing lists/ forums, people would love helping you for sure! Best Regards, On Sat, Aug 7, 2010 at 11:31 PM, Meftah Tayeb wrote: > go hack the skype protocol and build your own skype endpoint module > you need to say thanks to giovanni and apreciate his contribution to the > project > Le 06/08/2010 18:28, Shamun toha md a ?crit : > > But why is so? > > 1. Why should whole Skype client software need to be drop? And put 10 years > of life time to build a new Sip clients for cross platforms, which is always > a night mare? > > N.B: Skype has inbound solution, which some one will need to make? > a. skype for windows > b. skype of linux > c. skype of mac > d. skype of iphone > e. skype of android > f. skype of symbian > g. skype of windows mobile > h. skype of blackberry > > > So, you are saying not skype client is possible? make SIP client for those > platform one by one? Is there no call forwarding/call redirect to skype > network from freeSwitch? > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > -- > Meftah Tayeb > alg?rie t?l?com SPA > phone: +21321761805 > phone (INUM): +883510001289101 > mobile : +213660347746 > mobile (INUM: +883510001289110http://www.algerietelecom.dz > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100807/f342b90f/attachment.html From tgraziano at myitdepartment.net Fri Aug 6 16:31:40 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Fri, 6 Aug 2010 19:31:40 -0400 Subject: [Freeswitch-users] freeswitch as fax/voice router In-Reply-To: References: Message-ID: Hi Milosz. "FreeSWITCH now has T38 support. The old mod_fax and many of the codecs in FreeSWITCH have now merged to one module called mod_spandsp which takes advantage of all the DSP features found in the spandsp library including T.38 endpoint and gateway functionality." This should be considered (see sipfoundry JIRA, item for that now) for the FS 1.07 release in sipxecs. In the meantime, you should consider using a FS system using mod_spandsp and using t.38. Good to see you here too. Tony On Fri, Aug 6, 2010 at 6:30 PM, milosz wrote: > hi all, > > i am looking to use freeswitch to do uc fax/voice routing for my user > did's. > > my users are on an ip pbx already (sipx, but the pre-freeswitch version, at > least for now). > > i have calls coming into a pri and i'd like to use freeswitch to do fax > detection on the call and do either fax-to-email or send the call to sipx. > > is anyone doing anything like this? is it feasible? clues for how to get > started? > > thanks > > milosz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100806/85436a8c/attachment.html From brian at microcomaustralia.com.au Sat Aug 7 01:35:08 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Sat, 7 Aug 2010 18:35:08 +1000 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: On 4 August 2010 22:28, Steven Ayre wrote: > Well, you probably only have one file - core.db On my system I seem to have a number of db files: -rw-r--r-- 1 freeswitch daemon 14336 2010-08-04 21:37 call_limit.db -rw-r--r-- 1 freeswitch daemon 129024 2010-08-07 18:33 core.db -rw-r--r-- 1 freeswitch daemon 5120 2010-08-07 18:27 fifo.db -rw-r--r-- 1 freeswitch daemon 70656 2010-08-07 16:18 sofia_reg_external.db -rw-r--r-- 1 freeswitch daemon 1502208 2010-08-07 18:27 sofia_reg_internal.db -rw-r--r-- 1 freeswitch daemon 70656 2010-08-07 16:18 sofia_reg_internal-ipv6.db -rw-r--r-- 1 freeswitch daemon 14336 2010-08-04 21:43 voicemail_default.db > You shouldn't have lost anything important in there - it mostly just stored > information on current calls, sip registrations that kind of thing. Ok, good. > As for how to do sqlite data recovery, try googling. e.g. > http://www.mail-archive.com/sqlite-users at sqlite.org/msg17538.html Thanks for the reference. I tried looking but obviously wasn't looking for the right search term. -- Brian May From a.afzali2003 at gmail.com Sat Aug 7 12:36:24 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 8 Aug 2010 00:06:24 +0430 Subject: [Freeswitch-users] No audio for 3 seconds on some calls In-Reply-To: <97E5A79A-7A70-4395-BC3F-199E0B2D82C9@archerdrive.com> References: <43156CB8-D7FF-4D9C-A96E-7DB6F097E475@archerdrive.com> <97E5A79A-7A70-4395-BC3F-199E0B2D82C9@archerdrive.com> Message-ID: Hi Ash, Did you change your kernel to -preempt (10.04 Lucid Lynx) ? -- afshin On Fri, Aug 6, 2010 at 7:27 PM, Ash wrote: > I think I may have solved my own problem... I am running on Ubuntu (I know > its not recommended) and my kernel timer was set to 100hz. Running > time_test 1000 1000 it would avg from 997 to 1300 when I kept running it. I > have now set the kernel to be 1000hz and it now avg around 1008-1014 on > multiple tests. > > Unfortunately its not one of those always happens issues so I will have to > see how it goes. > > > > > On 06/08/2010, at 6:38 PM, Ash wrote: > > > Hi All, > > > > I am running freeswitch 1.0.6 and some users have reported that they will > be on a call and the audio will go silent for 3 seconds. This doesn't > appear to be happening to all users. So far, I have done the following: > > > > - Internal and External profiles set value="false" /> ( I was getting the RTP warning about ptime being 30 and > all phones are set to 20ms) > > - Set the phone to use PCMA and made sure its PCMA through to the > provider > > - We have installed a SIP Proxy at the other location so the voip phones > all go via the SIP Proxy as we thought it might be a NAT issue. > > - Disabled VAD and CNG on the phone > > - The remote phones are connected via a DSL service but this appears to > be rock solid and averages response times between 10ms and 40ms. > > > > I thought it might be something with VAD, is VAD disabled by default or > should I add to the internal and external profiles > > > > > > > > Has anyone else come across something like this before? Maybe some > other ways I can investigate what is going wrong? > > > > Any help would be much appreciated. > > > > Ash. > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/a6a21d66/attachment-0001.html From ritzalam at gmail.com Sat Aug 7 13:13:23 2010 From: ritzalam at gmail.com (Richard Alam) Date: Sat, 7 Aug 2010 16:13:23 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: Message-ID: Yes, this can be done. We are doing this for voice conference using Flash, Red5 and FreeSWITHCH/Asterisk. We basically have a SIP client based on Red5Phone. The SIP client takes audiodata from RTMP, transcodes Nellymoser to Ulaw, then send the audio data using RTP to FS/Asterisk. With FS, we can use Speex Wideband so there is no need to transcode but to just take audio data from RTMP then shove it into an RTP packet to FS. You can take a look at the code here http://github.com/bigbluebutton/bigbluebutton/tree/master/bbb-voice/. Let me know if you have any questions. Richard On Fri, Aug 6, 2010 at 8:01 AM, Dennis wrote: > hi, > > we are currently playing with streaming calls. > > with fs and icecast (over mod_shout) it works quite well. the problem > is, that http-streaming to a website in conjuction with a flashplayer, > a lot of overhead is produced. therefore this is no optimal solution. > > the best for streaming to a flashplayer embedded in a website are > rtmp-streams (which are also used by webradios). red5 is an open > source streaming server (written in java), which supports rtmp-streams > and mp3. > > the problem seems to be, that fs "only" supports icecast and > shoutcast. we can not get fs to work with red5. > > does someone have experiences with fs and red5 and can tell me, if > there might be a way to get it working? or are there any technical > issues, why this can't work? > > > thanks and kind regards > dennis > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton From cmrienzo at gmail.com Sat Aug 7 19:53:03 2010 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Sat, 7 Aug 2010 22:53:03 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: Message-ID: I believe math is working on an endpoint module for rtmp, if my memory of his cluecon presentation is correct. On Aug 6, 2010 8:08 AM, "Dennis" wrote: hi, we are currently playing with streaming calls. with fs and icecast (over mod_shout) it works quite well. the problem is, that http-streaming to a website in conjuction with a flashplayer, a lot of overhead is produced. therefore this is no optimal solution. the best for streaming to a flashplayer embedded in a website are rtmp-streams (which are also used by webradios). red5 is an open source streaming server (written in java), which supports rtmp-streams and mp3. the problem seems to be, that fs "only" supports icecast and shoutcast. we can not get fs to work with red5. does someone have experiences with fs and red5 and can tell me, if there might be a way to get it working? or are there any technical issues, why this can't work? thanks and kind regards dennis _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100807/158fad57/attachment.html From mrene_lists at avgs.ca Sat Aug 7 21:17:19 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 8 Aug 2010 00:17:19 -0400 Subject: [Freeswitch-users] FS performance In-Reply-To: <4C5C7F9E.9040400@gmail.com> References: <4C5C54F0.6000000@gmail.com> <201008061445.19345.sos@sokhapkin.dyndns.org> <4C5C638E.3060002@gmail.com> <201008061612.36416.sos@sokhapkin.dyndns.org> <4C5C7F9E.9040400@gmail.com> Message-ID: Hi, While FreeSWICH is a back-to-back user agent (B2BUA), you can do some magic on sip<->sip calls. Kamailio wont do any media proxying on its own, to do the same with FS, you need to use bypass_media (http://wiki.freeswitch.org/wiki/Bypass_media) That will use considerably less resources than relaying an rtp packet every 20ms (the default setting). In fact, it wont use anything at all once the call is up. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-08-06, at 5:33 PM, Jean-Yves F. Barbier wrote: > Le 06/08/2010 22:12, Sergey Okhapkin a ?crit : >> You can't use FS as a pure SIP router. It's b2bua. > > Ah, I still need to study then. > I miss some of the VoIP vocabulary, could you point me to a place > that could be a kind of dictionnary PLS? > > -- > The only people who make love all the time are liars. > -- Louis Jordan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From xyangni at gmail.com Sun Aug 8 05:55:48 2010 From: xyangni at gmail.com (xuyan yang) Date: Sun, 8 Aug 2010 20:55:48 +0800 Subject: [Freeswitch-users] Skype message event miss match in concurrent call Message-ID: Hi, I am using skypopne with 2 account A and B. And A is the callee to receive call. During the call skype chat message is used as a method of input. According to skypopen logic, when C call A first and then D call A. B will pick up the call from D. On some mobile version skype, the transfer from A to B is not reported to mobile user. So user D will still send message to A. In my event processing script(in js). The session C->A will receive all message D sent to A while D to B session receive nothing. Hence caller D will not be responsed by system. Is there any way to handler this situation? For example, when C->A session receive a message from D, can I fire it again into D to B session? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/00821822/attachment.html From gmaruzz at celliax.org Sun Aug 8 07:52:49 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 8 Aug 2010 16:52:49 +0200 Subject: [Freeswitch-users] Skype message event miss match in concurrent call In-Reply-To: References: Message-ID: Mmmmh, I'm one of the developers of skypopen, and at first tought I think it is not possible to solve this problem. But I would ask you to open a jira on this, because I would like to have it documented, and maybe in future releases to find a workaround. No guarantees on solution, but if you open a jira at least this will be given more toughts. -giovanni On 8/8/10, xuyan yang wrote: > Hi, I am using skypopne with 2 account A and B. And A is the callee to > receive call. During the call skype chat message is used as a method of > input. > According to skypopen logic, when C call A first and then D call A. B will > pick up the call from D. On some mobile version skype, the transfer from A > to B is not reported to mobile user. So user D will still send message to A. > In my event processing script(in js). The session C->A will receive all > message D sent to A while D to B session receive nothing. Hence caller D > will not be responsed by system. > Is there any way to handler this situation? For example, when C->A session > receive a message from D, can I fire it again into D to B session? > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From mnhassan at usa.net Sun Aug 8 08:45:33 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sun, 8 Aug 2010 21:45:33 +0600 Subject: [Freeswitch-users] Mod_easyroute w/postgresql-odbc Cannot Find Data Source Name and Default Driver In-Reply-To: References: Message-ID: Nobody replied to this in the last 3 weeks, I hope you have found your answer. I was stumped on this issue too, and fixed it following the wiki. The basic test that I ran before anything was: isql It was failing at first, then I tracked the error to incorrect "Database" in my "dsn" section of the "odbc.ini". The "isql" command helped in troubleshooting the basic operation of ODBC outside of Freeswitch. Hope you have resolved it by now. Regards HASSAN On Sat, Jul 17, 2010 at 00:04, Jerry Richards wrote: > > I'm having trouble getting mod_easyroute to work with odbc postgresql. > Please see the error I am getting below. Is there someone who has done > this > and could tell me what I'm doing wrong? > > Here is the error I'm getting: > [ERR] switch_odbc.c:313 STATE IM002 CODE 0 ERROR: [unixODBC][Driver > Manager]Data source name not found, and no default driver specified > [CRIT] mod_easyroute.c:147 Cannot Open ODBC Database! > > Here is my conf/autoload_configs/easyroute_conf.xml content: > > > > > Here is my odbc.ini content: > [ez] > Description = PostgreSQL Unicode > Driver = PostgreSQL Unicode > Trace = No > TraceFile = /home/TeoUser/db/psqlodbc.log > Database = ez > Servername = 127.0.0.1 > UserName = postgres > Password = password > Port = 5432 > ReadOnly = Yes > RowVersioning = No > ShowSystemTables = No > ShowOidColumn = No > FakeOidIndex = No > ConnSettings = > > Here is my odbcinst.ini file content: > [PostgreSQL] > Description = ODBC for PostgreSQL > Driver = /usr/lib/libodbcpsql.so > Setup = /usr/lib/libodbcpsqlS.so > FileUsage = 1 > > Thanks And Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/b83b98ca/attachment.html From ken at ukgb.net Sun Aug 8 10:46:01 2010 From: ken at ukgb.net (Ken Gillett) Date: Sun, 8 Aug 2010 18:46:01 +0100 Subject: [Freeswitch-users] Modems Message-ID: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> I asked this before, but never got any response, so hope someone can help me this time. Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls? I plan on installing FreeSwitch on a Mac and it would be really handy to be able to use Apple's USB modem in this way as otherwise there's no way to add such PSTN functionality on the Mac Mini. Although my requirement is Mac based, I guess this question applies to any FreeSwitch installation. Ken G i l l e t t _/_/_/_/_/_/_/_/ From gmaruzz at celliax.org Sun Aug 8 11:43:41 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 8 Aug 2010 20:43:41 +0200 Subject: [Freeswitch-users] Modems In-Reply-To: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> Message-ID: No, at the moment is not possible. -giovanni On 8/8/10, Ken Gillett wrote: > I asked this before, but never got any response, so hope someone can help me > this time. > > Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. > be able to accept incoming PSTN calls and also make PSTN calls? > > I plan on installing FreeSwitch on a Mac and it would be really handy to be > able to use Apple's USB modem in this way as otherwise there's no way to add > such PSTN functionality on the Mac Mini. Although my requirement is Mac > based, I guess this question applies to any FreeSwitch installation. > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From tgraziano at myitdepartment.net Sun Aug 8 12:49:10 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Sun, 8 Aug 2010 15:49:10 -0400 Subject: [Freeswitch-users] Modems In-Reply-To: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> Message-ID: You can always use a standalone sip gateway (patton, audiocodes, etc.). On 8/8/10, Ken Gillett wrote: > I asked this before, but never got any response, so hope someone can help me > this time. > > Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. > be able to accept incoming PSTN calls and also make PSTN calls? > > I plan on installing FreeSwitch on a Mac and it would be really handy to be > able to use Apple's USB modem in this way as otherwise there's no way to add > such PSTN functionality on the Mac Mini. Although my requirement is Mac > based, I guess this question applies to any FreeSwitch installation. > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. From asrivas at gmail.com Sun Aug 8 00:38:47 2010 From: asrivas at gmail.com (Anurag Srivastava) Date: Sun, 8 Aug 2010 00:38:47 -0700 Subject: [Freeswitch-users] FS frequent crashes when using txfax Message-ID: Segfault in fs that happens pretty consistently (but not always) when using txfax. Attaching the backtrace from gdb on the core file and the log on fs_cli. http://pastebin.freeswitch.org/13602 I did a "make current" to get on the latest version. Running on ubuntu hardy thanks Anurag git log. -------- commit 9b4ab6f87798b8e7a65d05357ee7fcd82551cfd5 Date: Fri Aug 6 11:23:31 2010 +0200 gdb core backtrace ---------------------------- (gdb) bt #0 0xb7b88869 in strcasecmp () from /lib/tls/i686/cmov/libc.so.6 #1 0xb6b9d093 in negotiate_t38 (pvt=0x82e41e0) at mod_spandsp_fax.c:760 #2 0xb6ba1c06 in mod_spandsp_fax_process_fax (session=0x82dbca8, .... -- Regards Anurag Srivastava -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/d08f16cb/attachment.html From ash at archerdrive.com Sun Aug 8 03:39:51 2010 From: ash at archerdrive.com (Ash) Date: Sun, 8 Aug 2010 20:39:51 +1000 Subject: [Freeswitch-users] No audio for 3 seconds on some calls In-Reply-To: References: <43156CB8-D7FF-4D9C-A96E-7DB6F097E475@archerdrive.com> <97E5A79A-7A70-4395-BC3F-199E0B2D82C9@archerdrive.com> Message-ID: <8BBE105C-F670-4BC4-8BAF-483C69E17EB8@archerdrive.com> Hi afshin, I changed to the linux-image-rt (real time preemption patch) and so far the calls have been good. A bulk of the calls will happen tomorrow so I will have more of and idea if it worked. Looking at it now its set to 1000HZ so I hope this solves the issue. Ash. On 08/08/2010, at 5:36 AM, afshin afzali wrote: > Hi Ash, > > Did you change your kernel to -preempt (10.04 Lucid Lynx) ? > > -- afshin > > On Fri, Aug 6, 2010 at 7:27 PM, Ash wrote: > I think I may have solved my own problem... I am running on Ubuntu (I know its not recommended) and my kernel timer was set to 100hz. Running time_test 1000 1000 it would avg from 997 to 1300 when I kept running it. I have now set the kernel to be 1000hz and it now avg around 1008-1014 on multiple tests. > > Unfortunately its not one of those always happens issues so I will have to see how it goes. > > > > > On 06/08/2010, at 6:38 PM, Ash wrote: > > > Hi All, > > > > I am running freeswitch 1.0.6 and some users have reported that they will be on a call and the audio will go silent for 3 seconds. This doesn't appear to be happening to all users. So far, I have done the following: > > > > - Internal and External profiles set ( I was getting the RTP warning about ptime being 30 and all phones are set to 20ms) > > - Set the phone to use PCMA and made sure its PCMA through to the provider > > - We have installed a SIP Proxy at the other location so the voip phones all go via the SIP Proxy as we thought it might be a NAT issue. > > - Disabled VAD and CNG on the phone > > - The remote phones are connected via a DSL service but this appears to be rock solid and averages response times between 10ms and 40ms. > > > > I thought it might be something with VAD, is VAD disabled by default or should I add to the internal and external profiles > > > > > > > > Has anyone else come across something like this before? Maybe some other ways I can investigate what is going wrong? > > > > Any help would be much appreciated. > > > > Ash. > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/6f477bc5/attachment.html From andrea at medu.it Sun Aug 8 12:29:47 2010 From: andrea at medu.it (Andrea Medusei) Date: Sun, 8 Aug 2010 21:29:47 +0200 Subject: [Freeswitch-users] freeswitch and zrtp In-Reply-To: References: Message-ID: <000001cb3730$121a6a30$364f3e90$@it> Hi all, I'm deploying a ippbx with freeswitch and zrtp. It seems to work, but when i place a call i can't hear clean. It's impossible to understand what people say. I'm using tivi client for symbian s60. Both end have zrtp support (one unlimited and one passive licence). I also installed FS with zrtp support. Ccalls went to secure mode correctly but I can't hear nothing. I tried to install another FS, I get this error: switch_rtp.c:2122 Error: zRTP protection drop with code 9. I'm a little desperate :-) Any similar experience or possible solutions? From jprsa at yahoo.com Sat Aug 7 07:31:01 2010 From: jprsa at yahoo.com (jprsa) Date: Sat, 7 Aug 2010 07:31:01 -0700 (PDT) Subject: [Freeswitch-users] ACLs through proxy Message-ID: <29331510.post@talk.nabble.com> Hello, Were you able to finally get your UAs to register on your FS boxes behind the proxy (opensip) server? Thanks Bill W-3 wrote: > > That's fantastic! FreeSWITCH ROCKS! > > I'll update the wiki. > > Thanks, > Bill > > > > Brian West wrote: >> use "apply-proxy-acl" on the sofia profile. >> >> /b >> >> On Dec 15, 2009, at 10:58 PM, Bill W wrote: >> >>> However, having the proxy in the path effectively negates using IP >>> based >>> ACLS. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/ACLs-through-proxy-tp26806529p29331510.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From shamun.toha at gmail.com Sat Aug 7 04:19:56 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sat, 7 Aug 2010 13:19:56 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: Please kindly check this: http://gist.github.com/512698 Step 1: done Step 2: I am not finding any head nor tail, what to compile here? What exactly i should do to getting started? Thank you Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100807/ec8e343e/attachment.html From shamun.toha at gmail.com Sat Aug 7 16:22:19 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 8 Aug 2010 01:22:19 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: Can't make it working yet ( i got 1 pc to test ). Please advise. 1. Compile -------------------------------------- [ ok ] [root at example configs]# gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 2. Instance ------------------------------------- [ fail ] [root at example configs]# ssh -X localhost ssh: connect to host localhost port 22: Connection refused [root at example configs]# /usr/bin/skype & [1] 20521 3. skypopen_auth ----------------------------- [ fail ] [root at example configs]# ./skypopen_auth :101 Cannot open X Display ':101', exiting [root at example configs]# ./skypopen_auth Skype instance found on display ':0.0', with id #94371918 RECEIVED==> ERROR 68 RECEIVED==> OK ^C [1]+ Done /usr/bin/skype Question: ========= Url said: http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#How_to_prepare_the_configuration_directory_of_Skype_clients_on_Linux_using_ssh_-X_and_xauth Expecting this following output, but not getting this why? $ ./skypopen_auth Skype instance found with id #27263062 RECEIVED==> OK RECEIVED==> PROTOCOL 6 RECEIVED==> CONNSTATUS ONLINE RECEIVED==> CURRENTUSERHANDLE gmaruzz3 RECEIVED==> USERSTATUS INVISIBLE Many thanks Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/79996b2c/attachment-0001.html From shamun.toha at gmail.com Sun Aug 8 08:05:18 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 8 Aug 2010 17:05:18 +0200 Subject: [Freeswitch-users] FreeSWITCH - Skype capacity Message-ID: Before moving to production, i am just collecting few information's not to get surprise in future. Therefore, can anyone provide estimated suggestion, how many channels/node can make server 100% reserve? System plan: =========== - Single Quad Core Intel Xeon Processor uit de 5400-reeks met maximaal 3,16 GHz - 8GM RAM Question: ======= 1. Can i handle 100 or 300 simultaneous calls for Skype IN and Skype Out using FS? Thank you Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/5fa02bd0/attachment.html From shamun.toha at gmail.com Sun Aug 8 09:04:10 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 8 Aug 2010 18:04:10 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: Am i correct now? I am connected. Tips for dummies like me: ===================== To setup skype user1, please follow this spoon feeding (documentation made me really confuse, just to understand this part). 1. We are not doing it via second PC, 1 PC and 1 head, so: [sun at example configs]$ /usr/bin/skype & [1] 5554 2. Connect to point 1 GUI, that we started [sun at example configs]$ ./skypopen_auth $DISPLAY Skype instance found on display ':0.0', with id #102760508 RECEIVED==> ERROR 68 RECEIVED==> OK ^C 3. Show time [sun at example configs]$ ./skypopen_auth Skype instance found on display ':0.0', with id #102760508 RECEIVED==> OK RECEIVED==> PROTOCOL 6 RECEIVED==> CONNSTATUS ONLINE RECEIVED==> CURRENTUSERHANDLE __lol___ RECEIVED==> USERSTATUS ONLINE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/0a3fff1a/attachment.html From shamun.toha at gmail.com Sun Aug 8 09:56:03 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 8 Aug 2010 18:56:03 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: After the first *make && make install*, copy mod_skypopen/configs/alsa/dummy.c to alsa-driver-1.0.20sound/drivers/dummy.c In doc its mention to do second make && make install , [!] But i dont find it in my system. [root at example /]# locate dummy.c /home/sun/Downloads/freeswitch-1.0.6/libs/tiff-3.8.2/port/dummy.c /home/sun/Downloads/freeswitch-1.0.6/libs/xmlrpc-c/src/cpp/test/testclient_dummy.cpp /home/sun/Downloads/freeswitch-1.0.6/libs/xmlrpc-c/src/test/client_dummy.c /home/sun/Downloads/freeswitch-1.0.6/src/mod/endpoints/mod_skypopen/alsa/alsa-driver-1.0.20-dummy.c /home/sun/Downloads/freeswitch-1.0.6/src/mod/endpoints/mod_skypopen/alsa/dummy.c /home/sun/Downloads/pjsip/pjproject-1.6/pjlib/src/pj/ioqueue_dummy.c /home/sun/Downloads/pjsip/pjproject-1.6/pjlib/src/pj/sock_qos_dummy.c /home/sun/Downloads/pjsip/pjproject-1.6/pjsip-apps/build/dummy.c /home/sun/Downloads/pjsip/trunk/pjlib/src/pj/ioqueue_dummy.c /home/sun/Downloads/pjsip/trunk/pjlib/src/pj/sock_qos_dummy.c /home/sun/Downloads/pjsip/trunk/pjsip-apps/build/dummy.c /opt/qtsdk-2010.04/qt/doc/src/classes/qpatternistdummy.cpp /opt/qtsdk-2010.04/qt/src/3rdparty/freetype/src/autofit/afdummy.c /opt/qtsdk-2010.04/qt/src/3rdparty/libtiff/port/dummy.c -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/df3a636d/attachment.html From shamun.toha at gmail.com Sun Aug 8 09:59:41 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 8 Aug 2010 18:59:41 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: alsa/dummy.c got it [root at example mod_skypopen]# tree . ??? alsa ? ??? alsa-driver-1.0.20-dummy.c -> ./dummy.c ? ??? dummy.c ??? asterisk ? ??? chan_skypiax.c ? ??? cyg_no_pthread_kill.c ? ??? Makefile ? ??? README ? ??? skypiax.conf ? ??? skypiax.h ??? configs ? ??? client.c ? ??? copy ? ??? create ? ??? multiple-instance-same-skype-username ? ? ??? multi.sh ? ? ??? README ? ? ??? skypopen.conf.xml ? ??? README.skypopen_auth ? ??? skypopen_auth ? ??? skypopen_auth.c ? ??? skypopen.conf.xml ? ??? startskype.bat ? ??? startskype.sh ? ??? wait.bat ? ??? windows-service ? ??? startskype.cmd ? ??? wait.cmd ??? Makefile ??? Makefile.am ??? Makefile.in ??? mod_skypopen.2008.vcproj ??? mod_skypopen.c ??? mod_skypopen.la ??? mod_skypopen_la-mod_skypopen.lo ??? mod_skypopen_la-mod_skypopen.o ??? mod_skypopen_la-skypopen_protocol.lo ??? mod_skypopen_la-skypopen_protocol.o ??? mod_skypopen.log ??? README ??? skypopen.h ??? skypopen_protocol.c 5 directories, 37 files [root at example mod_skypopen]# -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/94be4f67/attachment.html From shamun.toha at gmail.com Sun Aug 8 11:39:44 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 8 Aug 2010 20:39:44 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: OK - this far i came alone, can you please kindly tell me what is this now? I am not getting this... [root at example configs]# sh ./startskype2.sh ERROR: Module snd_hda_intel is in use ERROR: Module snd_dummy does not exist in /proc/modules (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) (EE) config/hal: NewInputDeviceRequest failed (2) [root at example configs]# -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/c3e9eb9b/attachment.html From shamun.toha at gmail.com Sun Aug 8 11:49:44 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 8 Aug 2010 20:49:44 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: NOW - completely error?? Can you please guide, what is that mean????? FreeSWITCH Version 1.0.6 (svn-exported) Started. Max Sessions[1000] Session Rate[30] SQL [Enabled] freeswitch at example> load mod_skypopen 2010-08-08 20:46:40.318122 [WARNING] mod_skypopen.c:1447 rev [(nil)|37 ][WARNINGA 1447 ][interface1][-1, 0, 0] STARTING interface_id=1 2010-08-08 20:46:40.322064 [ERR] skypopen_protocol.c:1430 rev [(nil)|37 ][ERRORA 1430 ][none ][-1,-1,-1] Received error code 3 from X Server 2010-08-08 20:46:40.322064 [ERR] skypopen_protocol.c:1483 rev [(nil)|37 ][ERRORA 1483 ][interface1][-1, 0, 0] Sending message failed with status 3 2010-08-08 20:46:40.322064 [ERR] skypopen_protocol.c:1611 rev [(nil)|37 ][ERRORA 1611 ][interface1][-1, 0, 0] Sending message failed - probably Skype crashed. Please run/restart Skype manually and launch Skypopen again -ERR [module load file routine returned an error] 2010-08-08 20:46:40.518886 [NOTICE] mod_skypopen.c:1472 rev [(nil)|37 ][NOTICA 1472 ][interface1][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2010-08-08 20:46:40.518886 [NOTICE] mod_skypopen.c:1481 rev [(nil)|37 ][NOTICA 1481 ][interface1][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==__lol__ 2010-08-08 20:46:40.518886 [ERR] mod_skypopen.c:1523 rev [(nil)|37 ][ERRORA 1523 ][interface1][-1, 0, 0] The Skype client to which we are connected FAILED to gave us CURRENTUSERHANDLE=__lol__, interface_id=1 FAILED to start. No Skype client logged in as '__lol__' has been found. Please (re)launch a Skype client logged in as '__lol__'. Skypopen exiting now 2010-08-08 20:46:40.518886 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_skypopen.so **Module load routine returned an error** freeswitch at example> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/f8c54317/attachment.html From Nabble at slickdeals.endjunk.com Sun Aug 8 13:31:05 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Sun, 8 Aug 2010 13:31:05 -0700 (PDT) Subject: [Freeswitch-users] Modems In-Reply-To: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> Message-ID: <1281299465390-5386969.post@n2.nabble.com> Ken Gillett wrote: > Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, > i.e. be able to accept incoming PSTN calls and also make PSTN calls? >From a hardware point of view, I believe this is possible. For instance, take a closer look at some DSL modem/router and a VoIP Gateway/Router that is based on a TI AR7 platform. Chances are their hardware designs are the same and also use the same (Legerity) chipset, yet their functions are different due to their drivers and software applications. That said, if you can manage to find the source code for your hardware and are willing to invest your time on working on it, I am sure you will be able to succeed to make your DSL modem acting as a VoIP device. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Modems-tp5386739p5386969.html Sent from the freeswitch-users mailing list archive at Nabble.com. From errotan at elder.hu Sun Aug 8 15:04:19 2010 From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Mon, 9 Aug 2010 00:04:19 +0200 Subject: [Freeswitch-users] FS frequent crashes when using txfax In-Reply-To: References: Message-ID: <201008090004.19633.errotan@elder.hu> Hi. Please read this: http://wiki.freeswitch.org/wiki/Reporting_Bugs 2010. augusztus 8. 09:38:47 d?tummal Anurag Srivastava az al?bbiakat ?rta: > Segfault in fs that happens pretty consistently (but not always) when using > txfax. Attaching the backtrace from gdb on the core file and the log on > fs_cli. http://pastebin.freeswitch.org/13602 > I did a "make current" to get on the latest version. Running on ubuntu > hardy > thanks > Anurag > > git log. > -------- > commit 9b4ab6f87798b8e7a65d05357ee7fcd82551cfd5 > Date: Fri Aug 6 11:23:31 2010 +0200 > > gdb core backtrace > ---------------------------- > (gdb) bt > #0 0xb7b88869 in strcasecmp () from /lib/tls/i686/cmov/libc.so.6 > #1 0xb6b9d093 in negotiate_t38 (pvt=0x82e41e0) at mod_spandsp_fax.c:760 > #2 0xb6ba1c06 in mod_spandsp_fax_process_fax (session=0x82dbca8, > .... From errotan at elder.hu Sun Aug 8 15:08:15 2010 From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Mon, 9 Aug 2010 00:08:15 +0200 Subject: [Freeswitch-users] FreeSWITCH - Skype capacity In-Reply-To: References: Message-ID: <201008090008.15118.errotan@elder.hu> HI. Nobody can tell you how many calls your system can handle. You must try it. You can find some hint here: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Performance_and_Resource_Usage 2010. augusztus 8. 17:05:18 d?tummal Shamun toha md az al?bbiakat ?rta: > Before moving to production, i am just collecting few information's not to > get surprise in future. Therefore, can anyone provide estimated suggestion, > how many channels/node can make server 100% reserve? > > System plan: > =========== > - Single Quad Core Intel Xeon Processor uit de 5400-reeks met maximaal 3,16 > GHz > - 8GM RAM > > Question: > ======= > 1. Can i handle 100 or 300 simultaneous calls for Skype IN and Skype Out > using FS? > > > Thank you > Best regards From errotan at elder.hu Sun Aug 8 15:18:14 2010 From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Mon, 9 Aug 2010 00:18:14 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype In-Reply-To: References: Message-ID: <201008090018.14510.errotan@elder.hu> Hi. Please provide some config files posted to pastebin so we can help. It would be better if you could join us on freenode so we can discuss further. 2010. augusztus 8. 20:49:44 d?tummal Shamun toha md az al?bbiakat ?rta: > NOW - completely error?? Can you please guide, what is that mean????? > > FreeSWITCH Version 1.0.6 (svn-exported) Started. > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > freeswitch at example> load mod_skypopen > 2010-08-08 20:46:40.318122 [WARNING] mod_skypopen.c:1447 rev [(nil)|37 > ][WARNINGA 1447 ][interface1][-1, 0, 0] STARTING interface_id=1 > 2010-08-08 20:46:40.322064 [ERR] skypopen_protocol.c:1430 rev [(nil)|37 > ][ERRORA 1430 ][none ][-1,-1,-1] Received error code 3 from X Server > > 2010-08-08 20:46:40.322064 [ERR] skypopen_protocol.c:1483 rev [(nil)|37 > ][ERRORA 1483 ][interface1][-1, 0, 0] Sending message failed with status 3 > 2010-08-08 20:46:40.322064 [ERR] skypopen_protocol.c:1611 rev [(nil)|37 > ][ERRORA 1611 ][interface1][-1, 0, 0] Sending message failed - probably > Skype crashed. Please run/restart Skype manually and launch Skypopen again > > -ERR [module load file routine returned an error] > > 2010-08-08 20:46:40.518886 [NOTICE] mod_skypopen.c:1472 rev [(nil)|37 > ][NOTICA 1472 ][interface1][-1, 0, 0] WAITING roughly 10 seconds to find a > running Skype client and connect to its SKYPE API for interface_id=1 > 2010-08-08 20:46:40.518886 [NOTICE] mod_skypopen.c:1481 rev [(nil)|37 > ][NOTICA 1481 ][interface1][-1, 0, 0] Found a running Skype client, > connected to its SKYPE API for interface_id=1, waiting 60 seconds for > CURRENTUSERHANDLE==__lol__ > 2010-08-08 20:46:40.518886 [ERR] mod_skypopen.c:1523 rev [(nil)|37 > ][ERRORA 1523 ][interface1][-1, 0, 0] The Skype client to which we are > connected FAILED to gave us CURRENTUSERHANDLE=__lol__, interface_id=1 > FAILED to start. No Skype client logged in as '__lol__' has been found. > Please (re)launch a Skype client logged in as '__lol__'. Skypopen exiting > now 2010-08-08 20:46:40.518886 [CRIT] switch_loadable_module.c:882 Error > Loading module /usr/local/freeswitch/mod/mod_skypopen.so > **Module load routine returned an error** > freeswitch at example> From fs-list at communicatefreely.net Sun Aug 8 16:24:18 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Sun, 08 Aug 2010 19:24:18 -0400 Subject: [Freeswitch-users] new install on freebsd 8.1 In-Reply-To: References: Message-ID: <4C5F3CA2.3080203@communicatefreely.net> Hmm. I'm running FS an FreeBSD 8.0 and I used those same instructions. It put a bin folder in /usr/local/freeswitch (which I then added to my path). It looks like something went wrong with the make install part of the process. You have the samples and sounds, but not any of the actual work directories. There is more than just bin, you should also have scripts, db, lib, include, and others. All the directories made from compiled objects seem to be missing. Are you sure that you didn't miss a step? -Tim D Cubi wrote: > Forgot to add URL of instructions followed in original posting to list :( > > Followed these instructions. > http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD > All seemed to have installed properly, but can't find bin folder : > > # pwd > /usr/local/freeswitch > # ls -l > total 8 > drwxr-xr-x 11 root wheel 512 Aug 2 11:24 conf > drwxr-xr-x 2 root wheel 512 Aug 2 11:24 htdocs > drwxr-xr-x 2 root wheel 512 Aug 2 12:44 lib > drwxr-xr-x 4 root wheel 512 Aug 2 12:12 sounds > > Does anyone have any idea where I went wrong? > Thnx > > Darren > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs-list at communicatefreely.net Sun Aug 8 16:44:36 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Sun, 08 Aug 2010 19:44:36 -0400 Subject: [Freeswitch-users] Distributing SIP registrations using ODBC In-Reply-To: References: Message-ID: <4C5F4164.5070104@communicatefreely.net> Yes, that solution will fix it just fine. There is another way, just in case it makes more sense for you: If you have a NAT device that has a proper SIP ALG in it, (such as any of the Netopia 3300-ENT routers), they will accept traffic from either server. The other way that I'm about to try (I'm in the same boat, with all my clients behind NAT), is to use a simple SIP proxy / load balancer in front of Freeswitch. FS will send SIP back through the load balancer to the client, so the traffic will originate from the load balancer IP, no matter which FS box originates. As long as each client registers through the same proxy each time (the correct SRV record will ensure this), the NAT rules won't be a problem. Media isn't as big an issue with Freeswitch, as it will detect the incoming RTP and use that as the return path if the sdp is wrong. -Tim Dan Lane wrote: > Yup, that was how I plan to fix it using LUA to figure out where to > route the call (possibly proxying it via the other FS server). > > Just wanted to check I wasn't re-inventing the wheel before I go ahead > and do it as it seems like the sort of issue other people would have > dealt with. > > On Mon, Aug 2, 2010 at 5:10 PM, Steven Ayre wrote: >> I haven't tried this myself, but routing the call via the FS server they >> registered too might work since it'll be the IP the NAT router is expecting >> traffic from. >> >> -Steve >> >> >> On 2 August 2010 16:10, Dan Lane wrote: >>> Hi Guys, >>> >>> I have a cluster of FS boxes all sharing SIP registrations using ODBC >>> and under certain circumstances I can register Client-A on FS-A and >>> Client-B on FS-B and make calls between Client-A and Client-B without >>> issue as the relevant FS box knows how to get a call to the client >>> from the shared registrations table. >>> >>> However when the callee client is behind NAT (or using a SIP client >>> configured to only allow calls from a server it has registered with) >>> the call gets dropped by the NAT router (or client) >>> >>> I can fix this easily with some LUA but before I do I wanted to check >>> if there was an advisable or recognised FreeSWITCH way to work around >>> this. >>> ro >>> Has anyone else solved this issue already? >>> >>> Regards, >>> Dan >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Sun Aug 8 16:54:37 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 9 Aug 2010 01:54:37 +0200 Subject: [Freeswitch-users] FreeSWITCH - Skype capacity In-Reply-To: References: Message-ID: I don't think you can have so many concurrent skype calls on one machine. The skype client is cpu hungry, and you will need one skype client running for each concurrent call. Anyway, do your tests, but I'm pretty sure that you can have much less than you want. -giovanni On 8/8/10, Shamun toha md wrote: > Before moving to production, i am just collecting few information's not to > get surprise in future. Therefore, can anyone provide estimated suggestion, > how many channels/node can make server 100% reserve? > > System plan: > =========== > - Single Quad Core Intel Xeon Processor uit de 5400-reeks met maximaal 3,16 > GHz > - 8GM RAM > > Question: > ======= > 1. Can i handle 100 or 300 simultaneous calls for Skype IN and Skype Out > using FS? > > > Thank you > Best regards > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From infos at madovsky.org Sun Aug 8 17:26:56 2010 From: infos at madovsky.org (Madovsky) Date: Sun, 8 Aug 2010 20:26:56 -0400 Subject: [Freeswitch-users] FS and kernel PREEMPT_RT in Fedora 10 Message-ID: Just tried PREEMPT_RT patch for kernel 2.6.33.7 on Fedora 10 64bits set on 1000HZ. Everything works fine -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/fde17e75/attachment.html From steveu at coppice.org Sun Aug 8 17:38:59 2010 From: steveu at coppice.org (Steve Underwood) Date: Mon, 09 Aug 2010 08:38:59 +0800 Subject: [Freeswitch-users] Modems In-Reply-To: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> Message-ID: <4C5F4E23.2010202@coppice.org> On 08/09/2010 01:46 AM, Ken Gillett wrote: > I asked this before, but never got any response, so hope someone can help me this time. > > Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. be able to accept incoming PSTN calls and also make PSTN calls? > > I plan on installing FreeSwitch on a Mac and it would be really handy to be able to use Apple's USB modem in this way as otherwise there's no way to add such PSTN functionality on the Mac Mini. Although my requirement is Mac based, I guess this question applies to any FreeSwitch installation. > > > Ken G i l l e t t People constantly ask this same question, but nobody wants to write the drivers. It shouldn't be that hard. There are packages to use many winmodems as modems on Linux machines. The DSP code is usually supplied as a binary object, but the interface to the hardware is supplied as source code. Modify that source, and you could turn it into, say, a DAHDI driver that would hook the device into Asterisk, Callweaver or Freeswitch. Steve From dujinfang at gmail.com Sun Aug 8 18:24:20 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 9 Aug 2010 09:24:20 +0800 Subject: [Freeswitch-users] we are under attack Message-ID: Hi, We suffered an SIP attack from 67.23.236.75. It attempted to register to our SIP server using bruce force. We are running FS on a PC as our office PBX. When all phone failed, we noticed a high CPU load with 90%+ waiting or nice, and in the meantime it used up memory and start swapping to disk. It's a cheap PC with only 700MB memory, and we are running FS, DB, Rails and other system on it. So it took me some time to check every part. And it didn't help even I did a full server reboot. Finally I turned on sip trace in FS and found thousands and millions of illegal registers. And then I blocked the IP in iptables. During the hard time, I noticed: 1) It stucks on one CPU even I have 2 core since sofia-sip is single threaded ? 2) CPU also waiting page swap when used up memory. 3) After I dropped all packets from that IP, FS still kept sending register error sip messages for quite a long time before I restarted FS. Now looking to add http://wiki.freeswitch.org/wiki/Fail2ban, hope this helps . Hope this helps if some one also suffered this. 7. -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From infos at madovsky.org Sun Aug 8 19:34:42 2010 From: infos at madovsky.org (Madovsky) Date: Sun, 8 Aug 2010 22:34:42 -0400 Subject: [Freeswitch-users] we are under attack References: Message-ID: <48D32C41F1DC43609A4C119F105E1809@MOBILEE1705> fail2ban is works like a charm for that... ----- Original Message ----- From: "Seven Du" To: "freeswitch-users" Sent: Sunday, August 08, 2010 9:24 PM Subject: [Freeswitch-users] we are under attack Hi, We suffered an SIP attack from 67.23.236.75. It attempted to register to our SIP server using bruce force. We are running FS on a PC as our office PBX. When all phone failed, we noticed a high CPU load with 90%+ waiting or nice, and in the meantime it used up memory and start swapping to disk. It's a cheap PC with only 700MB memory, and we are running FS, DB, Rails and other system on it. So it took me some time to check every part. And it didn't help even I did a full server reboot. Finally I turned on sip trace in FS and found thousands and millions of illegal registers. And then I blocked the IP in iptables. During the hard time, I noticed: 1) It stucks on one CPU even I have 2 core since sofia-sip is single threaded ? 2) CPU also waiting page swap when used up memory. 3) After I dropped all packets from that IP, FS still kept sending register error sip messages for quite a long time before I restarted FS. Now looking to add http://wiki.freeswitch.org/wiki/Fail2ban, hope this helps . Hope this helps if some one also suffered this. 7. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From shamun.toha at gmail.com Sun Aug 8 14:09:38 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 8 Aug 2010 23:09:38 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype call route Message-ID: A receive calls, B receive calls, they are operator (mod_skypopen) - Caller party 1 calls to A - A receive the call talk for 2 minute - A decide he needs to send the call two B How can A tell caller party 1 (instant), please hold on, i am forwarding this call to B? Can i do this? Is it done via dial plane? Can you please just hints me? Thank you Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/0c46311c/attachment.html From marian at jozep.com.au Sun Aug 8 18:36:56 2010 From: marian at jozep.com.au (marian szczepkowski) Date: Mon, 09 Aug 2010 11:36:56 +1000 Subject: [Freeswitch-users] delay in establishing a call through the event interface Message-ID: <4C5F5BB8.8060903@jozep.com.au> Hi I did some testing with freeswitch and I have an appreciable delay between issueing an originate and the resulting call attempts by the server. Is there a way to shorten this delay down or is this fixed in some wierd form? From paul.gore.j at gmail.com Sun Aug 8 19:38:50 2010 From: paul.gore.j at gmail.com (paul gore) Date: Sun, 8 Aug 2010 22:38:50 -0400 Subject: [Freeswitch-users] Leg_timeout question Message-ID: Hi there, I am trying to accomplish a pretty simple thing in FS dialplan - ring one or multiple destinations with individual timeout per destination, if nobody answers - forward to a voicemail. Here is dial plan I use: What happens is incoming call from my provider rings ext 1002, but instead of going to the VM after 20 sec and no answer FS sends 602 (alloted_timeout) and the ring repeats again and so 3 times, after which call dies. But never goes to voice mail. What I am doing wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100808/9f7ddc13/attachment.html From gmaruzz at celliax.org Sun Aug 8 22:09:47 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 9 Aug 2010 07:09:47 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype call route In-Reply-To: References: Message-ID: Look at the skypopen wiki page, there is a paragraph for messaging ( chat ). -giovanni On 8/8/10, Shamun toha md wrote: > A receive calls, B receive calls, they are operator (mod_skypopen) > - Caller party 1 calls to A > - A receive the call talk for 2 minute > - A decide he needs to send the call two B > > > How can A tell caller party 1 (instant), please hold on, i am forwarding > this call to B? > > Can i do this? Is it done via dial plane? Can you please just hints me? > > Thank you > Best regards > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From brian at microcomaustralia.com.au Sun Aug 8 22:10:29 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Mon, 9 Aug 2010 15:10:29 +1000 Subject: [Freeswitch-users] freetdm dialtone Message-ID: Hello, Since upgrading to the latest freeswitch and freetdm, TDM400 based card, the dialtone behavior seems to have changed. Now when I dial a number, the dial tone continues unchanged. I normally expect the dialtone should stop when I start dialing the number - this is kind of disconcerting and makes it appear that it isn't registering the number I am dialing when it is. Any ideas? Thanks -- Brian May From mustafa.pk at gmail.com Sun Aug 8 22:14:00 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Mon, 9 Aug 2010 10:14:00 +0500 Subject: [Freeswitch-users] FreeSwitch - Skype call route In-Reply-To: References: Message-ID: Yes, please read more about how xml dialplan logic works. it's very easy On Mon, Aug 9, 2010 at 2:09 AM, Shamun toha md wrote: > A receive calls, B receive calls, they are operator (mod_skypopen) > - Caller party 1 calls to A > - A receive the call talk for 2 minute > - A decide he needs to send the call two B > > > How can A tell caller party 1 (instant), please hold on, i am forwarding > this call to B? > > Can i do this? Is it done via dial plane? Can you please just hints me? > > Thank you > Best regards > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/31db3a25/attachment.html From gmaruzz at celliax.org Sun Aug 8 22:16:30 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 9 Aug 2010 07:16:30 +0200 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Just to let you know, I'm making progresses on this, got a prototype working, need more work and tests, some more days. -giovanni On 7/28/10, Jason Jeffords wrote: > Giovanni, > > No worries, thank you for all your help. We will work on other things > until > you are ready. > > Have a great holiday, > > Jason > > On Wed, Jul 28, 2010 at 9:30 AM, Giovanni Maruzzelli > wrote: > >> Jason, >> >> sorry for the inconvenience. >> >> I had a look (a long look) at the problem, and no way I can fix it in >> a decent way right now because of how the code is presently structured >> and because of my tools here (trying to operate on a remote server >> with a wobbly bluetooth connection via the cellphone from a tiny >> laptop). >> >> Also, I will need to do extensive testing after changing the answering >> mechanism, to be sure it will not break the various cases (many >> channels with same username, many channels with different usernames, >> earlymedia, etc etc). >> >> So, please be patient until next week and I'll give it a thoroughly >> solution. >> >> Bye for now, >> >> -giovanni >> >> On Wed, Jul 28, 2010 at 12:11 AM, Giovanni Maruzzelli >> wrote: >> > Wup, I was confused, hehe. >> > I'll fix it tomorrow, while the kid is doing his own things with his >> > friends (certain times I just don't realize he's big boy now ;) ). >> > -giovanni >> > >> > On 7/27/10, Giovanni Maruzzelli wrote: >> >> Hi Jason >> >> >> >> that is mod_skypopen fault (my fault). >> >> >> >> mod_skypopen directly answers an incoming call, without taking into >> >> account if it is directed to do it or not. >> >> >> >> I've also filed a Jira to myself about it, but not yet taken care of. >> >> >> >> I will fix it as soon as possible, but probably not before net week >> >> (I'm in holyday with the son right now, he would not like that so >> >> much). >> >> >> >> Sorry for making you and anthm waste time in debugging an inexplicable >> >> behavior, >> >> >> >> -giovanni >> >> >> >> >> >> On Tue, Jul 27, 2010 at 8:24 PM, Jason Jeffords >> >> wrote: >> >>> Hi Anthony, >> >>> It looks like mod_skypopen is answering the Skype call and opening the >> >>> channel before any >> >>> dialplan lookups are reached :( >> >>> Here is part of the log: >> >>> 2010-07-27 18:07:28.553971 [DEBUG] skypopen_protocol.c:176 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 176 >> >>> ][interface1][-1, 0, 0] READING: |||CALL 2114 CONF_ID 0||| >> >>> 2010-07-27 18:07:28.563966 [DEBUG] skypopen_protocol.c:176 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 176 >> >>> ][interface1][-1, 0, 0] READING: |||USER jjeffords.com TIMEZONE >> 72000||| >> >>> 2010-07-27 18:07:28.633980 [DEBUG] skypopen_protocol.c:176 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 176 >> >>> ][interface1][-1, 0, 0] READING: |||CALL 2114 STATUS RINGING||| >> >>> 2010-07-27 18:07:28.633980 [DEBUG] skypopen_protocol.c:556 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 556 >> >>> ][interface1][-1, 0, 0] NO ACTIVE calls in this moment, skype_call >> 2114 >> >>> is >> >>> RINGING, to ask PARTNER_HANDLE >> >>> 2010-07-27 18:07:28.633980 [DEBUG] skypopen_protocol.c:1498 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 1498 >> >>> ][interface1][-1, 0, 0] SENDING: |||GET CALL 2114 PARTNER_HANDLE|||| >> >>> 2010-07-27 18:07:28.663964 [DEBUG] skypopen_protocol.c:176 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 176 >> >>> ][interface1][-1, 0, 0] READING: |||CALL 2114 PARTNER_HANDLE >> >>> jjeffords.com||| >> >>> 2010-07-27 18:07:28.663964 [DEBUG] skypopen_protocol.c:481 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 481 >> >>> ][interface1][-1, 0, 0] Call 2114 TRY ANSWER >> >>> 2010-07-27 18:07:28.663964 [DEBUG] mod_skypopen.c:2301 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 2301 >> >>> ][interface1][-1, 0, 0] NOT FOUND >> >>> 2010-07-27 18:07:28.663964 [DEBUG] skypopen_protocol.c:1498 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 1498 >> >>> ][interface1][-1, 0, 0] SENDING: |||GET CALL 2114 PARTNER_DISPNAME|||| >> >>> 2010-07-27 18:07:28.683968 [DEBUG] skypopen_protocol.c:1498 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 1498 >> >>> ][interface1][-1,11, 0] SENDING: |||ALTER CALL 2114 ANSWER|||| >> >>> 2010-07-27 18:07:28.703965 [DEBUG] mod_skypopen.c:2314 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 2314 >> >>> ][interface1][-1,11, 0] We answered a Skype RING on skype_call 2114 >> >>> 2010-07-27 18:07:28.703965 [DEBUG] mod_skypopen.c:2322 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 2322 >> >>> ][interface1][-1,11, 0] NEW! name: interface1, state: 11, >> >>> value=jjeffords.com, tech_pvt->callid_number=jjeffords.com, >> >>> tech_pvt->skype_user=cloud-tree-admin >> >>> 2010-07-27 18:07:28.703965 [DEBUG] skypopen_protocol.c:176 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 176 >> >>> ][interface1][-1,11, 0] READING: |||CALL 2114 PARTNER_DISPNAME Jason >> >>> Jeffords||| >> >>> 2010-07-27 18:07:28.714033 [DEBUG] skypopen_protocol.c:176 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 176 >> >>> ][interface1][-1,11, 0] READING: |||ALTER CALL 2114 ANSWER||| >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:176 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 176 >> >>> ][interface1][-1,11, 0] READING: |||CALL 2114 STATUS INPROGRESS||| >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:660 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 660 >> >>> ][interface1][-1,11, 0] no tech_pvt->session_uuid_str >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:666 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 666 >> >>> ][interface1][-1,11, 0] skype_call: 2114 is now active >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:673 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 673 >> >>> ][interface1][-1, 5,21] START start_audio_threads >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:83 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 83 >> >>> ][interface1][-1, 5,21] Binded! *which_port=32769, >> >>> tech_pvt->tcp_cli_port=32770, tech_pvt->tcp_srv_port=32769 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:88 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 88 >> >>> ][interface1][-1, 5,21] 1 SO_RCVBUF is 87380, size is 4 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:92 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 92 >> >>> ][interface1][-1, 5,21] 1 SO_SNDBUF is 16384, size is 4 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:113 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 113 >> >>> ][interface1][-1, 5,21] 2 SO_RCVBUF is 87380, size is 4 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:132 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 132 >> >>> ][interface1][-1, 5,21] 2 SO_SNDBUF is 16384, size is 4 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:136 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 136 >> >>> ][interface1][-1, 5,21] TCP_NODELAY is 0 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:143 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 143 >> >>> ][interface1][-1, 5,21] TCP_NODELAY is 0 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:773 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 773 >> >>> ][interface1][-1, 5,21] started tcp_srv_thread thread. >> >>> 2010-07-27 18:07:28.733950 [DEBUG] mod_skypopen.c:1889 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 1889 >> >>> ][interface1][-1, 5,21] started tcp_srv_thread thread. >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:83 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 83 >> >>> ][interface1][-1, 5,21] Binded! *which_port=32770, >> >>> tech_pvt->tcp_cli_port=32770, tech_pvt->tcp_srv_port=32769 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:88 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 88 >> >>> ][interface1][-1, 5,21] 1 SO_RCVBUF is 87380, size is 4 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:92 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 92 >> >>> ][interface1][-1, 5,21] 1 SO_SNDBUF is 16384, size is 4 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:113 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 113 >> >>> ][interface1][-1, 5,21] 2 SO_RCVBUF is 87380, size is 4 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:132 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 132 >> >>> ][interface1][-1, 5,21] 2 SO_SNDBUF is 16384, size is 4 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:136 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 136 >> >>> ][interface1][-1, 5,21] TCP_NODELAY is 0 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:143 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 143 >> >>> ][interface1][-1, 5,21] TCP_NODELAY is 0 >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:924 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 924 >> >>> ][interface1][-1, 5,21] started tcp_cli_thread thread. >> >>> 2010-07-27 18:07:28.733950 [DEBUG] mod_skypopen.c:1899 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 1899 >> >>> ][interface1][-1, 5,21] started tcp_cli_thread thread. >> >>> 2010-07-27 18:07:28.853968 [DEBUG] skypopen_protocol.c:1498 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 1498 >> >>> ][interface1][-1, 5,21] SENDING: |||ALTER CALL 2114 SET_INPUT >> >>> PORT="32770"|||| >> >>> 2010-07-27 18:07:28.864023 [DEBUG] skypopen_protocol.c:953 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 953 >> >>> ][interface1][-1, 5,21] ACCEPTED here you send me 32770 >> >>> 2010-07-27 18:07:28.864023 [DEBUG] skypopen_protocol.c:958 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 958 >> >>> ][interface1][-1, 5,21] 4 SO_RCVBUF is 87380, size is 4 >> >>> 2010-07-27 18:07:28.864023 [DEBUG] skypopen_protocol.c:962 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 962 >> >>> ][interface1][-1, 5,21] 4 SO_SNDBUF is 16384, size is 4 >> >>> 2010-07-27 18:07:28.873942 [DEBUG] skypopen_protocol.c:1498 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 1498 >> >>> ][interface1][-1, 5,21] SENDING: |||#output ALTER CALL 2114 SET_OUTPUT >> >>> PORT="32769"|||| >> >>> 2010-07-27 18:07:28.883950 [DEBUG] skypopen_protocol.c:688 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 688 >> >>> ][interface1][-1, 5,21] New Inbound Channel! >> >>> >> >>> >> >>> 2010-07-27 18:07:28.883950 [DEBUG] mod_skypopen.c:1920 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 1920 >> >>> ][interface1][-1, 5,21] 2 SESSION_REQUEST >> >>> d18a03ae-99a9-11df-9273-dfe9044f32b8 >> >>> 2010-07-27 18:07:28.883950 [DEBUG] mod_skypopen.c:258 rev >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 258 >> >>> ][interface1][-1, 5,21] skypopen_tech_init SUCCESS >> >>> 2010-07-27 18:07:28.883950 [NOTICE] switch_channel.c:776 New Channel >> >>> skypopen/interface1 [d18a03ae-99a9-11df-9273-dfe9044f32b8] >> >>> 2010-07-27 18:07:28.883950 [DEBUG] mod_skypopen.c:1944 >> >>> (skypopen/interface1) >> >>> State Change CS_NEW -> CS_INIT >> >>> >> >>> >> >>> >> >>> On Tue, Jul 27, 2010 at 2:06 PM, Anthony Minessale >> >>> wrote: >> >>>> >> >>>> don't answer the A leg? bridge to B with it unanswered and it should >> do >> >>>> what you want unless somehow the skype channel answers explicitly >> which >> >>>> would be a possible bug in mod_skype_open >> >>>> >> >>>> On Tue, Jul 27, 2010 at 9:43 AM, Jason Jeffords >> >>>> wrote: >> >>>>> >> >>>>> Hi Anthony, >> >>>>> Thank you for your very fast response and your excellent work on >> >>>>> Freeswitch :) >> >>>>> We have tried setting ignore_early_media=true in the bridge dial >> string >> >>>>> (the B >> >>>>> leg). This has the expected effect of removing ringing from the >> >>>>> channel >> >>>>> (the >> >>>>> A leg no longer hears ringing, neither Skype nor PSTN). >> >>>>> The problem is the A leg has answered the Skype call (mod_skypopen) >> and >> >>>>> established a channel before the B leg has answered. We want to tie >> >>>>> these >> >>>>> call state machines together to allow the A leg to continue to Skype >> >>>>> ring >> >>>>> until >> >>>>> the B leg answers. >> >>>>> We tried setting ignore_early_media on the A leg as well in several >> >>>>> different >> >>>>> locations in the dialplan without success. >> >>>>> Do you have another other suggestions? Is there a different >> >>>>> variable >> >>>>> that >> >>>>> allows the state machines to be tied together as described above? >> >>>>> Thanks for your help, >> >>>>> Jason >> >>>>> >> >>>>> >> >>>>> On Thu, Jul 22, 2010 at 6:01 PM, Anthony Minessale >> >>>>> wrote: >> >>>>>> >> >>>>>> add {ignore_early_media=true} to your bridge dial string >> >>>>>> >> >>>>>> On Thu, Jul 22, 2010 at 3:51 PM, Jason Jeffords < >> jason at cloudtree.net> >> >>>>>> wrote: >> >>>>>>> >> >>>>>>> My specific use case is an inbound Skype call using mod_skypopen >> >>>>>>> to >> a >> >>>>>>> SIP phone. >> >>>>>>> I would like the Skype call to keep ringing as a Skype call (not a >> >>>>>>> PSTN >> >>>>>>> call) >> >>>>>>> until the B-leg is answered. The default behaviour is the Skype >> call >> >>>>>>> is answered >> >>>>>>> by freeswitch and ringback is played over the established Skype >> call. >> >>>>>>> Thanks in advance, >> >>>>>>> Jason >> >>>>>>> >> >>>>>>> >> >>>>>>> _______________________________________________ >> >>>>>>> FreeSWITCH-users mailing list >> >>>>>>> FreeSWITCH-users at lists.freeswitch.org >> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>>>> >> >>>>>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>>> http://www.freeswitch.org >> >>>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> -- >> >>>>>> Anthony Minessale II >> >>>>>> >> >>>>>> FreeSWITCH http://www.freeswitch.org/ >> >>>>>> ClueCon http://www.cluecon.com/ >> >>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >> >>>>>> >> >>>>>> AIM: anthm >> >>>>>> MSN:anthony_minessale at hotmail.com >> >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >>>>>> IRC: irc.freenode.net #freeswitch >> >>>>>> >> >>>>>> FreeSWITCH Developer Conference >> >>>>>> sip:888 at conference.freeswitch.org >> >>>>>> googletalk:conf+888 at conference.freeswitch.org >> >>>>>> pstn:+19193869900 >> >>>>>> >> >>>>>> _______________________________________________ >> >>>>>> FreeSWITCH-users mailing list >> >>>>>> FreeSWITCH-users at lists.freeswitch.org >> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>> http://www.freeswitch.org >> >>>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> _______________________________________________ >> >>>>> FreeSWITCH-users mailing list >> >>>>> FreeSWITCH-users at lists.freeswitch.org >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>> http://www.freeswitch.org >> >>>>> >> >>>> >> >>>> >> >>>> >> >>>> -- >> >>>> Anthony Minessale II >> >>>> >> >>>> FreeSWITCH http://www.freeswitch.org/ >> >>>> ClueCon http://www.cluecon.com/ >> >>>> Twitter: http://twitter.com/FreeSWITCH_wire >> >>>> >> >>>> AIM: anthm >> >>>> MSN:anthony_minessale at hotmail.com >> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >>>> IRC: irc.freenode.net #freeswitch >> >>>> >> >>>> FreeSWITCH Developer Conference >> >>>> sip:888 at conference.freeswitch.org >> >>>> googletalk:conf+888 at conference.freeswitch.org >> >>>> pstn:+19193869900 >> >>>> >> >>>> _______________________________________________ >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >>> >> >> >> >> >> >> >> >> -- >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> Cell : +39-347-2665618 >> >> >> > >> > -- >> > Sent from my mobile device >> > >> > Sincerely, >> > >> > Giovanni Maruzzelli >> > Cell : +39-347-2665618 >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From terrymr at gmail.com Mon Aug 9 00:03:51 2010 From: terrymr at gmail.com (Terry Moore-Read) Date: Mon, 9 Aug 2010 00:03:51 -0700 Subject: [Freeswitch-users] freetdm dialtone In-Reply-To: References: Message-ID: I hate to send me too posts, but I just noticed the same thing. Sent from my iPad On Aug 8, 2010, at 10:10 PM, Brian May wrote: > Hello, > > Since upgrading to the latest freeswitch and freetdm, TDM400 based > card, the dialtone behavior seems to have changed. > > Now when I dial a number, the dial tone continues unchanged. I > normally expect the dialtone should stop when I start dialing the > number - this is kind of disconcerting and makes it appear that it > isn't registering the number I am dialing when it is. > > Any ideas? > > Thanks > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jan.berger at video24.no Mon Aug 9 00:03:58 2010 From: jan.berger at video24.no (Jan Berger) Date: Mon, 9 Aug 2010 09:03:58 +0200 Subject: [Freeswitch-users] we are under attack In-Reply-To: References: Message-ID: <02416836BE07480BB29CEA6BF0606D04@dell9400> 1) It stucks on one CPU even I have 2 core since sofia-sip is single threaded ? Your FS uses a single process, and the operating systems need multiple processes to fully utilize multiple cores. It's major difference between threads and processes on this. Jan From lists at infosecurity.ch Mon Aug 9 00:28:35 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Mon, 09 Aug 2010 09:28:35 +0200 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: Message-ID: <4C5FAE23.3000406@infosecurity.ch> That's mono-directional streaming or bi-directional streaming? I mean: UserA: Browser via Web connected trough Flash on Red5 User B: SIP client via PC connected trough SIP on FS Those clients can communicate bidirectionally in realtime like a voice call or it's just a mono-directional from SIP Client to to Web client? On 07/08/10 22.13, Richard Alam wrote: > Yes, this can be done. > > We are doing this for voice conference using Flash, Red5 and > FreeSWITHCH/Asterisk. > > We basically have a SIP client based on Red5Phone. The SIP client > takes audiodata from RTMP, transcodes Nellymoser to Ulaw, then send > the audio data using RTP to FS/Asterisk. > > With FS, we can use Speex Wideband so there is no need to transcode > but to just take audio data from RTMP then shove it into an RTP packet > to FS. > > You can take a look at the code here > http://github.com/bigbluebutton/bigbluebutton/tree/master/bbb-voice/. > > Let me know if you have any questions. > > Richard > > On Fri, Aug 6, 2010 at 8:01 AM, Dennis wrote: > >> hi, >> >> we are currently playing with streaming calls. >> >> with fs and icecast (over mod_shout) it works quite well. the problem >> is, that http-streaming to a website in conjuction with a flashplayer, >> a lot of overhead is produced. therefore this is no optimal solution. >> >> the best for streaming to a flashplayer embedded in a website are >> rtmp-streams (which are also used by webradios). red5 is an open >> source streaming server (written in java), which supports rtmp-streams >> and mp3. >> >> the problem seems to be, that fs "only" supports icecast and >> shoutcast. we can not get fs to work with red5. >> >> does someone have experiences with fs and red5 and can tell me, if >> there might be a way to get it working? or are there any technical >> issues, why this can't work? >> >> >> thanks and kind regards >> dennis >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > From steveayre at gmail.com Mon Aug 9 00:40:25 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 9 Aug 2010 08:40:25 +0100 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <29331510.post@talk.nabble.com> References: <29331510.post@talk.nabble.com> Message-ID: Just found that parameter was listed but undocumented on the Wiki, i've fixed that. http://wiki.freeswitch.org/wiki/Sofia.conf.xml#apply-proxy-acl -Steve On 7 August 2010 15:31, jprsa wrote: > > Hello, > > Were you able to finally get your UAs to register on your FS boxes behind > the proxy (opensip) server? > > > > Thanks > > > > Bill W-3 wrote: > > > > That's fantastic! FreeSWITCH ROCKS! > > > > I'll update the wiki. > > > > Thanks, > > Bill > > > > > > > > Brian West wrote: > >> use "apply-proxy-acl" on the sofia profile. > >> > >> /b > >> > >> On Dec 15, 2009, at 10:58 PM, Bill W wrote: > >> > >>> However, having the proxy in the path effectively negates using IP > >>> based > >>> ACLS. > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/ACLs-through-proxy-tp26806529p29331510.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/4a56ca29/attachment.html From jaybinks at gmail.com Mon Aug 9 01:19:43 2010 From: jaybinks at gmail.com (jay binks) Date: Mon, 9 Aug 2010 18:19:43 +1000 Subject: [Freeswitch-users] we are under attack In-Reply-To: <02416836BE07480BB29CEA6BF0606D04@dell9400> References: <02416836BE07480BB29CEA6BF0606D04@dell9400> Message-ID: THIS : http://etel.wiki.oreilly.com/wiki/index.php/SIP_DoS/DDoS_Mitigation also works quite well for these attacks.. they wont stop the brute force ( you will need to configure fail2ban for that ) but this will setup iptables to rate limit any sip connections, so your pc wont get that loaded. you can thank Kristian Kielhofner ( a member of the FS community ) for that neat little script :) maybe we can copy that script ( Kristian ?? ) to the Freeswitch wiki.. ( with a few modifications ).. be careful of the STOP command on that script its defective ! ( and will wipe all firewall rules, especially dangerous if you drop all by default ) Jay On Mon, Aug 9, 2010 at 5:03 PM, Jan Berger wrote: > > 1) It stucks on one CPU even I have 2 core since sofia-sip is single > threaded ? > > Your FS uses a single process, and the operating systems need multiple > processes to fully utilize multiple cores. It's major difference between > threads and processes on this. > > Jan > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/642a435f/attachment.html From foxb at abv.bg Mon Aug 9 03:42:27 2010 From: foxb at abv.bg (Hristo Benev) Date: Mon, 9 Aug 2010 13:42:27 +0300 (EEST) Subject: [Freeswitch-users] we are under attack Message-ID: <850670260.77710.1281350547995.JavaMail.apache@mail22.abv.bg> If it is just one IP it is not that bad... First block it in iptables, then file a complain with your ISP(they will help you block at entry point) and his ISP. Fail2 ban works well with brute force...(or at least forces them to extent the attack) >-------- ?????????? ????? -------- >??: Seven Du >???????: [Freeswitch-users] we are under attack >??: freeswitch-users >????????? ??: ??????????, 2010, ?????? 9 04:24:20 EEST >Hi, > >We suffered an SIP attack from 67.23.236.75. It attempted to register >to our SIP server using bruce force. > >We are running FS on a PC as our office PBX. When all phone failed, >we noticed a high CPU load with 90%+ waiting or nice, and in the >meantime it used up memory and start swapping to disk. > >It's a cheap PC with only 700MB memory, and we are running FS, DB, >Rails and other system on it. So it took me some time to check every >part. And it didn't help even I did a full server reboot. Finally I >turned on sip trace in FS and found thousands and millions of illegal >registers. And then I blocked the IP in iptables. > >During the hard time, I noticed: > >1) It stucks on one CPU even I have 2 core since sofia-sip is single threaded ? > >2) CPU also waiting page swap when used up memory. > >3) After I dropped all packets from that IP, FS still kept sending >register error sip messages for quite a long time before I restarted >FS. > >Now looking to add http://wiki.freeswitch.org/wiki/Fail2ban, hope this helps . > >Hope this helps if some one also suffered this. > >7. > >-- >Blog: http://www.dujinfang.com >Proj:? http://www.freeswitch.org.cn > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > From 12ukwn at gmail.com Mon Aug 9 04:26:38 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Mon, 9 Aug 2010 13:26:38 +0200 Subject: [Freeswitch-users] FS performance In-Reply-To: References: <4C5C54F0.6000000@gmail.com> <201008061445.19345.sos@sokhapkin.dyndns.org> <4C5C638E.3060002@gmail.com> <201008061612.36416.sos@sokhapkin.dyndns.org> <4C5C7F9E.9040400@gmail.com> Message-ID: <20100809132638.2e7d3e67@anubis.defcon1> Le Sun, 8 Aug 2010 00:17:19 -0400, Mathieu Rene a ?crit : > Hi, > > While FreeSWICH is a back-to-back user agent (B2BUA), you can do some > magic on sip<->sip calls. > > Kamailio wont do any media proxying on its own, to do the same with FS, > you need to use bypass_media > (http://wiki.freeswitch.org/wiki/Bypass_media) > > That will use considerably less resources than relaying an rtp packet > every 20ms (the default setting). In fact, it wont use anything at all > once the call is up. > > Mathieu Rene Thanks Mat, I now see the differences. In fact I think my best solution would be a melting between Kamailio and FS. -- A closed mouth gathers no foot. From moises.silva at gmail.com Mon Aug 9 06:27:53 2010 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 9 Aug 2010 09:27:53 -0400 Subject: [Freeswitch-users] freetdm dialtone In-Reply-To: References: Message-ID: I'll look into it and post back the results. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com On Mon, Aug 9, 2010 at 1:10 AM, Brian May wrote: > Hello, > > Since upgrading to the latest freeswitch and freetdm, TDM400 based > card, the dialtone behavior seems to have changed. > > Now when I dial a number, the dial tone continues unchanged. I > normally expect the dialtone should stop when I start dialing the > number - this is kind of disconcerting and makes it appear that it > isn't registering the number I am dialing when it is. > > Any ideas? > > Thanks > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/113b2872/attachment.html From ash at archerdrive.com Mon Aug 9 01:52:47 2010 From: ash at archerdrive.com (Ash) Date: Mon, 9 Aug 2010 18:52:47 +1000 Subject: [Freeswitch-users] we are under attack In-Reply-To: References: Message-ID: <4045405D-8AFA-4697-B093-15958006D4D1@archerdrive.com> I had a similar problem a couple of weeks ago. I was hit from IP's in China and Finland. In the end I created an iptables chain to just allow Australian IP's only as we only expect to see Australian IP's, I obtained the list from http://www.ipaddresslocation.org. Then I create a log rule to log any drop connections to /var/log/messages and use nagios check_log to email me if there are any drops, this way I can be sure its only Australian IP's. I also added some rate limiting to ports 5060-5080. To date we have not had anymore issues... Touch wood!. On 09/08/2010, at 11:24 AM, Seven Du wrote: > Hi, > > We suffered an SIP attack from 67.23.236.75. It attempted to register > to our SIP server using bruce force. > > We are running FS on a PC as our office PBX. When all phone failed, > we noticed a high CPU load with 90%+ waiting or nice, and in the > meantime it used up memory and start swapping to disk. > > It's a cheap PC with only 700MB memory, and we are running FS, DB, > Rails and other system on it. So it took me some time to check every > part. And it didn't help even I did a full server reboot. Finally I > turned on sip trace in FS and found thousands and millions of illegal > registers. And then I blocked the IP in iptables. > > During the hard time, I noticed: > > 1) It stucks on one CPU even I have 2 core since sofia-sip is single threaded ? > > 2) CPU also waiting page swap when used up memory. > > 3) After I dropped all packets from that IP, FS still kept sending > register error sip messages for quite a long time before I restarted > FS. > > Now looking to add http://wiki.freeswitch.org/wiki/Fail2ban, hope this helps . > > Hope this helps if some one also suffered this. > > 7. > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at cloudtree.net Mon Aug 9 06:27:41 2010 From: jason at cloudtree.net (Jason Jeffords) Date: Mon, 9 Aug 2010 09:27:41 -0400 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Thank you Giovanni, This is great news :) Let us know if we can do anything to help and to test. Thanks again, Jason On Mon, Aug 9, 2010 at 1:16 AM, Giovanni Maruzzelli wrote: > Just to let you know, I'm making progresses on this, got a prototype > working, need more work and tests, some more days. > -giovanni > > On 7/28/10, Jason Jeffords wrote: > > Giovanni, > > > > No worries, thank you for all your help. We will work on other things > > until > > you are ready. > > > > Have a great holiday, > > > > Jason > > > > On Wed, Jul 28, 2010 at 9:30 AM, Giovanni Maruzzelli > > wrote: > > > >> Jason, > >> > >> sorry for the inconvenience. > >> > >> I had a look (a long look) at the problem, and no way I can fix it in > >> a decent way right now because of how the code is presently structured > >> and because of my tools here (trying to operate on a remote server > >> with a wobbly bluetooth connection via the cellphone from a tiny > >> laptop). > >> > >> Also, I will need to do extensive testing after changing the answering > >> mechanism, to be sure it will not break the various cases (many > >> channels with same username, many channels with different usernames, > >> earlymedia, etc etc). > >> > >> So, please be patient until next week and I'll give it a thoroughly > >> solution. > >> > >> Bye for now, > >> > >> -giovanni > >> > >> On Wed, Jul 28, 2010 at 12:11 AM, Giovanni Maruzzelli > >> wrote: > >> > Wup, I was confused, hehe. > >> > I'll fix it tomorrow, while the kid is doing his own things with his > >> > friends (certain times I just don't realize he's big boy now ;) ). > >> > -giovanni > >> > > >> > On 7/27/10, Giovanni Maruzzelli wrote: > >> >> Hi Jason > >> >> > >> >> that is mod_skypopen fault (my fault). > >> >> > >> >> mod_skypopen directly answers an incoming call, without taking into > >> >> account if it is directed to do it or not. > >> >> > >> >> I've also filed a Jira to myself about it, but not yet taken care of. > >> >> > >> >> I will fix it as soon as possible, but probably not before net week > >> >> (I'm in holyday with the son right now, he would not like that so > >> >> much). > >> >> > >> >> Sorry for making you and anthm waste time in debugging an > inexplicable > >> >> behavior, > >> >> > >> >> -giovanni > >> >> > >> >> > >> >> On Tue, Jul 27, 2010 at 8:24 PM, Jason Jeffords > > >> >> wrote: > >> >>> Hi Anthony, > >> >>> It looks like mod_skypopen is answering the Skype call and opening > the > >> >>> channel before any > >> >>> dialplan lookups are reached :( > >> >>> Here is part of the log: > >> >>> 2010-07-27 18:07:28.553971 [DEBUG] skypopen_protocol.c:176 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 176 > >> >>> ][interface1][-1, 0, 0] READING: |||CALL 2114 CONF_ID 0||| > >> >>> 2010-07-27 18:07:28.563966 [DEBUG] skypopen_protocol.c:176 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 176 > >> >>> ][interface1][-1, 0, 0] READING: |||USER jjeffords.com TIMEZONE > >> 72000||| > >> >>> 2010-07-27 18:07:28.633980 [DEBUG] skypopen_protocol.c:176 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 176 > >> >>> ][interface1][-1, 0, 0] READING: |||CALL 2114 STATUS RINGING||| > >> >>> 2010-07-27 18:07:28.633980 [DEBUG] skypopen_protocol.c:556 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 556 > >> >>> ][interface1][-1, 0, 0] NO ACTIVE calls in this moment, skype_call > >> 2114 > >> >>> is > >> >>> RINGING, to ask PARTNER_HANDLE > >> >>> 2010-07-27 18:07:28.633980 [DEBUG] skypopen_protocol.c:1498 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 1498 > >> >>> ][interface1][-1, 0, 0] SENDING: |||GET CALL 2114 PARTNER_HANDLE|||| > >> >>> 2010-07-27 18:07:28.663964 [DEBUG] skypopen_protocol.c:176 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 176 > >> >>> ][interface1][-1, 0, 0] READING: |||CALL 2114 PARTNER_HANDLE > >> >>> jjeffords.com||| > >> >>> 2010-07-27 18:07:28.663964 [DEBUG] skypopen_protocol.c:481 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 481 > >> >>> ][interface1][-1, 0, 0] Call 2114 TRY ANSWER > >> >>> 2010-07-27 18:07:28.663964 [DEBUG] mod_skypopen.c:2301 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 2301 > >> >>> ][interface1][-1, 0, 0] NOT FOUND > >> >>> 2010-07-27 18:07:28.663964 [DEBUG] skypopen_protocol.c:1498 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 1498 > >> >>> ][interface1][-1, 0, 0] SENDING: |||GET CALL 2114 > PARTNER_DISPNAME|||| > >> >>> 2010-07-27 18:07:28.683968 [DEBUG] skypopen_protocol.c:1498 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 1498 > >> >>> ][interface1][-1,11, 0] SENDING: |||ALTER CALL 2114 ANSWER|||| > >> >>> 2010-07-27 18:07:28.703965 [DEBUG] mod_skypopen.c:2314 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 2314 > >> >>> ][interface1][-1,11, 0] We answered a Skype RING on skype_call 2114 > >> >>> 2010-07-27 18:07:28.703965 [DEBUG] mod_skypopen.c:2322 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 2322 > >> >>> ][interface1][-1,11, 0] NEW! name: interface1, state: 11, > >> >>> value=jjeffords.com, tech_pvt->callid_number=jjeffords.com, > >> >>> tech_pvt->skype_user=cloud-tree-admin > >> >>> 2010-07-27 18:07:28.703965 [DEBUG] skypopen_protocol.c:176 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 176 > >> >>> ][interface1][-1,11, 0] READING: |||CALL 2114 PARTNER_DISPNAME > Jason > >> >>> Jeffords||| > >> >>> 2010-07-27 18:07:28.714033 [DEBUG] skypopen_protocol.c:176 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 176 > >> >>> ][interface1][-1,11, 0] READING: |||ALTER CALL 2114 ANSWER||| > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:176 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 176 > >> >>> ][interface1][-1,11, 0] READING: |||CALL 2114 STATUS INPROGRESS||| > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:660 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 660 > >> >>> ][interface1][-1,11, 0] no tech_pvt->session_uuid_str > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:666 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 666 > >> >>> ][interface1][-1,11, 0] skype_call: 2114 is now active > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:673 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 673 > >> >>> ][interface1][-1, 5,21] START start_audio_threads > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:83 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 83 > >> >>> ][interface1][-1, 5,21] Binded! *which_port=32769, > >> >>> tech_pvt->tcp_cli_port=32770, tech_pvt->tcp_srv_port=32769 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:88 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 88 > >> >>> ][interface1][-1, 5,21] 1 SO_RCVBUF is 87380, size is 4 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:92 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 92 > >> >>> ][interface1][-1, 5,21] 1 SO_SNDBUF is 16384, size is 4 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:113 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 113 > >> >>> ][interface1][-1, 5,21] 2 SO_RCVBUF is 87380, size is 4 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:132 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 132 > >> >>> ][interface1][-1, 5,21] 2 SO_SNDBUF is 16384, size is 4 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:136 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 136 > >> >>> ][interface1][-1, 5,21] TCP_NODELAY is 0 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:143 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 143 > >> >>> ][interface1][-1, 5,21] TCP_NODELAY is 0 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:773 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 773 > >> >>> ][interface1][-1, 5,21] started tcp_srv_thread thread. > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] mod_skypopen.c:1889 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 1889 > >> >>> ][interface1][-1, 5,21] started tcp_srv_thread thread. > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:83 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 83 > >> >>> ][interface1][-1, 5,21] Binded! *which_port=32770, > >> >>> tech_pvt->tcp_cli_port=32770, tech_pvt->tcp_srv_port=32769 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:88 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 88 > >> >>> ][interface1][-1, 5,21] 1 SO_RCVBUF is 87380, size is 4 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:92 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE 92 > >> >>> ][interface1][-1, 5,21] 1 SO_SNDBUF is 16384, size is 4 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:113 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 113 > >> >>> ][interface1][-1, 5,21] 2 SO_RCVBUF is 87380, size is 4 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:132 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 132 > >> >>> ][interface1][-1, 5,21] 2 SO_SNDBUF is 16384, size is 4 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:136 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 136 > >> >>> ][interface1][-1, 5,21] TCP_NODELAY is 0 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:143 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 143 > >> >>> ][interface1][-1, 5,21] TCP_NODELAY is 0 > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] skypopen_protocol.c:924 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 924 > >> >>> ][interface1][-1, 5,21] started tcp_cli_thread thread. > >> >>> 2010-07-27 18:07:28.733950 [DEBUG] mod_skypopen.c:1899 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 1899 > >> >>> ][interface1][-1, 5,21] started tcp_cli_thread thread. > >> >>> 2010-07-27 18:07:28.853968 [DEBUG] skypopen_protocol.c:1498 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 1498 > >> >>> ][interface1][-1, 5,21] SENDING: |||ALTER CALL 2114 SET_INPUT > >> >>> PORT="32770"|||| > >> >>> 2010-07-27 18:07:28.864023 [DEBUG] skypopen_protocol.c:953 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 953 > >> >>> ][interface1][-1, 5,21] ACCEPTED here you send me 32770 > >> >>> 2010-07-27 18:07:28.864023 [DEBUG] skypopen_protocol.c:958 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 958 > >> >>> ][interface1][-1, 5,21] 4 SO_RCVBUF is 87380, size is 4 > >> >>> 2010-07-27 18:07:28.864023 [DEBUG] skypopen_protocol.c:962 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 962 > >> >>> ][interface1][-1, 5,21] 4 SO_SNDBUF is 16384, size is 4 > >> >>> 2010-07-27 18:07:28.873942 [DEBUG] skypopen_protocol.c:1498 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 1498 > >> >>> ][interface1][-1, 5,21] SENDING: |||#output ALTER CALL 2114 > SET_OUTPUT > >> >>> PORT="32769"|||| > >> >>> 2010-07-27 18:07:28.883950 [DEBUG] skypopen_protocol.c:688 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 688 > >> >>> ][interface1][-1, 5,21] New Inbound Channel! > >> >>> > >> >>> > >> >>> 2010-07-27 18:07:28.883950 [DEBUG] mod_skypopen.c:1920 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 1920 > >> >>> ][interface1][-1, 5,21] 2 SESSION_REQUEST > >> >>> d18a03ae-99a9-11df-9273-dfe9044f32b8 > >> >>> 2010-07-27 18:07:28.883950 [DEBUG] mod_skypopen.c:258 rev > >> >>> git2svn-syncpoint-master-132-g91a87e9[(nil)|37 ][DEBUG_SKYPE > 258 > >> >>> ][interface1][-1, 5,21] skypopen_tech_init SUCCESS > >> >>> 2010-07-27 18:07:28.883950 [NOTICE] switch_channel.c:776 New Channel > >> >>> skypopen/interface1 [d18a03ae-99a9-11df-9273-dfe9044f32b8] > >> >>> 2010-07-27 18:07:28.883950 [DEBUG] mod_skypopen.c:1944 > >> >>> (skypopen/interface1) > >> >>> State Change CS_NEW -> CS_INIT > >> >>> > >> >>> > >> >>> > >> >>> On Tue, Jul 27, 2010 at 2:06 PM, Anthony Minessale > >> >>> wrote: > >> >>>> > >> >>>> don't answer the A leg? bridge to B with it unanswered and it > should > >> do > >> >>>> what you want unless somehow the skype channel answers explicitly > >> which > >> >>>> would be a possible bug in mod_skype_open > >> >>>> > >> >>>> On Tue, Jul 27, 2010 at 9:43 AM, Jason Jeffords < > jason at cloudtree.net> > >> >>>> wrote: > >> >>>>> > >> >>>>> Hi Anthony, > >> >>>>> Thank you for your very fast response and your excellent work on > >> >>>>> Freeswitch :) > >> >>>>> We have tried setting ignore_early_media=true in the bridge dial > >> string > >> >>>>> (the B > >> >>>>> leg). This has the expected effect of removing ringing from the > >> >>>>> channel > >> >>>>> (the > >> >>>>> A leg no longer hears ringing, neither Skype nor PSTN). > >> >>>>> The problem is the A leg has answered the Skype call > (mod_skypopen) > >> and > >> >>>>> established a channel before the B leg has answered. We want to > tie > >> >>>>> these > >> >>>>> call state machines together to allow the A leg to continue to > Skype > >> >>>>> ring > >> >>>>> until > >> >>>>> the B leg answers. > >> >>>>> We tried setting ignore_early_media on the A leg as well in > several > >> >>>>> different > >> >>>>> locations in the dialplan without success. > >> >>>>> Do you have another other suggestions? Is there a different > >> >>>>> variable > >> >>>>> that > >> >>>>> allows the state machines to be tied together as described above? > >> >>>>> Thanks for your help, > >> >>>>> Jason > >> >>>>> > >> >>>>> > >> >>>>> On Thu, Jul 22, 2010 at 6:01 PM, Anthony Minessale > >> >>>>> wrote: > >> >>>>>> > >> >>>>>> add {ignore_early_media=true} to your bridge dial string > >> >>>>>> > >> >>>>>> On Thu, Jul 22, 2010 at 3:51 PM, Jason Jeffords < > >> jason at cloudtree.net> > >> >>>>>> wrote: > >> >>>>>>> > >> >>>>>>> My specific use case is an inbound Skype call using mod_skypopen > >> >>>>>>> to > >> a > >> >>>>>>> SIP phone. > >> >>>>>>> I would like the Skype call to keep ringing as a Skype call (not > a > >> >>>>>>> PSTN > >> >>>>>>> call) > >> >>>>>>> until the B-leg is answered. The default behaviour is the Skype > >> call > >> >>>>>>> is answered > >> >>>>>>> by freeswitch and ringback is played over the established Skype > >> call. > >> >>>>>>> Thanks in advance, > >> >>>>>>> Jason > >> >>>>>>> > >> >>>>>>> > >> >>>>>>> _______________________________________________ > >> >>>>>>> FreeSWITCH-users mailing list > >> >>>>>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>>>> > >> >>>>>>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>>>> http://www.freeswitch.org > >> >>>>>>> > >> >>>>>> > >> >>>>>> > >> >>>>>> > >> >>>>>> -- > >> >>>>>> Anthony Minessale II > >> >>>>>> > >> >>>>>> FreeSWITCH http://www.freeswitch.org/ > >> >>>>>> ClueCon http://www.cluecon.com/ > >> >>>>>> Twitter: http://twitter.com/FreeSWITCH_wire > >> >>>>>> > >> >>>>>> AIM: anthm > >> >>>>>> MSN:anthony_minessale at hotmail.com > > > > >> >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> >>>>>> IRC: irc.freenode.net #freeswitch > >> >>>>>> > >> >>>>>> FreeSWITCH Developer Conference > >> >>>>>> sip:888 at conference.freeswitch.org > > > > >> >>>>>> googletalk:conf+888 at conference.freeswitch.org > > > > >> >>>>>> pstn:+19193869900 > >> >>>>>> > >> >>>>>> _______________________________________________ > >> >>>>>> FreeSWITCH-users mailing list > >> >>>>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>>> http://www.freeswitch.org > >> >>>>>> > >> >>>>> > >> >>>>> > >> >>>>> > >> >>>>> > >> >>>>> _______________________________________________ > >> >>>>> FreeSWITCH-users mailing list > >> >>>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>> http://www.freeswitch.org > >> >>>>> > >> >>>> > >> >>>> > >> >>>> > >> >>>> -- > >> >>>> Anthony Minessale II > >> >>>> > >> >>>> FreeSWITCH http://www.freeswitch.org/ > >> >>>> ClueCon http://www.cluecon.com/ > >> >>>> Twitter: http://twitter.com/FreeSWITCH_wire > >> >>>> > >> >>>> AIM: anthm > >> >>>> MSN:anthony_minessale at hotmail.com > > > > >> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> >>>> IRC: irc.freenode.net #freeswitch > >> >>>> > >> >>>> FreeSWITCH Developer Conference > >> >>>> sip:888 at conference.freeswitch.org > > > > >> >>>> googletalk:conf+888 at conference.freeswitch.org > > > > >> >>>> pstn:+19193869900 > >> >>>> > >> >>>> _______________________________________________ > >> >>>> FreeSWITCH-users mailing list > >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>> http://www.freeswitch.org > >> >>>> > >> >>> > >> >>> > >> >>> > >> >>> _______________________________________________ > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >>> > >> >>> > >> >> > >> >> > >> >> > >> >> -- > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> Cell : +39-347-2665618 > >> >> > >> > > >> > -- > >> > Sent from my mobile device > >> > > >> > Sincerely, > >> > > >> > Giovanni Maruzzelli > >> > Cell : +39-347-2665618 > >> > > >> > >> > >> > >> -- > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > -- > Sent from my mobile device > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/0832138c/attachment-0001.html From jesse at cronomagic.com Mon Aug 9 06:06:20 2010 From: jesse at cronomagic.com (Jesse Cloutier) Date: Mon, 09 Aug 2010 09:06:20 -0400 Subject: [Freeswitch-users] Custom Headers Message-ID: <4C5FFD4C.2050409@cronomagic.com> Hi all, Is there a way to read custom sip headers (eg: P-MyHeader or X-MyHeader) in the freeswitch dialplan? I am using freeswitch as a SBC and would like to pass certain variables to it for the cdr. Thanks! Jesse Cloutier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/7d9316f8/attachment.html From prayersts at gmail.com Mon Aug 9 05:22:11 2010 From: prayersts at gmail.com (Tae-Sung Shin) Date: Mon, 9 Aug 2010 08:22:11 -0400 Subject: [Freeswitch-users] Calling internal extensions to/from outside is acting strange Message-ID: <012301cb37bd$7f399d60$7dacd820$@com> Hello Guys First of all, I am a new user of Freeswitch. I spent a couple of days on this issue and am desperate for some help. Briefly, my problems are 1. Calling out from SPA2102 (ext 1002) or a softphone (phonerlite) (ext 1003) via gateway voip is disconnected as soon as it got answered after ringing in the other side. 2. Calling from outside is disconnected 30 seconds after it got answered. I don't have this problem with calls between the internal extensions. Without extensions (direct communication between gateway and SPA2102), I verified SPA2102 is working fine My environment: --- --- As Freeswitch wiki, suggested, I have following xml contents . Sip_profiles/external/voipms.xml . Dialplan/public/voipms.xml I think I went through all wiki and other articles on the web but could find a resolution for this issue. I would appreciate if you give me any hint. Thanks Tae-Sung Shin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/96d2da0b/attachment.html From brian at freeswitch.org Mon Aug 9 06:34:55 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Aug 2010 08:34:55 -0500 Subject: [Freeswitch-users] Custom Headers In-Reply-To: <4C5FFD4C.2050409@cronomagic.com> References: <4C5FFD4C.2050409@cronomagic.com> Message-ID: Have you used the info app to see that most if not all those are already turned into variables for you to use. ;) /b On Aug 9, 2010, at 8:06 AM, Jesse Cloutier wrote: > Hi all, > Is there a way to read custom sip headers (eg: P-MyHeader or X-MyHeader) in the freeswitch dialplan? > > I am using freeswitch as a SBC and would like to pass certain variables to it for the cdr. > > Thanks! > Jesse Cloutier From sos at sokhapkin.dyndns.org Mon Aug 9 06:36:15 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 9 Aug 2010 09:36:15 -0400 Subject: [Freeswitch-users] Custom Headers In-Reply-To: <4C5FFD4C.2050409@cronomagic.com> References: <4C5FFD4C.2050409@cronomagic.com> Message-ID: <201008090936.15325.sos@sokhapkin.dyndns.org> ${sip_h_X-MyHeader} On Monday 09 August 2010, Jesse Cloutier wrote: > Hi all, > Is there a way to read custom sip headers (eg: P-MyHeader or X-MyHeader) > in the freeswitch dialplan? > > I am using freeswitch as a SBC and would like to pass certain variables > to it for the cdr. > > Thanks! > Jesse Cloutier > From ritzalam at gmail.com Mon Aug 9 06:53:46 2010 From: ritzalam at gmail.com (Richard Alam) Date: Mon, 9 Aug 2010 09:53:46 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: <4C5FAE23.3000406@infosecurity.ch> References: <4C5FAE23.3000406@infosecurity.ch> Message-ID: Full duplex Flash -> Red5 -> FS and vice-versa. Richard On Mon, Aug 9, 2010 at 3:28 AM, Fabio Pietrosanti (naif) wrote: > That's mono-directional streaming or bi-directional streaming? > > I mean: > UserA: Browser via Web connected trough Flash on Red5 > User B: SIP client via PC connected trough SIP on FS > > Those clients can communicate bidirectionally in realtime like a voice > call or it's just a mono-directional from SIP Client to to Web client? > > On 07/08/10 22.13, Richard Alam wrote: >> Yes, this can be done. >> >> We are doing this for voice conference using Flash, Red5 and >> FreeSWITHCH/Asterisk. >> >> We basically have a SIP client based on Red5Phone. The SIP client >> takes audiodata from RTMP, transcodes Nellymoser to Ulaw, then send >> the audio data using RTP to FS/Asterisk. >> >> With FS, we can use Speex Wideband so there is no need to transcode >> but to just take audio data from RTMP then shove it into an RTP packet >> to FS. >> >> You can take a look at the code here >> http://github.com/bigbluebutton/bigbluebutton/tree/master/bbb-voice/. >> >> Let me know if you have any questions. >> >> Richard >> >> On Fri, Aug 6, 2010 at 8:01 AM, Dennis wrote: >> >>> hi, >>> >>> we are currently playing with streaming calls. >>> >>> with fs and icecast (over mod_shout) it works quite well. the problem >>> is, that http-streaming to a website in conjuction with a flashplayer, >>> a lot of overhead is produced. therefore this is no optimal solution. >>> >>> the best for streaming to a flashplayer embedded in a website are >>> rtmp-streams (which are also used by webradios). red5 is an open >>> source streaming server (written in java), which supports rtmp-streams >>> and mp3. >>> >>> the problem seems to be, that fs "only" supports icecast and >>> shoutcast. we can not get fs to work with red5. >>> >>> does someone have experiences with fs and red5 and can tell me, if >>> there might be a way to get it working? or are there any technical >>> issues, why this can't work? >>> >>> >>> thanks and kind regards >>> dennis >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton From moises.silva at gmail.com Mon Aug 9 07:13:09 2010 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 9 Aug 2010 10:13:09 -0400 Subject: [Freeswitch-users] freetdm dialtone In-Reply-To: References: Message-ID: I just fixed it in git head. Please try it and let me know if you find any other quirk. Thanks! Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com On Mon, Aug 9, 2010 at 9:27 AM, Moises Silva wrote: > I'll look into it and post back the results. > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON > L3R 9R6 Canada > t. 1 905 474 1990 x128 | e. moy at sangoma.com > > > > On Mon, Aug 9, 2010 at 1:10 AM, Brian May wrote: > >> Hello, >> >> Since upgrading to the latest freeswitch and freetdm, TDM400 based >> card, the dialtone behavior seems to have changed. >> >> Now when I dial a number, the dial tone continues unchanged. I >> normally expect the dialtone should stop when I start dialing the >> number - this is kind of disconcerting and makes it appear that it >> isn't registering the number I am dialing when it is. >> >> Any ideas? >> >> Thanks >> -- >> Brian May >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/247d48e2/attachment.html From kris at kriskinc.com Mon Aug 9 07:23:21 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 9 Aug 2010 09:23:21 -0500 Subject: [Freeswitch-users] we are under attack In-Reply-To: References: <02416836BE07480BB29CEA6BF0606D04@dell9400> Message-ID: The current version of the script and a little background can be found here: http://blog.krisk.org/2008/07/sip-dosddos-mitigation.html I have no problem linking the FS wiki to either of these but the eTel wiki will probably get taken down eventually. As far as the issue with the stop command... I remember us talking about it but I don't remember the specific problem. Care to refresh my memory? On Mon, Aug 9, 2010 at 3:19 AM, jay binks wrote: > THIS :?http://etel.wiki.oreilly.com/wiki/index.php/SIP_DoS/DDoS_Mitigation > also works quite well for these attacks.. > they wont stop the brute force ( you will need to configure fail2ban for > that ) > but this will setup iptables to rate limit any sip connections, so your pc > wont get that loaded. > you can thank?Kristian Kielhofner ( a member of the FS community ) for that > neat little script :) > maybe we can copy that script ( Kristian ?? ) to the Freeswitch wiki.. > ( with a few modifications ).. > be careful of the STOP command on that script its defective ! > ( and will wipe all firewall rules, especially dangerous if you drop all by > default ) > Jay > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From vipkilla at gmail.com Mon Aug 9 07:51:15 2010 From: vipkilla at gmail.com (vip killa) Date: Mon, 9 Aug 2010 10:51:15 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> Message-ID: this would be ideal if the CELT codec could be implemented!! cd quality voice chat via browser!! On Mon, Aug 9, 2010 at 9:53 AM, Richard Alam wrote: > Full duplex > > Flash -> Red5 -> FS and vice-versa. > > Richard > > On Mon, Aug 9, 2010 at 3:28 AM, Fabio Pietrosanti (naif) > wrote: > > That's mono-directional streaming or bi-directional streaming? > > > > I mean: > > UserA: Browser via Web connected trough Flash on Red5 > > User B: SIP client via PC connected trough SIP on FS > > > > Those clients can communicate bidirectionally in realtime like a voice > > call or it's just a mono-directional from SIP Client to to Web client? > > > > On 07/08/10 22.13, Richard Alam wrote: > >> Yes, this can be done. > >> > >> We are doing this for voice conference using Flash, Red5 and > >> FreeSWITHCH/Asterisk. > >> > >> We basically have a SIP client based on Red5Phone. The SIP client > >> takes audiodata from RTMP, transcodes Nellymoser to Ulaw, then send > >> the audio data using RTP to FS/Asterisk. > >> > >> With FS, we can use Speex Wideband so there is no need to transcode > >> but to just take audio data from RTMP then shove it into an RTP packet > >> to FS. > >> > >> You can take a look at the code here > >> http://github.com/bigbluebutton/bigbluebutton/tree/master/bbb-voice/. > >> > >> Let me know if you have any questions. > >> > >> Richard > >> > >> On Fri, Aug 6, 2010 at 8:01 AM, Dennis wrote: > >> > >>> hi, > >>> > >>> we are currently playing with streaming calls. > >>> > >>> with fs and icecast (over mod_shout) it works quite well. the problem > >>> is, that http-streaming to a website in conjuction with a flashplayer, > >>> a lot of overhead is produced. therefore this is no optimal solution. > >>> > >>> the best for streaming to a flashplayer embedded in a website are > >>> rtmp-streams (which are also used by webradios). red5 is an open > >>> source streaming server (written in java), which supports rtmp-streams > >>> and mp3. > >>> > >>> the problem seems to be, that fs "only" supports icecast and > >>> shoutcast. we can not get fs to work with red5. > >>> > >>> does someone have experiences with fs and red5 and can tell me, if > >>> there might be a way to get it working? or are there any technical > >>> issues, why this can't work? > >>> > >>> > >>> thanks and kind regards > >>> dennis > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > --- > BigBlueButton > http://www.bigbluebutton.org > http://code.google.com/p/bigbluebutton > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/882af089/attachment.html From jan.berger at video24.no Mon Aug 9 07:11:18 2010 From: jan.berger at video24.no (Jan Berger) Date: Mon, 9 Aug 2010 16:11:18 +0200 Subject: [Freeswitch-users] GPL Wins Again Message-ID: Interesting article about GPL and lawsuits http://www.linuxplanet.com/linuxplanet/reports/7145/1/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/76a9500b/attachment.html From brian at freeswitch.org Mon Aug 9 08:21:43 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Aug 2010 10:21:43 -0500 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: References: Message-ID: too bad westinghouse is going to go belly up. /b On Aug 9, 2010, at 9:11 AM, Jan Berger wrote: > Interesting article about GPL and lawsuits > > http://www.linuxplanet.com/linuxplanet/reports/7145/1/ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tgraziano at myitdepartment.net Mon Aug 9 08:30:42 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Mon, 9 Aug 2010 11:30:42 -0400 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: References: Message-ID: On Mon, Aug 9, 2010 at 11:21 AM, Brian West wrote: > too bad westinghouse is going to go belly up. > > (and after all Tesla did for the company to help them compete against edison too, using all original thinking and creating things, seems they have lost their way a long time ago) > /b > > On Aug 9, 2010, at 9:11 AM, Jan Berger wrote: > > > Interesting article about GPL and lawsuits > > > > http://www.linuxplanet.com/linuxplanet/reports/7145/1/ > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/5545dac8/attachment-0001.html From robert.hadley at teotech.com Mon Aug 9 08:46:13 2010 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 9 Aug 2010 08:46:13 -0700 Subject: [Freeswitch-users] Leg_timeout question In-Reply-To: References: Message-ID: <97D5CE44FD7A4B7F9D58FC89BF18809F@greyhawk.tonecommander.com> Try _____ From: paul gore [mailto:paul.gore.j at gmail.com] Sent: Sunday, August 08, 2010 7:39 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Leg_timeout question Hi there, I am trying to accomplish a pretty simple thing in FS dialplan - ring one or multiple destinations with individual timeout per destination, if nobody answers - forward to a voicemail. Here is dial plan I use: What happens is incoming call from my provider rings ext 1002, but instead of going to the VM after 20 sec and no answer FS sends 602 (alloted_timeout) and the ring repeats again and so 3 times, after which call dies. But never goes to voice mail. What I am doing wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/e9848a0b/attachment.html From infos at madovsky.org Mon Aug 9 08:46:31 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 9 Aug 2010 11:46:31 -0400 Subject: [Freeswitch-users] GPL Wins Again References: Message-ID: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> Open source starts to be Open Fools ----- Original Message ----- From: Tony Graziano To: FreeSWITCH Users Help Sent: Monday, August 09, 2010 11:30 AM Subject: Re: [Freeswitch-users] GPL Wins Again On Mon, Aug 9, 2010 at 11:21 AM, Brian West wrote: too bad westinghouse is going to go belly up. (and after all Tesla did for the company to help them compete against edison too, using all original thinking and creating things, seems they have lost their way a long time ago) /b On Aug 9, 2010, at 9:11 AM, Jan Berger wrote: > Interesting article about GPL and lawsuits > > http://www.linuxplanet.com/linuxplanet/reports/7145/1/ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/4eddd6c6/attachment.html From brian at freeswitch.org Mon Aug 9 08:54:50 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Aug 2010 10:54:50 -0500 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> Message-ID: <1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> Can we please learn to properly reply to posts? Highlight and only include the parts you're replying to. These 10 mile long threads with the footer 100 times gets kinda old! ;) /b On Aug 9, 2010, at 10:46 AM, Madovsky wrote: > Open source starts to be Open Fools From infos at madovsky.org Mon Aug 9 09:08:06 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 9 Aug 2010 12:08:06 -0400 Subject: [Freeswitch-users] GPL Wins Again References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> <1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> Message-ID: <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> I read well don't worry, I say that because companies start to instrumentalize open source. so after years some developers (for google summer source code for ex) realise that they spent all their lifetime for a big company for free... that is why I called Open Fools.... ----- Original Message ----- From: "Brian West" To: "FreeSWITCH Users Help" Sent: Monday, August 09, 2010 11:54 AM Subject: Re: [Freeswitch-users] GPL Wins Again > Can we please learn to properly reply to posts? Highlight and only > include the parts you're replying to. These 10 mile long threads with the > footer 100 times gets kinda old! ;) > > /b > > On Aug 9, 2010, at 10:46 AM, Madovsky wrote: > >> Open source starts to be Open Fools > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From terrymr at gmail.com Mon Aug 9 09:24:15 2010 From: terrymr at gmail.com (Terry Moore-Read) Date: Mon, 9 Aug 2010 09:24:15 -0700 Subject: [Freeswitch-users] freetdm dialtone In-Reply-To: References: Message-ID: Wow that's fast - Can't test until tonight. I'll let you know. Terry On Mon, Aug 9, 2010 at 7:13 AM, Moises Silva wrote: > I just fixed it in git head.? Please try it and let me know if you find any > other quirk. > > Thanks! > > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R > 9R6 Canada > t. 1 905 474 1990 x128 | e. moy at sangoma.com > > > On Mon, Aug 9, 2010 at 9:27 AM, Moises Silva wrote: >> >> I'll look into it and post back the results. >> >> Moises Silva >> Senior Software Engineer >> Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON >> L3R 9R6 Canada >> t. 1 905 474 1990 x128 | e. moy at sangoma.com >> >> >> On Mon, Aug 9, 2010 at 1:10 AM, Brian May >> wrote: >>> >>> Hello, >>> >>> Since upgrading to the latest freeswitch and freetdm, TDM400 based >>> card, the dialtone behavior seems to have changed. >>> >>> Now when I dial a number, the dial tone continues unchanged. I >>> normally expect the dialtone should stop when I start dialing the >>> number - this is kind of disconcerting and makes it appear that it >>> isn't registering the number I am dialing when it is. >>> >>> Any ideas? >>> >>> Thanks >>> -- >>> Brian May >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mranga at gmail.com Mon Aug 9 09:24:43 2010 From: mranga at gmail.com (M. Ranganathan) Date: Mon, 9 Aug 2010 12:24:43 -0400 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> <1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> Message-ID: On Mon, Aug 9, 2010 at 12:08 PM, Madovsky wrote: > I read well don't worry, > I say that because companies start to instrumentalize open source. > so after years some developers (for google summer source code for ex) > realise that they spent all their lifetime for a big company for free... > that is why I called Open Fools.... > > > ----- Original Message ----- I speak for myself (as an individual) here : I disagree with your characterization. Open source is engineers collaborating informally on a shared piece of basic infrastructure code so as to amortize engineering cost, testing cost and beta testing. As an open source developer, you give away something to get a lot in return so you can build on the sum total of collaborative effort. Open source is also an excellent way for students to establish a track record. Yes it is free labor but try getting a job these days as a fresh graduate with no experience and you will see what I am talking about. Ranga -- M. Ranganathan From lists at infosecurity.ch Mon Aug 9 09:27:12 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Mon, 09 Aug 2010 18:27:12 +0200 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> Message-ID: <4C602C60.4010803@infosecurity.ch> Wow! Are there some ready-made, very simple, opensource Flash client to use Red5 along with FS for web telephony? Fabio On 09/08/10 15.53, Richard Alam wrote: > Full duplex > > Flash -> Red5 -> FS and vice-versa. > > Richard > From infos at madovsky.org Mon Aug 9 09:59:58 2010 From: infos at madovsky.org (Madovsky) Date: Mon, 9 Aug 2010 12:59:58 -0400 Subject: [Freeswitch-users] GPL Wins Again References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705><1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> Message-ID: <2F9511642CCD492CA057DB89F09BF420@MOBILEE1705> its' also an excellent way for big companies to avoid to pay employees to make all the dev work also.... ----- Original Message ----- From: "M. Ranganathan" To: "FreeSWITCH Users Help" Sent: Monday, August 09, 2010 12:24 PM Subject: Re: [Freeswitch-users] GPL Wins Again > On Mon, Aug 9, 2010 at 12:08 PM, Madovsky wrote: >> I read well don't worry, >> I say that because companies start to instrumentalize open source. >> so after years some developers (for google summer source code for ex) >> realise that they spent all their lifetime for a big company for free... >> that is why I called Open Fools.... >> >> >> ----- Original Message ----- > > I speak for myself (as an individual) here : > > I disagree with your characterization. Open source is engineers > collaborating informally on a shared piece of basic infrastructure > code so as to amortize engineering cost, testing cost and beta > testing. As an open source developer, you give away something to get a > lot in return so you can build on the sum total of collaborative > effort. > > Open source is also an excellent way for students to establish a track > record. Yes it is free labor but try getting a job these days as a > fresh graduate with no experience and you will see what I am talking > about. > > Ranga > > > -- > M. Ranganathan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgende at gendesign.com Mon Aug 9 11:16:52 2010 From: mgende at gendesign.com (Michael Gende) Date: Mon, 9 Aug 2010 13:16:52 -0500 Subject: [Freeswitch-users] Asterisk to Freeswitch Message-ID: Hello, I was wondering if someone could please give me a shove in the right direction concerning some "translations". I've got some dial plan examples for a vendor's handset that are created for Asterisk. Naturally, we use FreeSwitch. Any "dial plan translation" pages to be found? I found the Rosetta Stone on the FS site. Very comprehensive, but I didn't see things for the commands in my examples (i.e, Asterisk's Pickup = FS command X) I'll figure it out but wondered if someone else had already been down this road. Regards, Mike G. P.S. Nice new FS book, by the way. Got our copy last week, very instructive and a good read to boot. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/7aca5743/attachment.html From macedoslm at gmail.com Mon Aug 9 11:47:22 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Mon, 9 Aug 2010 15:47:22 -0300 Subject: [Freeswitch-users] Video Support Message-ID: Hi, I've already read that FS supports video, but I've never seen anyone using or talking about how to use it. Does FS support video conferences? Or only video calls? Is there any documentation about this subject? Thanks, -- Samuel Macedo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/2cf15520/attachment.html From ken at ukgb.net Mon Aug 9 11:49:26 2010 From: ken at ukgb.net (Ken Gillett) Date: Mon, 9 Aug 2010 19:49:26 +0100 Subject: [Freeswitch-users] Modems In-Reply-To: References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> Message-ID: <197186CA-C15C-4F73-82A0-FB219FF9E2EA@ukgb.net> Well that's the alternative, but I confess to being unclear on one aspect of such devices. In a simple scenario of several SIP devices registered to a SIP server plus a PSTN analog line, it is relatively simple to see how PSTN to VOIP bridging can be implemented with a gateway device that includes an FXS, FXO and ethernet port. An incoming call on the PSTN causes the gateway to initiate a call via the SIP server to SIP client devices registered on that ID and then putting the 2 ends of the call together so the PSTN call is 'bridged' onto VOIP and being handled by a SIP client device, i.e. a softphone or SIP telephone. But what about the reverse? My requirement is not only for the above basic ATA, but to be able to bridge VOIP to PSTN. IOW so that a SIP client can make an outgoing call onto the PSTN. Both the gateway and the SIP client are registered to a SIP server (same one in fact), how does the SIP client place a call onto the PSTN? The gateway is not a SIP server, what is the process by which a SIP client can tell such a gateway to place a call on the PSTN to the number that is somehow provided to the gateway by the SIP client? I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation. I'd be grateful for any advice about this. Thanks. On 8 Aug 2010, at 20:49, Tony Graziano wrote: > You can always use a standalone sip gateway (patton, audiocodes, etc.). > > On 8/8/10, Ken Gillett wrote: >> I asked this before, but never got any response, so hope someone can help me >> this time. >> >> Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. >> be able to accept incoming PSTN calls and also make PSTN calls? >> >> I plan on installing FreeSwitch on a Mac and it would be really handy to be >> able to use Apple's USB modem in this way as otherwise there's no way to add >> such PSTN functionality on the Mac Mini. Although my requirement is Mac >> based, I guess this question applies to any FreeSwitch installation. Ken G i l l e t t _/_/_/_/_/_/_/_/ From jesse at cronomagic.com Mon Aug 9 06:47:37 2010 From: jesse at cronomagic.com (Jesse Cloutier) Date: Mon, 09 Aug 2010 09:47:37 -0400 Subject: [Freeswitch-users] Custom Headers In-Reply-To: <201008090936.15325.sos@sokhapkin.dyndns.org> References: <4C5FFD4C.2050409@cronomagic.com> <201008090936.15325.sos@sokhapkin.dyndns.org> Message-ID: <4C6006F9.5010409@cronomagic.com> Perfect!, after more digging I found this line in the documentation of for mod sofia All inbound SIP calls will install any X- headers into local variables but I wasnt sure about the syntax, Thanks! On 08/09/2010 09:36 AM, Sergey Okhapkin wrote: > ${sip_h_X-MyHeader} > > On Monday 09 August 2010, Jesse Cloutier wrote: >> Hi all, >> Is there a way to read custom sip headers (eg: P-MyHeader or X-MyHeader) >> in the freeswitch dialplan? >> >> I am using freeswitch as a SBC and would like to pass certain variables >> to it for the cdr. >> >> Thanks! >> Jesse Cloutier >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/6048103c/attachment.html From John.Hermanski at dialogic.com Mon Aug 9 12:08:02 2010 From: John.Hermanski at dialogic.com (John Hermanski) Date: Mon, 9 Aug 2010 15:08:02 -0400 Subject: [Freeswitch-users] Video Support In-Reply-To: References: Message-ID: <9053AD1B23AABE42A65FB0B31B1E1CE7022F34B0C3@MBX.dialogic.com> Hi Samuel, Project Diastar from Dialogic has been doing video plays, records, transcoding and conferencing with Asterisk for quite some time now. We have recently produced a "mod_woomera" to work with FS and showed it last week at ClueCon. While it's still in a somewhat green state, it is in our nightly builds and is available for testing. Please take a look at www.projectdiastar.org for more details. Thanks, John Hermanski Technical Marketing Engineer Dialogic From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samuel Macedo Sent: Monday, August 09, 2010 2:47 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Video Support Hi, I've already read that FS supports video, but I've never seen anyone using or talking about how to use it. Does FS support video conferences? Or only video calls? Is there any documentation about this subject? Thanks, -- Samuel Macedo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/fc0336c6/attachment.html From macedoslm at gmail.com Mon Aug 9 13:04:47 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Mon, 9 Aug 2010 17:04:47 -0300 Subject: [Freeswitch-users] Video Support In-Reply-To: <9053AD1B23AABE42A65FB0B31B1E1CE7022F34B0C3@MBX.dialogic.com> References: <9053AD1B23AABE42A65FB0B31B1E1CE7022F34B0C3@MBX.dialogic.com> Message-ID: Hi John, Can I test the "mod_woomera" without a DiaStar Server? Thanks, -- Samuel Macedo On 9 August 2010 16:08, John Hermanski wrote: > Hi Samuel, > > > > Project Diastar from Dialogic has been doing video plays, records, > transcoding and conferencing with Asterisk for quite some time now. We have > recently produced a ?mod_woomera? to work with FS and showed it last week at > ClueCon. While it?s still in a somewhat green state, it is in our nightly > builds and is available for testing. > > > > Please take a look at www.projectdiastar.org for more details. > > > > Thanks, > > > > John Hermanski > > Technical Marketing Engineer > > Dialogic > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Samuel > Macedo > *Sent:* Monday, August 09, 2010 2:47 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Video Support > > > > Hi, > > > > I've already read that FS supports video, but I've never seen anyone using > or talking about how to use it. > > Does FS support video conferences? Or only video calls? Is there any > documentation about this subject? > > > > Thanks, > > -- > > Samuel Macedo > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/213f179f/attachment-0001.html From msc at freeswitch.org Mon Aug 9 13:43:42 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Aug 2010 13:43:42 -0700 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: <2F9511642CCD492CA057DB89F09BF420@MOBILEE1705> References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> <1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> <2F9511642CCD492CA057DB89F09BF420@MOBILEE1705> Message-ID: On Mon, Aug 9, 2010 at 9:59 AM, Madovsky wrote: > its' also an excellent way for big companies to avoid > to pay employees to make all the dev work also.... > Indeed it is. Most of us in the OSS world say, "So what?" We've given our work away for "free" in return for other considerations: free advertising, free distribution (via Internet downloads), bragging rights, and growing a software-based ecosystem that allows us to tap into other revenue streams like private consulting or even writing a book. If big companies "take" our stuff and use it then they're growing the ecosystem. The choice of OSS licenses available to us gives us the necessary protection from large corporations hijacking our stuff. (This includes things like CC for documents, photos, sounds/music, etc.) A lot of OSS is so good that corporations that use it find that they need more of it. Some of them hire community members as consultants or employees. It can frequently be a win-win scenario, and even when it isn't, it is still better than the lose-lose scenario of closed/proprietary lock-down. OSS has WAY more upside than downside. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/75b64bbc/attachment.html From Jeff.Dworkin at dialogic.com Mon Aug 9 13:27:21 2010 From: Jeff.Dworkin at dialogic.com (Jeff Dworkin) Date: Mon, 9 Aug 2010 16:27:21 -0400 Subject: [Freeswitch-users] Video Support (Samuel Macedo) In-Reply-To: References: Message-ID: <0596E30D385AD34A9AA4C5452514442142F2DC43@MBX.dialogic.com> The DiaStar Server is all software, just download and install from the iso. You can do single port video with just what is on the iso. If you want to test conferencing you have to fill out the form at http://www.dialogic.com/technologies/open-source/diastar.htm?regID=38692 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Monday, August 09, 2010 4:05 PM To: freeswitch-users at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 50, Issue 45 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: GPL Wins Again (Madovsky) 2. Asterisk to Freeswitch (Michael Gende) 3. Video Support (Samuel Macedo) 4. Re: Modems (Ken Gillett) 5. Re: Custom Headers (Jesse Cloutier) 6. Re: Video Support (John Hermanski) 7. Re: Video Support (Samuel Macedo) ---------------------------------------------------------------------- Message: 1 Date: Mon, 9 Aug 2010 12:59:58 -0400 From: "Madovsky" Subject: Re: [Freeswitch-users] GPL Wins Again To: "FreeSWITCH Users Help" Message-ID: <2F9511642CCD492CA057DB89F09BF420 at MOBILEE1705> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=original its' also an excellent way for big companies to avoid to pay employees to make all the dev work also.... ----- Original Message ----- From: "M. Ranganathan" To: "FreeSWITCH Users Help" Sent: Monday, August 09, 2010 12:24 PM Subject: Re: [Freeswitch-users] GPL Wins Again > On Mon, Aug 9, 2010 at 12:08 PM, Madovsky wrote: >> I read well don't worry, >> I say that because companies start to instrumentalize open source. >> so after years some developers (for google summer source code for ex) >> realise that they spent all their lifetime for a big company for free... >> that is why I called Open Fools.... >> >> >> ----- Original Message ----- > > I speak for myself (as an individual) here : > > I disagree with your characterization. Open source is engineers > collaborating informally on a shared piece of basic infrastructure > code so as to amortize engineering cost, testing cost and beta > testing. As an open source developer, you give away something to get a > lot in return so you can build on the sum total of collaborative > effort. > > Open source is also an excellent way for students to establish a track > record. Yes it is free labor but try getting a job these days as a > fresh graduate with no experience and you will see what I am talking > about. > > Ranga > > > -- > M. Ranganathan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ------------------------------ Message: 2 Date: Mon, 9 Aug 2010 13:16:52 -0500 From: Michael Gende Subject: [Freeswitch-users] Asterisk to Freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: Content-Type: text/plain; charset="iso-8859-1" Hello, I was wondering if someone could please give me a shove in the right direction concerning some "translations". I've got some dial plan examples for a vendor's handset that are created for Asterisk. Naturally, we use FreeSwitch. Any "dial plan translation" pages to be found? I found the Rosetta Stone on the FS site. Very comprehensive, but I didn't see things for the commands in my examples (i.e, Asterisk's Pickup = FS command X) I'll figure it out but wondered if someone else had already been down this road. Regards, Mike G. P.S. Nice new FS book, by the way. Got our copy last week, very instructive and a good read to boot. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/7aca5743/attachment-0001.html ------------------------------ Message: 3 Date: Mon, 9 Aug 2010 15:47:22 -0300 From: Samuel Macedo Subject: [Freeswitch-users] Video Support To: FreeSWITCH Users Help Message-ID: Content-Type: text/plain; charset="iso-8859-1" Hi, I've already read that FS supports video, but I've never seen anyone using or talking about how to use it. Does FS support video conferences? Or only video calls? Is there any documentation about this subject? Thanks, -- Samuel Macedo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/2cf15520/attachment-0001.html ------------------------------ Message: 4 Date: Mon, 9 Aug 2010 19:49:26 +0100 From: Ken Gillett Subject: Re: [Freeswitch-users] Modems To: FreeSWITCH Users Help Message-ID: <197186CA-C15C-4F73-82A0-FB219FF9E2EA at ukgb.net> Content-Type: text/plain; charset=us-ascii Well that's the alternative, but I confess to being unclear on one aspect of such devices. In a simple scenario of several SIP devices registered to a SIP server plus a PSTN analog line, it is relatively simple to see how PSTN to VOIP bridging can be implemented with a gateway device that includes an FXS, FXO and ethernet port. An incoming call on the PSTN causes the gateway to initiate a call via the SIP server to SIP client devices registered on that ID and then putting the 2 ends of the call together so the PSTN call is 'bridged' onto VOIP and being handled by a SIP client device, i.e. a softphone or SIP telephone. But what about the reverse? My requirement is not only for the above basic ATA, but to be able to bridge VOIP to PSTN. IOW so that a SIP client can make an outgoing call onto the PSTN. Both the gateway and the SIP client are registered to a SIP server (same one in fact), how does the SIP client place a call onto the PSTN? The gateway is not a SIP server, what is the process by which a SIP client can tell such a gateway to place a call on the PSTN to the number that is somehow provided to the gateway by the SIP client? I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation. I'd be grateful for any advice about this. Thanks. On 8 Aug 2010, at 20:49, Tony Graziano wrote: > You can always use a standalone sip gateway (patton, audiocodes, etc.). > > On 8/8/10, Ken Gillett wrote: >> I asked this before, but never got any response, so hope someone can help me >> this time. >> >> Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, i.e. >> be able to accept incoming PSTN calls and also make PSTN calls? >> >> I plan on installing FreeSwitch on a Mac and it would be really handy to be >> able to use Apple's USB modem in this way as otherwise there's no way to add >> such PSTN functionality on the Mac Mini. Although my requirement is Mac >> based, I guess this question applies to any FreeSwitch installation. Ken G i l l e t t _/_/_/_/_/_/_/_/ ------------------------------ Message: 5 Date: Mon, 09 Aug 2010 09:47:37 -0400 From: Jesse Cloutier Subject: Re: [Freeswitch-users] Custom Headers To: FreeSWITCH Users Help Message-ID: <4C6006F9.5010409 at cronomagic.com> Content-Type: text/plain; charset="iso-8859-1" Perfect!, after more digging I found this line in the documentation of for mod sofia All inbound SIP calls will install any X- headers into local variables but I wasnt sure about the syntax, Thanks! On 08/09/2010 09:36 AM, Sergey Okhapkin wrote: > ${sip_h_X-MyHeader} > > On Monday 09 August 2010, Jesse Cloutier wrote: >> Hi all, >> Is there a way to read custom sip headers (eg: P-MyHeader or X-MyHeader) >> in the freeswitch dialplan? >> >> I am using freeswitch as a SBC and would like to pass certain variables >> to it for the cdr. >> >> Thanks! >> Jesse Cloutier >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/6048103c/attachment-0001.html ------------------------------ Message: 6 Date: Mon, 9 Aug 2010 15:08:02 -0400 From: John Hermanski Subject: Re: [Freeswitch-users] Video Support To: FreeSWITCH Users Help Message-ID: <9053AD1B23AABE42A65FB0B31B1E1CE7022F34B0C3 at MBX.dialogic.com> Content-Type: text/plain; charset="us-ascii" Hi Samuel, Project Diastar from Dialogic has been doing video plays, records, transcoding and conferencing with Asterisk for quite some time now. We have recently produced a "mod_woomera" to work with FS and showed it last week at ClueCon. While it's still in a somewhat green state, it is in our nightly builds and is available for testing. Please take a look at www.projectdiastar.org for more details. Thanks, John Hermanski Technical Marketing Engineer Dialogic From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samuel Macedo Sent: Monday, August 09, 2010 2:47 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Video Support Hi, I've already read that FS supports video, but I've never seen anyone using or talking about how to use it. Does FS support video conferences? Or only video calls? Is there any documentation about this subject? Thanks, -- Samuel Macedo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/fc0336c6/attachment-0001.html ------------------------------ Message: 7 Date: Mon, 9 Aug 2010 17:04:47 -0300 From: Samuel Macedo Subject: Re: [Freeswitch-users] Video Support To: FreeSWITCH Users Help Message-ID: Content-Type: text/plain; charset="windows-1252" Hi John, Can I test the "mod_woomera" without a DiaStar Server? Thanks, -- Samuel Macedo On 9 August 2010 16:08, John Hermanski wrote: > Hi Samuel, > > > > Project Diastar from Dialogic has been doing video plays, records, > transcoding and conferencing with Asterisk for quite some time now. We have > recently produced a ?mod_woomera? to work with FS and showed it last week at > ClueCon. While it?s still in a somewhat green state, it is in our nightly > builds and is available for testing. > > > > Please take a look at www.projectdiastar.org for more details. > > > > Thanks, > > > > John Hermanski > > Technical Marketing Engineer > > Dialogic > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Samuel > Macedo > *Sent:* Monday, August 09, 2010 2:47 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Video Support > > > > Hi, > > > > I've already read that FS supports video, but I've never seen anyone using > or talking about how to use it. > > Does FS support video conferences? Or only video calls? Is there any > documentation about this subject? > > > > Thanks, > > -- > > Samuel Macedo > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/213f179f/attachment.html ------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of FreeSWITCH-users Digest, Vol 50, Issue 45 ************************************************ From John.Hermanski at dialogic.com Mon Aug 9 14:45:30 2010 From: John.Hermanski at dialogic.com (John Hermanski) Date: Mon, 9 Aug 2010 17:45:30 -0400 Subject: [Freeswitch-users] Video Support Message-ID: <9053AD1B23AABE42A65FB0B31B1E1CE7022F34B18B@MBX.dialogic.com> Hi Samuel, Well, I guess you could, but as it's a Freeswitch client that is specifically designed to work with a Diastar server, it would be kind of pointless. The "server" is all software and runs on a moderately powerful PC. So, if you have a PC with a decent multicore CPU and 1 GB of memory, you can download the server software, get a trial license (http://www.dialogic.com/technologies/open-source/diastar.htm?regID=38692) and start working with mod_woomera. We distribute the software as a bootable ISO that self-installs - CentOS included - quite easily. It will, however, overwrite your disk drive. John Hermanski Technical Marketing Engineer Dialogic This e-mail is intended only for the named recipient(s) and may contain information that is privileged, confidential and/or exempt from disclosure under applicable law. No waiver of privilege, confidence or otherwise is intended by virtue of communication via the internet. Any unauthorized use, dissemination or copying is strictly prohibited. If you have received this e-mail in error, or are not named as a recipient, please immediately notify the sender and destroy all copies of this e-mail. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/a468ca15/attachment.html From brian at microcomaustralia.com.au Mon Aug 9 16:30:23 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Tue, 10 Aug 2010 09:30:23 +1000 Subject: [Freeswitch-users] freetdm dialtone In-Reply-To: References: Message-ID: On 10 August 2010 02:24, Terry Moore-Read wrote: > Wow that's fast - Can't test until tonight. ? I'll let you know. Same here. In fact my weekday time is restricted, so I may not be able to test until the weekend. Thanks! -- Brian May From msc at freeswitch.org Mon Aug 9 16:33:49 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Aug 2010 16:33:49 -0700 Subject: [Freeswitch-users] Asterisk to Freeswitch In-Reply-To: References: Message-ID: On Mon, Aug 9, 2010 at 11:16 AM, Michael Gende wrote: > Hello, > > I was wondering if someone could please give me a shove in the right > direction concerning some "translations". > > I've got some dial plan examples for a vendor's handset that are created > for Asterisk. Naturally, we use FreeSwitch. > > Any "dial plan translation" pages to be found? > > I found the Rosetta Stone on the FS site. Very comprehensive, but I didn't > see things for the commands in my examples (i.e, Asterisk's Pickup = FS > command X) > Hmm... that's a good one. Nothing comprehensive that I'm aware of. A one-to-one, where possible, sounds like a good idea. If someone could assist with collecting the list of Asterisk dialplan commands then I will organize the community effort to get it all documented. To answer your specific question, though, I believe X = "intercept" :P http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_intercept > I'll figure it out but wondered if someone else had already been down this > road. > If/when you figure anything out please keep us informed. We can expand the Rosetta Stone page. I'd like to make sure that everything you learn gets put down in the wiki... Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/c75dbcf6/attachment.html From mrene_lists at avgs.ca Mon Aug 9 16:44:42 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 9 Aug 2010 19:44:42 -0400 Subject: [Freeswitch-users] Video Support In-Reply-To: References: Message-ID: <725A1E83-CC4E-482B-B20F-BBD913381DB2@avgs.ca> You can look at mod_fsv, you can record and play videos provided you call with an IP phone. mod_conference also will automatically switch the video stream to the one of the person speaking. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-08-09, at 2:47 PM, Samuel Macedo wrote: > Hi, > > I've already read that FS supports video, but I've never seen anyone using or talking about how to use it. > Does FS support video conferences? Or only video calls? Is there any documentation about this subject? > > Thanks, > -- > Samuel Macedo > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jaybinks at gmail.com Mon Aug 9 16:55:53 2010 From: jaybinks at gmail.com (jay binks) Date: Tue, 10 Aug 2010 09:55:53 +1000 Subject: [Freeswitch-users] we are under attack In-Reply-To: References: <02416836BE07480BB29CEA6BF0606D04@dell9400> Message-ID: with your permission ill probably copy the script to the FS wiki, and attribute it with links to the other locations. ( that way we dont rely on a 3rd party site to stay up ) as for the issue with STOP... .. i removes all rules :) and if you have the default set to drop all incoming, then .. well your stuffed.. STOP should just back out the rules that the script added, not remove everything J On Tue, Aug 10, 2010 at 12:23 AM, Kristian Kielhofner wrote: > The current version of the script and a little background can be found > here: > > http://blog.krisk.org/2008/07/sip-dosddos-mitigation.html > > I have no problem linking the FS wiki to either of these but the > eTel wiki will probably get taken down eventually. > > As far as the issue with the stop command... I remember us talking > about it but I don't remember the specific problem. Care to refresh > my memory? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/a5c1b597/attachment.html From kris at kriskinc.com Mon Aug 9 17:04:51 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 9 Aug 2010 20:04:51 -0400 Subject: [Freeswitch-users] we are under attack Message-ID: <8ec19884cc9fe4010e5132cbb78bb7dd@mail.gmail.com> Oh yeah, that was it :). I leave default to accept and add my own drop everything at the end. -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces at lists.freeswitch.org < freeswitch-users-bounces at lists.freeswitch.org> *To*: FreeSWITCH Users Help *Sent*: Mon Aug 09 19:55:53 2010 *Subject*: Re: [Freeswitch-users] we are under attack with your permission ill probably copy the script to the FS wiki, and attribute it with links to the other locations. ( that way we dont rely on a 3rd party site to stay up ) as for the issue with STOP... .. i removes all rules :) and if you have the default set to drop all incoming, then .. well your stuffed.. STOP should just back out the rules that the script added, not remove everything J On Tue, Aug 10, 2010 at 12:23 AM, Kristian Kielhofner wrote: > The current version of the script and a little background can be found > here: > > http://blog.krisk.org/2008/07/sip-dosddos-mitigation.html > > I have no problem linking the FS wiki to either of these but the > eTel wiki will probably get taken down eventually. > > As far as the issue with the stop command... I remember us talking > about it but I don't remember the specific problem. Care to refresh > my memory? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/96dae49d/attachment.html From vizentini at hotmail.com Mon Aug 9 17:24:23 2010 From: vizentini at hotmail.com (Paulo Vicentini) Date: Tue, 10 Aug 2010 00:24:23 +0000 Subject: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking Message-ID: Hi, With ESL: this->execute ( 'play_and_get_digits',... $var .. $digits = $this->api_uuid_getvar ( $uuid, $var ); But the problem is that play_and_get_digits *is not blocking* so that I can't get dtmf digits I made a simple test by sleeping a few seconds after calling play_and_get_digits so that I was able to get digits...Is play_and_get_digits supposed to be a blocking blocking instruction? (FreeSWITCH Version 1.0.head (git-c3d6c64 2010-07-05 10-08-30 -0500)) Thank youPaulo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/a70df9ff/attachment.html From msc at freeswitch.org Mon Aug 9 17:39:56 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Aug 2010 17:39:56 -0700 Subject: [Freeswitch-users] Modems In-Reply-To: <197186CA-C15C-4F73-82A0-FB219FF9E2EA@ukgb.net> References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> <197186CA-C15C-4F73-82A0-FB219FF9E2EA@ukgb.net> Message-ID: Check out the Linksys SPA-3102 -MC On Mon, Aug 9, 2010 at 11:49 AM, Ken Gillett wrote: > Well that's the alternative, but I confess to being unclear on one aspect > of such devices. > > In a simple scenario of several SIP devices registered to a SIP server plus > a PSTN analog line, it is relatively simple to see how PSTN to VOIP bridging > can be implemented with a gateway device that includes an FXS, FXO and > ethernet port. An incoming call on the PSTN causes the gateway to initiate a > call via the SIP server to SIP client devices registered on that ID and then > putting the 2 ends of the call together so the PSTN call is 'bridged' onto > VOIP and being handled by a SIP client device, i.e. a softphone or SIP > telephone. > > But what about the reverse? My requirement is not only for the above basic > ATA, but to be able to bridge VOIP to PSTN. IOW so that a SIP client can > make an outgoing call onto the PSTN. Both the gateway and the SIP client are > registered to a SIP server (same one in fact), how does the SIP client place > a call onto the PSTN? The gateway is not a SIP server, what is the process > by which a SIP client can tell such a gateway to place a call on the PSTN to > the number that is somehow provided to the gateway by the SIP client? > > I am actually trying this out with a Zoom 5801 which with an FXO and FXS > port and the ability to bridge in both directions can apparently do what I > require, but I cannot get my head around what I am even trying to get it to > do. And this is before I've even thought about bringing FreeSwitch into the > equation. > > I'd be grateful for any advice about this. Thanks. > > > On 8 Aug 2010, at 20:49, Tony Graziano wrote: > > > You can always use a standalone sip gateway (patton, audiocodes, etc.). > > > > On 8/8/10, Ken Gillett wrote: > >> I asked this before, but never got any response, so hope someone can > help me > >> this time. > >> > >> Is it possible to use a (USB) modem as a 'trunk' to and from the PSTN, > i.e. > >> be able to accept incoming PSTN calls and also make PSTN calls? > >> > >> I plan on installing FreeSwitch on a Mac and it would be really handy to > be > >> able to use Apple's USB modem in this way as otherwise there's no way to > add > >> such PSTN functionality on the Mac Mini. Although my requirement is Mac > >> based, I guess this question applies to any FreeSwitch installation. > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/9c75265a/attachment-0001.html From msc at freeswitch.org Mon Aug 9 17:40:35 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Aug 2010 17:40:35 -0700 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: References: <29331510.post@talk.nabble.com> Message-ID: On Mon, Aug 9, 2010 at 12:40 AM, Steven Ayre wrote: > Just found that parameter was listed but undocumented on the Wiki, i've > fixed that. > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#apply-proxy-acl > > -Steve Gold star for you! Thanks for updating the wiki. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/125aaf8a/attachment.html From msc at freeswitch.org Mon Aug 9 17:42:42 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Aug 2010 17:42:42 -0700 Subject: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking In-Reply-To: References: Message-ID: On Mon, Aug 9, 2010 at 5:24 PM, Paulo Vicentini wrote: > Hi, With ESL: this->execute ( 'play_and_get_digits',... $var .. $digits = > $this->api_uuid_getvar ( $uuid, $var ); > Paste your entire script, or at least a simple example that demonstrates the issue so that we can try it ourselves. Personally, I've not experienced this issue so I would like to know more about the circumstances where you're seeing this. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/0b54c7a7/attachment.html From Nabble at slickdeals.endjunk.com Mon Aug 9 17:52:27 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Mon, 9 Aug 2010 17:52:27 -0700 (PDT) Subject: [Freeswitch-users] Example 15: Speaking Clock Message-ID: <1281401547317-5391252.post@n2.nabble.com> Has anyone managed to implement the Example 15: Speaking Clock as shown http://wiki.freeswitch.org/wiki/Dialplan_XML here ? If so, I am puzzled as where to put the XML file? Also, in this example, it references to http://wiki.freeswitch.org/wiki/Mod_flite mod_flite and where would I put this mod_flite file? I put these two files under conf/dialplan/public directory. Then, what number should I dial to reach this Speaking Clock? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Example-15-Speaking-Clock-tp5391252p5391252.html Sent from the freeswitch-users mailing list archive at Nabble.com. From terrymr at gmail.com Mon Aug 9 18:14:47 2010 From: terrymr at gmail.com (Terry Moore-Read) Date: Mon, 9 Aug 2010 18:14:47 -0700 Subject: [Freeswitch-users] freetdm dialtone In-Reply-To: References: Message-ID: Works for me. On Mon, Aug 9, 2010 at 4:30 PM, Brian May wrote: > On 10 August 2010 02:24, Terry Moore-Read wrote: >> Wow that's fast - Can't test until tonight. ? I'll let you know. > > Same here. > > In fact my weekday time is restricted, so I may not be able to test > until the weekend. > > Thanks! > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Aug 9 18:20:17 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Aug 2010 18:20:17 -0700 Subject: [Freeswitch-users] Example 15: Speaking Clock In-Reply-To: <1281401547317-5391252.post@n2.nabble.com> References: <1281401547317-5391252.post@n2.nabble.com> Message-ID: You can create a new file under conf/dialplan/default and call it something like "clock.xml" The dialed number in the example is 2910 but you can use any available destination number you like You need to build mod_flite: from src dir do "make mod_flite-install" HINT: I covered this nicely in chapters 2 and 5 of the new FS book. :P -MC On Mon, Aug 9, 2010 at 5:52 PM, mazilo wrote: > > Has anyone managed to implement the Example 15: Speaking Clock as shown > http://wiki.freeswitch.org/wiki/Dialplan_XML here ? If so, I am puzzled as > where to put the XML file? Also, in this example, it references to > http://wiki.freeswitch.org/wiki/Mod_flite mod_flite and where would I put > this mod_flite file? I put these two files under conf/dialplan/public > directory. Then, what number should I dial to reach this Speaking Clock? > > ----- > don't and stop are the ONLY two 4-letter words considered offensive to men, > but not when used together. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Example-15-Speaking-Clock-tp5391252p5391252.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/d4a03dad/attachment.html From msc at freeswitch.org Mon Aug 9 18:22:04 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Aug 2010 18:22:04 -0700 Subject: [Freeswitch-users] Leg_timeout question In-Reply-To: <97D5CE44FD7A4B7F9D58FC89BF18809F@greyhawk.tonecommander.com> References: <97D5CE44FD7A4B7F9D58FC89BF18809F@greyhawk.tonecommander.com> Message-ID: On Mon, Aug 9, 2010 at 8:46 AM, Robert Hadley wrote: > Try > > > Just set this *before* the bridge app otherwise it won't do a whole lot. :P -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/e3ef29a6/attachment.html From moises.silva at gmail.com Mon Aug 9 18:44:06 2010 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 9 Aug 2010 21:44:06 -0400 Subject: [Freeswitch-users] freetdm dialtone In-Reply-To: References: Message-ID: cool, do not hesitate in reporting freetdm issues here or at jira.freeswitch.org (the latter is preferred and you have better chances of having it fixed if you assign the issue to me right away). Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com On Mon, Aug 9, 2010 at 9:14 PM, Terry Moore-Read wrote: > Works for me. > > On Mon, Aug 9, 2010 at 4:30 PM, Brian May > wrote: > > On 10 August 2010 02:24, Terry Moore-Read wrote: > >> Wow that's fast - Can't test until tonight. I'll let you know. > > > > Same here. > > > > In fact my weekday time is restricted, so I may not be able to test > > until the weekend. > > > > Thanks! > > -- > > Brian May > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/c7781cd8/attachment.html From Nabble at slickdeals.endjunk.com Mon Aug 9 18:49:56 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Mon, 9 Aug 2010 18:49:56 -0700 (PDT) Subject: [Freeswitch-users] Example 15: Speaking Clock In-Reply-To: References: <1281401547317-5391252.post@n2.nabble.com> Message-ID: <1281404996939-5391373.post@n2.nabble.com> mercutioviz wrote: > > You can create a new file under conf/dialplan/default and call it > something > like "clock.xml" > The dialed number in the example is 2910 but you can use any available > destination number you like Now, I put the XML file under conf/dialplan/default directory and it is working. Thank you very much. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Example-15-Speaking-Clock-tp5391252p5391373.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lists at ione.ch Mon Aug 9 20:02:43 2010 From: lists at ione.ch (Roman_) Date: Mon, 9 Aug 2010 20:02:43 -0700 (PDT) Subject: [Freeswitch-users] Gateway registration issue, can only be solved through "killgw" Message-ID: <1281409363332-5391507.post@n2.nabble.com> Hi all, I experienced an issue with gateway registration yesterday, where the registration first failed with a "Request Timeout (408)" error and after that it kept failing with "Bad Request (400)". It kept retrying periodically with increasing delays in between, but even after almost 24 hours the registration still failed (and with delays of about 40 minutes between attempts). Only then did I notice that the gateway had failed, and restarting the gateway ("killgw" then "rescan") immediately solved the problem and the gateway registered successfully. This is part of the log: 2010-08-08 16:31:30.464055 [ERR] sofia_reg.c:1502 gateyway.com Registration Failed with status Request Timeout [408]. failure #1 2010-08-08 16:31:35.444058 [WARNING] sofia_reg.c:387 gateway.com Failed Registration, setting retry to 60 seconds. 2010-08-08 16:32:54.674056 [NOTICE] sofia_reg.c:342 Registering gateway.com 2010-08-08 16:32:54.744052 [ERR] sofia_reg.c:1502 gateway.com Registration Failed with status Bad Request [400]. failure #2 2010-08-08 16:33:04.614052 [WARNING] sofia_reg.c:387 gateway.com Failed Registration, setting retry to 90 seconds. this then continued for almost 24 hours before I noticed it, always with error code 400. Now, what worries me about this is that the gateway stopped working, but was not able to recover from the problem by itself. Only manual intervention by restarting the gateway solved the issue. Is there any way I can make sure that the gateway does not get stuck like this and can recover by itself? Does the fact that it did not recover by itself not suggest that there might be a bug in Freeswitch when registration fails, which prevents a successful registration until one manually kills and restarts the gateway? I am using the Freeswitch version from the master branch of the git repository, compiled on July 19. Any help is appreciated, thanks. Best regards, Roman -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Gateway-registration-issue-can-only-be-solved-through-killgw-tp5391507p5391507.html Sent from the freeswitch-users mailing list archive at Nabble.com. From rupa at rupa.com Mon Aug 9 20:23:18 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 9 Aug 2010 22:23:18 -0500 Subject: [Freeswitch-users] Modems In-Reply-To: <197186CA-C15C-4F73-82A0-FB219FF9E2EA@ukgb.net> References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> <197186CA-C15C-4F73-82A0-FB219FF9E2EA@ukgb.net> Message-ID: I'm pretty sure the zoom does not support sip originated calls to the FXO port. It's FXO port is strictly used as failover or selectable via dialplan when the call originates from the FXS port (eg: dial 9 first to get FXO). Try: cisco 3102 audiocodes grandstream for atas that support full FXS/FXO operation. On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett wrote: > I am actually trying this out with a Zoom 5801 which with an FXO and FXS > port and the ability to bridge in both directions can apparently do what I > require, but I cannot get my head around what I am even trying to get it to > do. And this is before I've even thought about bringing FreeSwitch into the > equation. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/453f1f96/attachment.html From mike at van.lammeren.net Mon Aug 9 23:20:48 2010 From: mike at van.lammeren.net (Mike van Lammeren) Date: Tue, 10 Aug 2010 02:20:48 -0400 Subject: [Freeswitch-users] Question about SIP security In-Reply-To: <4C54282A.90503@puzzled.xs4all.nl> References: <208669808.150818.1280577262199.JavaMail.apache@mail22.abv.bg> <4C54282A.90503@puzzled.xs4all.nl> Message-ID: Thank you, everyone, for your suggestions. Both Fail2Ban.org and IPDeny.com are good solutions. Mike van Lammeren On Sat, Jul 31, 2010 at 9:42 AM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 07/31/2010 01:54 PM, Hristo Benev wrote: > > > > I also see a lot of attempts to connect to 5060, but it looks like a > botnet since I never have more than 1 form 1 IP - Here fail2ban is helpless > since attempts per IP limit is never reached. > > Unless you set the number of failed attempts to 1. > > If it fits your application you can also use something like > http://www.ipdeny.com/ to block port 5060 for certain countries. > > Regards, > Patrick > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/c19e6a0a/attachment.html From brokendash at gmail.com Mon Aug 9 23:31:08 2010 From: brokendash at gmail.com (broken dash) Date: Tue, 10 Aug 2010 01:31:08 -0500 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: <4C602C60.4010803@infosecurity.ch> References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: I'm looking for some method to stream in google video uri's much like what one could do with the playback+shout://.. Brian On Mon, Aug 9, 2010 at 11:27 AM, Fabio Pietrosanti (naif) wrote: > Wow! > > Are there some ready-made, very simple, opensource Flash client to use > Red5 along with FS for web telephony? > > Fabio > > On 09/08/10 15.53, Richard Alam wrote: >> Full duplex >> >> Flash -> Red5 -> FS and vice-versa. >> >> Richard >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From neilp at cs.stanford.edu Tue Aug 10 00:12:16 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Tue, 10 Aug 2010 12:42:16 +0530 Subject: [Freeswitch-users] CentOS 5.x and Python 2.6: segmentation fault importing ESL Message-ID: Hi All, I am using Python 2.6 on my CentOS installation, installed in parallel to the standard Python 2.4. I've compiled FS's python ESL module using 'make pymod', but the ESL python module causes a segfault when I try to import it in Python 2.6 (imports fine in 2.4). I'm guessing this is because the ESL module was compiled for Python 2.4. How do I change the ESL config/makefiles to compile for 2.6? I'm not familiar with SWIG, is there some configuration I need to change with it to use python2.6 executable? Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/a19ad8ad/attachment.html From paul.gore.j at gmail.com Mon Aug 9 19:53:07 2010 From: paul.gore.j at gmail.com (paul gore) Date: Mon, 9 Aug 2010 22:53:07 -0400 Subject: [Freeswitch-users] Leg_timeout question In-Reply-To: <97D5CE44FD7A4B7F9D58FC89BF18809F@greyhawk.tonecommander.com> References: <97D5CE44FD7A4B7F9D58FC89BF18809F@greyhawk.tonecommander.com> Message-ID: Thank you, that worked! But I have another problem now. If I don't want to forward to voice mail at the end, but just hangup, it does not work - call repeats 3 times and no hangup gets send to the call originator, even with "hangup" action at the end. Like this: Is this fixable? On Mon, Aug 9, 2010 at 11:46 AM, Robert Hadley wrote: > Try > > > > ------------------------------ > > *From:* paul gore [mailto:paul.gore.j at gmail.com] > *Sent:* Sunday, August 08, 2010 7:39 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Leg_timeout question > > > > Hi there, > I am trying to accomplish a pretty simple thing in FS dialplan - ring one > or multiple destinations with individual timeout per destination, if nobody > answers - forward to a voicemail. > Here is dial plan I use: > > continue="false"> expression="^(\d{7,20})$"/> > expression="^(call_route_16099860919)$"> > > > > > > > data="[leg_timeout=20]sofia/internal/1002%${domain_name},[leg_timeout=25]sofia/internal/1003%${domain_name}"/> > data="voicemail_greeting_number=1"/> data="default $${domain} 1002"/> > > What happens is incoming call from my provider rings ext 1002, but instead > of going to the VM after 20 sec and no answer FS sends 602 (alloted_timeout) > and the ring repeats again and so 3 times, after which call dies. But never > goes to voice mail. > What I am doing wrong? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/ee0d795d/attachment-0001.html From prayersts at gmail.com Mon Aug 9 19:59:28 2010 From: prayersts at gmail.com (Tae-Sung Shin) Date: Mon, 9 Aug 2010 22:59:28 -0400 Subject: [Freeswitch-users] Calling internal extensions to/from outside is acting strange In-Reply-To: <012301cb37bd$7f399d60$7dacd820$@com> References: <012301cb37bd$7f399d60$7dacd820$@com> Message-ID: <01bc01cb3838$0d471f40$27d55dc0$@com> I found the resolution. According to FS log, problem was FS getting "Transcoding_necessary" error and hangup whenever outbound call was answered. My solution was simply to upgrade FS. The problematic version was prebuild Windows version 1.0.4 but getting and compiling FS 1.0.6 source was worth the effort. Everything is working fine now. Hope this helps anybody having a similar problem. Tae-Sung Shin From: Tae-Sung Shin [mailto:prayersts at gmail.com] Sent: Monday, August 09, 2010 8:22 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Calling internal extensions to/from outside is acting strange Hello Guys First of all, I am a new user of Freeswitch. I spent a couple of days on this issue and am desperate for some help. Briefly, my problems are 1. Calling out from SPA2102 (ext 1002) or a softphone (phonerlite) (ext 1003) via gateway voip is disconnected as soon as it got answered after ringing in the other side. 2. Calling from outside is disconnected 30 seconds after it got answered. I don't have this problem with calls between the internal extensions. Without extensions (direct communication between gateway and SPA2102), I verified SPA2102 is working fine My environment: --- --- As Freeswitch wiki, suggested, I have following xml contents . Sip_profiles/external/voipms.xml . Dialplan/public/voipms.xml I think I went through all wiki and other articles on the web but could find a resolution for this issue. I would appreciate if you give me any hint. Thanks Tae-Sung Shin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100809/3edda4d1/attachment-0001.html From bernhard.suttner at winet.ch Mon Aug 9 23:45:24 2010 From: bernhard.suttner at winet.ch (Bernhard Suttner) Date: Tue, 10 Aug 2010 08:45:24 +0200 Subject: [Freeswitch-users] we are under attack Message-ID: <767f276a-7240-4a9e-a8f5-798233e5a83f@winet.ch> Hi, wouldn?t it be a good task for freeswitch, to detect registerspoofing and call a ?hook? (maybe a script) which than could close the firewall?Or that freeswitch will just ignore any request from the given IP for a acertain time. I tried to find a good application/script to prevent registerspoofing but the only thing was the fail2ban script. Couldn?t freeswitch justspeak directly with the firewall and drop these requests J ? Best regards, Bernhard Von:freeswitch-users-bounces at lists.freeswitch.org[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von KristianKielhofner Gesendet: Dienstag, 10. August 2010 02:05 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] we are under attack Oh yeah, that was it :). I leave default to accept and add my owndrop everything at the end. -- Kristian Kielhofner http://blog.krisk.org From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Mon Aug 09 19:55:53 2010 Subject: Re: [Freeswitch-users] we are under attack with your permission ill probably copy the script to the FSwiki, and attribute it with links to the other locations. ( that way we dont rely on a 3rd party site to stay up ) as for the issue with STOP... .. i removes all rules :) and if you have the default set to drop all incoming, then.. well your stuffed.. STOP should just back out the rules that the script added,not remove everything J On Tue, Aug 10, 2010 at 12:23 AM, Kristian Kielhofner wrote: The current version of the script and a littlebackground can be found here: http://blog.krisk.org/2008/07/sip-dosddos-mitigation.html I have no problem linking the FS wiki to either of these but the eTel wiki will probably get taken down eventually. As far as the issue with the stop command... I remember us talking about it but I don't remember the specific problem. Care to refresh my memory? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/5dff23b7/attachment.html From ken at ukgb.net Tue Aug 10 00:17:30 2010 From: ken at ukgb.net (Ken Gillett) Date: Tue, 10 Aug 2010 08:17:30 +0100 Subject: [Freeswitch-users] Modems In-Reply-To: References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> <197186CA-C15C-4F73-82A0-FB219FF9E2EA@ukgb.net> Message-ID: Well I did try a Linksys/Cisco/Sipura 3102 but it's a configuration nightmare. The problem mainly is as I said, I cannot figure how VOIP -> PSTN bridging can be achieved whatever the gateway device. A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the call. From that piece of data it gets the SIP domain and the ID of the user. So how is the gateway able to be inserted in this process. The SIP client has no knowledge of this device and no way to include that third piece of data into its process. Obviously such gateways do exist, but I do not yet understand how the process works. Some advice on the basic process flow would therefore assist me to set up what I need irrespective of what devices I am using. Sorry to be so ignorant about this, but anyone able and willing to help with an explanation? On 10 Aug 2010, at 04:23, Rupa Schomaker wrote: > I'm pretty sure the zoom does not support sip originated calls to the FXO port. It's FXO port is strictly used as failover or selectable via dialplan when the call originates from the FXS port (eg: dial 9 first to get FXO). > > Try: > > cisco 3102 > audiocodes > grandstream > > for atas that support full FXS/FXO operation. > > On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett wrote: > I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation. Ken G i l l e t t _/_/_/_/_/_/_/_/ From steveayre at gmail.com Tue Aug 10 00:37:38 2010 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 10 Aug 2010 08:37:38 +0100 Subject: [Freeswitch-users] Leg_timeout question In-Reply-To: References: <97D5CE44FD7A4B7F9D58FC89BF18809F@greyhawk.tonecommander.com> Message-ID: Setting hangup_after_bridge=true should do that... which you already have.. Can you post your log (debug level)? -Steve On 10 August 2010 03:53, paul gore wrote: > Thank you, that worked! > > But I have another problem now. If I don't want to forward to voice mail at > the end, but just hangup, it does not work - call repeats 3 times and no > hangup gets send to the call originator, even with "hangup" action at the > end. > > Like this: > > continue="false"> expression="^(\d{7,20})$"/> > expression="^(call_route_16099860919)$"> > > > > > > > > data="[leg_timeout=20]sofia/internal/1002%${domain_name},[leg_timeout=25]sofia/internal/1003%${domain_name}"/> > > > > Is this fixable? > > > On Mon, Aug 9, 2010 at 11:46 AM, Robert Hadley wrote: > >> Try >> >> >> >> ------------------------------ >> >> *From:* paul gore [mailto:paul.gore.j at gmail.com] >> *Sent:* Sunday, August 08, 2010 7:39 PM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Leg_timeout question >> >> >> >> Hi there, >> I am trying to accomplish a pretty simple thing in FS dialplan - ring one >> or multiple destinations with individual timeout per destination, if nobody >> answers - forward to a voicemail. >> Here is dial plan I use: >> >> > continue="false">> expression="^(\d{7,20})$"/> >> > expression="^(call_route_16099860919)$"> >> >> >> >> >> >> >> > data="[leg_timeout=20]sofia/internal/1002%${domain_name},[leg_timeout=25]sofia/internal/1003%${domain_name}"/> >> > data="voicemail_greeting_number=1"/>> data="default $${domain} 1002"/> >> >> What happens is incoming call from my provider rings ext 1002, but instead >> of going to the VM after 20 sec and no answer FS sends 602 (alloted_timeout) >> and the ring repeats again and so 3 times, after which call dies. But never >> goes to voice mail. >> What I am doing wrong? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/be16829f/attachment.html From steveayre at gmail.com Tue Aug 10 00:41:19 2010 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 10 Aug 2010 08:41:19 +0100 Subject: [Freeswitch-users] Gateway registration issue, can only be solved through "killgw" In-Reply-To: <1281409363332-5391507.post@n2.nabble.com> References: <1281409363332-5391507.post@n2.nabble.com> Message-ID: Do you have any packet traces to the gateway during that time? If not, are you able to reproduce the problem? It would be useful to see the SIP traces to see what FS was sending to check whether it was valid, and if not to see what was wrong. You can do that by running 'sofia profile profilename siptrace on' from the CLI, the SIP packets will then show in your logfile. Same command with off instead of on will turn it back off once you're done. -Steve On 10 August 2010 04:02, Roman_ wrote: > > Hi all, > > I experienced an issue with gateway registration yesterday, where the > registration first failed with a "Request Timeout (408)" error and after > that it kept failing with "Bad Request (400)". It kept retrying > periodically > with increasing delays in between, but even after almost 24 hours the > registration still failed (and with delays of about 40 minutes between > attempts). > Only then did I notice that the gateway had failed, and restarting the > gateway ("killgw" then "rescan") immediately solved the problem and the > gateway registered successfully. > > This is part of the log: > > 2010-08-08 16:31:30.464055 [ERR] sofia_reg.c:1502 gateyway.comRegistration > Failed with status Request Timeout [408]. failure #1 > 2010-08-08 16:31:35.444058 [WARNING] sofia_reg.c:387 gateway.com Failed > Registration, setting retry to 60 seconds. > 2010-08-08 16:32:54.674056 [NOTICE] sofia_reg.c:342 Registering > gateway.com > 2010-08-08 16:32:54.744052 [ERR] sofia_reg.c:1502 gateway.com Registration > Failed with status Bad Request [400]. failure #2 > 2010-08-08 16:33:04.614052 [WARNING] sofia_reg.c:387 gateway.com Failed > Registration, setting retry to 90 seconds. > > this then continued for almost 24 hours before I noticed it, always with > error code 400. > > Now, what worries me about this is that the gateway stopped working, but > was > not able to recover from the problem by itself. Only manual intervention by > restarting the gateway solved the issue. > > Is there any way I can make sure that the gateway does not get stuck like > this and can recover by itself? Does the fact that it did not recover by > itself not suggest that there might be a bug in Freeswitch when > registration > fails, which prevents a successful registration until one manually kills > and > restarts the gateway? > > I am using the Freeswitch version from the master branch of the git > repository, compiled on July 19. > > Any help is appreciated, thanks. > > Best regards, > > Roman > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Gateway-registration-issue-can-only-be-solved-through-killgw-tp5391507p5391507.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/5f2dc743/attachment-0001.html From lists at ione.ch Tue Aug 10 01:14:37 2010 From: lists at ione.ch (Roman_) Date: Tue, 10 Aug 2010 01:14:37 -0700 (PDT) Subject: [Freeswitch-users] Gateway registration issue, can only be solved through "killgw" In-Reply-To: References: <1281409363332-5391507.post@n2.nabble.com> Message-ID: <1281428077308-5392119.post@n2.nabble.com> I do not have any traces from that time. Just now I briefly tried to reproduce the problem by blocking replies from the gateway using iptables, and I did get the "Request Timeout (408)" when FS tried to re-new the registration. I then opened the firewall again, but FS immediately succeeded in registering again after the 60 second delay had passed. So it seems that the problem is not easily reproducible or maybe some other factors were also involved. Thanks for the help. Roman Steven Ayre wrote: > > Do you have any packet traces to the gateway during that time? If not, are > you able to reproduce the problem? > > It would be useful to see the SIP traces to see what FS was sending to > check > whether it was valid, and if not to see what was wrong. > > You can do that by running 'sofia profile profilename siptrace on' from > the > CLI, the SIP packets will then show in your logfile. Same command with off > instead of on will turn it back off once you're done. > > -Steve > > > On 10 August 2010 04:02, Roman_ wrote: > >> >> Hi all, >> >> I experienced an issue with gateway registration yesterday, where the >> registration first failed with a "Request Timeout (408)" error and after >> that it kept failing with "Bad Request (400)". It kept retrying >> periodically >> with increasing delays in between, but even after almost 24 hours the >> registration still failed (and with delays of about 40 minutes between >> attempts). >> Only then did I notice that the gateway had failed, and restarting the >> gateway ("killgw" then "rescan") immediately solved the problem and the >> gateway registered successfully. >> >> This is part of the log: >> >> 2010-08-08 16:31:30.464055 [ERR] sofia_reg.c:1502 >> gateyway.comRegistration >> Failed with status Request Timeout [408]. failure #1 >> 2010-08-08 16:31:35.444058 [WARNING] sofia_reg.c:387 gateway.com Failed >> Registration, setting retry to 60 seconds. >> 2010-08-08 16:32:54.674056 [NOTICE] sofia_reg.c:342 Registering >> gateway.com >> 2010-08-08 16:32:54.744052 [ERR] sofia_reg.c:1502 gateway.com >> Registration >> Failed with status Bad Request [400]. failure #2 >> 2010-08-08 16:33:04.614052 [WARNING] sofia_reg.c:387 gateway.com Failed >> Registration, setting retry to 90 seconds. >> >> this then continued for almost 24 hours before I noticed it, always with >> error code 400. >> >> Now, what worries me about this is that the gateway stopped working, but >> was >> not able to recover from the problem by itself. Only manual intervention >> by >> restarting the gateway solved the issue. >> >> Is there any way I can make sure that the gateway does not get stuck like >> this and can recover by itself? Does the fact that it did not recover by >> itself not suggest that there might be a bug in Freeswitch when >> registration >> fails, which prevents a successful registration until one manually kills >> and >> restarts the gateway? >> >> I am using the Freeswitch version from the master branch of the git >> repository, compiled on July 19. >> >> Any help is appreciated, thanks. >> >> Best regards, >> >> Roman >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Gateway-registration-issue-can-only-be-solved-through-killgw-tp5391507p5391507.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Gateway-registration-issue-can-only-be-solved-through-killgw-tp5391507p5392119.html Sent from the freeswitch-users mailing list archive at Nabble.com. From motosota at gmail.com Tue Aug 10 01:22:11 2010 From: motosota at gmail.com (Mike) Date: Tue, 10 Aug 2010 09:22:11 +0100 Subject: [Freeswitch-users] Conference Conrol GUI Message-ID: Hi, I'm looking to evaluate gui's for FreeSWITCH real-time conference contol (muting of participants, locking the conference etc.) I'm currently looking at FusionPBX - can anyone recommend any other packages worth looking at that support this functionality? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/141ed6ca/attachment.html From erkan at speedingtrade.com Tue Aug 10 02:08:51 2010 From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=) Date: Tue, 10 Aug 2010 12:08:51 +0300 Subject: [Freeswitch-users] SPA8000 Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C559E@server1.st.local> Hi all, I have a Linksys SPA8000 but I can't register a line. With softphones the user is work. But with SPA8000 don't register. The register to other provider is worked. So I can configure the SPA8000 correctly. The sample user can be registered with softphone and some hardphones but with SPA8000 can't register. Maybe I must change the XML in the directory folder. Everybody have idea? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/ed328e80/attachment.html From erkan at speedingtrade.com Tue Aug 10 04:00:49 2010 From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=) Date: Tue, 10 Aug 2010 14:00:49 +0300 Subject: [Freeswitch-users] SPA8000 References: <81C2CEF80046FB4F863A60D4347DD33A0C559E@server1.st.local> Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C55A1@server1.st.local> I think the problem is in FreeSwitch. If I use other voip server that works. For example VoipSwitch The SPA8000 is registered. But also by VoipSwitch as normal client don't work. You must select Common Client What is common client??? What kind of type is that? Is there exist also in FreeSwitch? Can I create a user in different type or forms. Look also at "Sip Messages Recv" and "Sip Bytes Recv:" If I used the freeswitch account they stay everytime on zero (0). So I think the freeswitch can not handle it to send the messages to the SPA8000. So the SPA8000 get nothing from FreeSwitch. Maybe a NAT setting in my FreeSwitch is not correct? But Softphones are worked. Hope the information's give you more ideas. Thank you for the help. Kind regards Erkan From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erkan ?nl? Sent: Tuesday, August 10, 2010 12:09 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] SPA8000 Hi all, I have a Linksys SPA8000 but I can't register a line. With softphones the user is work. But with SPA8000 don't register. The register to other provider is worked. So I can configure the SPA8000 correctly. The sample user can be registered with softphone and some hardphones but with SPA8000 can't register. Maybe I must change the XML in the directory folder. Everybody have idea? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/e1aea373/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 13410 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/e1aea373/attachment-0001.png From rupa at rupa.com Tue Aug 10 05:09:05 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 10 Aug 2010 07:09:05 -0500 Subject: [Freeswitch-users] Modems In-Reply-To: References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> <197186CA-C15C-4F73-82A0-FB219FF9E2EA@ukgb.net> Message-ID: http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo On Tue, Aug 10, 2010 at 2:17 AM, Ken Gillett wrote: > Well I did try a Linksys/Cisco/Sipura 3102 but it's a configuration > nightmare. The problem mainly is as I said, I cannot figure how VOIP -> PSTN > bridging can be achieved whatever the gateway device. > > A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the > call. From that piece of data it gets the SIP domain and the ID of the user. > So how is the gateway able to be inserted in this process. The SIP client > has no knowledge of this device and no way to include that third piece of > data into its process. > > Obviously such gateways do exist, but I do not yet understand how the > process works. Some advice on the basic process flow would therefore assist > me to set up what I need irrespective of what devices I am using. Sorry to > be so ignorant about this, but anyone able and willing to help with an > explanation? > > > On 10 Aug 2010, at 04:23, Rupa Schomaker wrote: > > > I'm pretty sure the zoom does not support sip originated calls to the FXO > port. It's FXO port is strictly used as failover or selectable via dialplan > when the call originates from the FXS port (eg: dial 9 first to get FXO). > > > > Try: > > > > cisco 3102 > > audiocodes > > grandstream > > > > for atas that support full FXS/FXO operation. > > > > On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett wrote: > > I am actually trying this out with a Zoom 5801 which with an FXO and FXS > port and the ability to bridge in both directions can apparently do what I > require, but I cannot get my head around what I am even trying to get it to > do. And this is before I've even thought about bringing FreeSwitch into the > equation. > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/b1c33b5f/attachment.html From tculjaga at gmail.com Tue Aug 10 05:34:11 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 10 Aug 2010 14:34:11 +0200 Subject: [Freeswitch-users] read function returns corrupted values Message-ID: hello guys, i got a problem with read function ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read) in dialplan. when this part of dialplan executes and i enter the DTMF digit immediately on 1st letter, the collected DTMF digits are not consistent... as there is a memory overwrite somewhere .... 2010-08-10 15:27:22.056491 [INFO] mod_dptools.c:946 ################# LangSel ################\n 2010-08-10 15:27:22.056491 [NOTICE] switch_core_session.c:1949 Execute set(WELCOME_PR=${DEFAULT_LANG_PATH}ivr/welcome.wav) EXECUTE sofia/external/38516659280 at 195.88.212.41set(WELCOME_PR=/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav) 2010-08-10 15:27:22.056491 [DEBUG] mod_dptools.c:816 sofia/external/ 38516659280 at 195.88.212.41 SET [WELCOME_PR]=[/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav] 2010-08-10 15:27:22.056491 [NOTICE] switch_core_session.c:1949 Execute execute_extension(LangSelInput XML NXIVR) EXECUTE sofia/external/38516659280 at 195.88.212.41execute_extension(LangSelInput XML NXIVR) 2010-08-10 15:27:22.056491 [INFO] mod_dialplan_xml.c:418 Processing 38516659280->LangSelInput in context NXIVR Dialplan: sofia/external/38516659280 at 195.88.212.41 parsing [NXIVR->LangSelInput] continue=false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSelInput] destination_number(LangSelInput) =~ /^LangSelInput$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(playback_delimiter=!) Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(playback_terminators=#*0123456789) Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(myLANG=) INLINE EXECUTE sofia/external/38516659280 at 195.88.212.41 set(myLANG=) 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ 38516659280 at 195.88.212.41 SET [myLANG]=[UNDEF] Dialplan: sofia/external/38516659280 at 195.88.212.41 Action read(0 1 ${WELCOME_PR}!${ERR_PR}!${LANG_PROMPT_STRING} myLANG ${LANG_TIMEOUT} ) Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(LANG_RETRIES=${expr(${LANG_RETRIES}+1)}) INLINE EXECUTE sofia/external/38516659280 at 195.88.212.41 set(LANG_RETRIES=1) 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ 38516659280 at 195.88.212.41 SET [LANG_RETRIES]=[1] Dialplan: sofia/external/38516659280 at 195.88.212.41 Action execute_extension(LangSel XML NXIVR) 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute set(playback_delimiter=!) EXECUTE sofia/external/38516659280 at 195.88.212.41 set(playback_delimiter=!) 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ 38516659280 at 195.88.212.41 SET [playback_delimiter]=[!] 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute set(playback_terminators=#*0123456789) EXECUTE sofia/external/38516659280 at 195.88.212.41set(playback_terminators=#*0123456789) 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ 38516659280 at 195.88.212.41 SET [playback_terminators]=[#*0123456789] 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute read(0 1 ${WELCOME_PR}!${ERR_PR}!${LANG_PROMPT_STRING} myLANG ${LANG_TIMEOUT} ) EXECUTE sofia/external/38516659280 at 195.88.212.41 read(0 1 /usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav!!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/for_croatian.wav!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/press.wav!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/one.wav!/usr/local/freeswitch/sounds/en/us/callie/ivr/for_en_press2.wav!/usr/local/freeswitch/sounds/de/de/helge/ivr/for_german_press3.wav!/usr/local/freeswitch/sounds/it/it/ambra/ivr/for_italian_press4.wav!/usr/local/freeswitch/sounds/fr/fr/celine/ivr/for_french_press5.wav myLANG 10000 ) 2010-08-10 15:27:22.059474 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms 2010-08-10 15:27:22.872525 [DEBUG] switch_ivr_play_say.c:1444 done playing file 2010-08-10 15:27:22.934526 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 1:752 2010-08-10 15:27:22.934526 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/.PCMA 2010-08-10 15:27:23.892521 [DEBUG] switch_ivr_play_say.c:1444 done playing file 2010-08-10 15:27:25.012528 [DEBUG] switch_ivr_play_say.c:1444 done playing file 2010-08-10 15:27:25.832521 [DEBUG] switch_ivr_play_say.c:1444 done playing file 2010-08-10 15:27:29.792528 [DEBUG] switch_ivr_play_say.c:1444 done playing file 2010-08-10 15:27:31.872524 [DEBUG] switch_ivr_play_say.c:1444 done playing file 2010-08-10 15:27:34.492524 [DEBUG] switch_ivr_play_say.c:1444 done playing file 2010-08-10 15:27:37.152526 [DEBUG] switch_ivr_play_say.c:1444 done playing file 2010-08-10 15:27:37.152526 [NOTICE] switch_core_session.c:1949 Execute execute_extension(LangSel XML NXIVR) EXECUTE sofia/external/38516659280 at 195.88.212.41 execute_extension(LangSel XML NXIVR) 2010-08-10 15:27:37.152526 [INFO] mod_dialplan_xml.c:418 Processing 38516659280->LangSel in context NXIVR Dialplan: sofia/external/38516659280 at 195.88.212.41 parsing [NXIVR->LangSel] continue=false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] destination_number(LangSel) =~ /^LangSel$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] ${myLANG}(1^?@?) =~ /^1$/ break=on-true Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] destination_number(LangSel) =~ /^LangSel$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] ${myLANG}(1^?@?) =~ /^2$/ break=on-true Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] destination_number(LangSel) =~ /^LangSel$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] ${myLANG}(1^?@?) =~ /^3$/ break=on-true Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] destination_number(LangSel) =~ /^LangSel$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] ${myLANG}(1^?@?) =~ /^4$/ break=on-true Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] destination_number(LangSel) =~ /^LangSel$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] ${myLANG}(1^?@?) =~ /^5$/ break=on-true Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] destination_number(LangSel) =~ /^LangSel$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] ${myLANG}(1^?@?) =~ /^[06789]$/ break=on-true Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] destination_number(LangSel) =~ /^LangSel$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] ${myLANG}(1^?@?) =~ /^$/ break=on-true Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] destination_number(LangSel) =~ /^LangSel$/ break=on-false Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] ${myLANG}(1^?@?) =~ /^.*$/ break=on-true Dialplan: sofia/external/38516659280 at 195.88.212.41 Action log(INFO ################# WRONG LANG SEL ################\n) Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(language=hr) Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(ERR_PR=${WRONG_LANG_PROMPT}) Dialplan: sofia/external/38516659280 at 195.88.212.41 Action execute_extension(LangRetries XML NXIVR) 2010-08-10 15:27:37.154487 [NOTICE] switch_core_session.c:1949 Execute log(INFO ################# WRONG LANG SEL ################\n) EXECUTE sofia/external/38516659280 at 195.88.212.41 log(INFO ################# WRONG LANG SEL ################\n) 2010-08-10 15:27:37.154487 [INFO] mod_dptools.c:946 ################# WRONG LANG SEL ################\n 2010-08-10 15:27:37.154487 [NOTICE] switch_core_session.c:1949 Execute set(language=hr) EXECUTE sofia/external/38516659280 at 195.88.212.41 set(language=hr) you can see myLANG variable holds "1^?@?" ... of course i dialed 1 as seen in the log (switch_rtp.c:2428 RTP RECV DTMF 1:752). now, i did some extra testing and found that it happens only when i use a playback_separator to append multiple files to be played.... i checked switch_ivr_read and switch_ivr_collect_digits_count functions but everything seems to be ok there... can anyone help locating the problem ? Ps: im running freeswitch at cxss01> version FreeSWITCH Version 1.0.6 (svn-exported) tried with Revision: 17032 before this one but seems to be the same.... [tculjaga at cxss01 src]$ cat /etc/issue CentOS release 5.4 (Final) Kernel \r on an \m [tculjaga at cxss01 src]$ [tculjaga at cxss01 src]$ uname -a Linux cxss01 2.6.18-164.el5 #1 SMP Thu Sep 3 03:28:30 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux [tculjaga at cxss01 src]$ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/764c0f92/attachment-0001.html From grsingh750 at gmail.com Tue Aug 10 05:53:56 2010 From: grsingh750 at gmail.com (guru singh) Date: Tue, 10 Aug 2010 18:23:56 +0530 Subject: [Freeswitch-users] Gateway registration issue, can only be solved through "killgw" In-Reply-To: <1281428077308-5392119.post@n2.nabble.com> References: <1281409363332-5391507.post@n2.nabble.com> <1281428077308-5392119.post@n2.nabble.com> Message-ID: Hi, I had similar problems a while back. The gateway would register intermittently. If you're FS server is behind NAT then it may be the case that your router is the culprit here. In my case the closest I came to reproducing the issue was that after some 408 responses I just restarted my router while FS was running and then it always registered. What router are you using? gsin From lists at ione.ch Tue Aug 10 06:14:39 2010 From: lists at ione.ch (Roman_) Date: Tue, 10 Aug 2010 06:14:39 -0700 (PDT) Subject: [Freeswitch-users] Gateway registration issue, can only be solved through "killgw" In-Reply-To: References: <1281409363332-5391507.post@n2.nabble.com> <1281428077308-5392119.post@n2.nabble.com> Message-ID: <1281446079091-5392975.post@n2.nabble.com> Hi, The server I am using is directly attached to the Internet with a public IP, so there is no NAT going on. And I would be surprised if the firewall on the server is the problem. Thanks, Roman guru singh wrote: > > Hi, > > I had similar problems a while back. The gateway would register > intermittently. If you're FS server is behind NAT then it may be the > case that your router is the culprit here. In my case the closest I > came to reproducing the issue was that after some 408 responses I just > restarted my router while FS was running and then it always > registered. What router are you using? > > gsin > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Gateway-registration-issue-can-only-be-solved-through-killgw-tp5391507p5392975.html Sent from the freeswitch-users mailing list archive at Nabble.com. From juanbackson at gmail.com Tue Aug 10 06:17:19 2010 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 10 Aug 2010 21:17:19 +0800 Subject: [Freeswitch-users] handling of re-invite on b-leg Message-ID: Hi, When freeswitch receives re-invite from the b-leg, is there any configuration in freeswitch that can tell freeswitch to simply proxy the re-invite back to the a-leg and not to enter the dialplan? Any help will be greatly appreciated. jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/dfe402ff/attachment.html From helmut.kuper at ewetel.de Tue Aug 10 06:21:06 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 10 Aug 2010 15:21:06 +0200 Subject: [Freeswitch-users] Question about srtp secured B-Leg Message-ID: <4C615242.80903@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have late codec negotiation enabled and using Snom devices for encrypted calls (inernal only). When sdp of A-leg (Snom) has a=crypto I export sip_secure_media=true in my dialplan, so that B-leg (Snom) is also encrypted via srtp. That works nicely. When the B-Leg doesn't support srtp I got an "incompatible_device" error. This is also ok so far. Unfortunately this way you can't use Snom's optional srtp mechanism which consists simply of two media profiles in A-Leg's SDP (first RTP/SAVP then RTP/AVP) so the target can choose to encrypt or not. I don't know whether this is RFC conform or not nor if this makes really sense. I would like to be able to call a non encrypting ATA for checking for eaxample fax devices by simply calling them. My question is now, is there a way to tell FS sending outbound calls also with a non crypto offer within same sdp to B-Leg, so that it can choose to encrypt or not? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMYVJB4tZeNddg3dwRApBGAJ9d4keb5RQqMlS5sJ6jfN3VLAQkhwCeO3/x QlwcZMCW2ipsAIjV7cet2aU= =qzpR -----END PGP SIGNATURE----- From 12ukwn at gmail.com Tue Aug 10 07:14:07 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Tue, 10 Aug 2010 16:14:07 +0200 Subject: [Freeswitch-users] collision risk with Debian iceweasel & icedove Message-ID: <20100810161407.225a43fb@anubis.defcon1> Hi list, FYI, if you use a Debian sid, there's a collision PB that prevents iceweasel and icedove to run on the same machine: > 13783 0.000076 open("/usr/local/freeswitch/lib/libnspr4.so", > O_RDONLY) = 32 > You have a libnspr4.so file getting in the way in your LD_LIBRARY_PATH. JY -- Very few profundities can be expressed in less than 80 characters. From 12ukwn at gmail.com Tue Aug 10 07:27:29 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Tue, 10 Aug 2010 16:27:29 +0200 Subject: [Freeswitch-users] Pb cleaning tree Message-ID: <20100810162729.7bdbb481@anubis.defcon1> Hi list, I can't rebuild after a 'git pull' because of cleaning failure: make clean Making clean in src /bin/bash: line 26: /usr/bin/make: Liste d'arguments trop longue (means: arguments list too long) make: *** [clean-recursive] Erreur 1 How can I avoid that? JY -- I disagree with unanimity. From macedoslm at gmail.com Tue Aug 10 06:00:33 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Tue, 10 Aug 2010 10:00:33 -0300 Subject: [Freeswitch-users] Video Support In-Reply-To: <725A1E83-CC4E-482B-B20F-BBD913381DB2@avgs.ca> References: <725A1E83-CC4E-482B-B20F-BBD913381DB2@avgs.ca> Message-ID: I have an IP phone and support for H.263 codec. First of all I want to make video calls using this IP phones, than I will try the video conferences. I had succeeded doing video calls with bypass_media flag enabled. I couldn't do the same with bypass_media disabled, my FS can't negotiate the video codecs. I checked the "mod_h26x" and "mod_fsv" and they are enabled. I am also sending the network trace of my tests. Is there any config in FS to make the video calls? Thanks, -- Samuel Macedo On 9 August 2010 20:44, Mathieu Rene wrote: > You can look at mod_fsv, you can record and play videos provided you call > with an IP phone. mod_conference also will automatically switch the video > stream to the one of the person speaking. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 2010-08-09, at 2:47 PM, Samuel Macedo wrote: > > > Hi, > > > > I've already read that FS supports video, but I've never seen anyone > using or talking about how to use it. > > Does FS support video conferences? Or only video calls? Is there any > documentation about this subject? > > > > Thanks, > > -- > > Samuel Macedo > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/e09de4bf/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: P2P_bypass_false.pcap Type: application/octet-stream Size: 551416 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/e09de4bf/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: P2P_bypass_true.pcap Type: application/octet-stream Size: 617913 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/e09de4bf/attachment-0003.obj From paul.gore.j at gmail.com Tue Aug 10 04:53:26 2010 From: paul.gore.j at gmail.com (paul gore) Date: Tue, 10 Aug 2010 07:53:26 -0400 Subject: [Freeswitch-users] Leg_timeout question In-Reply-To: References: <97D5CE44FD7A4B7F9D58FC89BF18809F@greyhawk.tonecommander.com> Message-ID: I should clarify myself, I talk about "no answer" scenario here, so wouldn't have any effect, correct? So in this case I assume should properly hangup the call, but it does not. On Mon, Aug 9, 2010 at 10:53 PM, paul gore wrote: > Thank you, that worked! > > But I have another problem now. If I don't want to forward to voice mail at > the end, but just hangup, it does not work - call repeats 3 times and no > hangup gets send to the call originator, even with "hangup" action at the > end. > > Like this: > > continue="false"> expression="^(\d{7,20})$"/> > expression="^(call_route_16099860919)$"> > > > > > > > > data="[leg_timeout=20]sofia/internal/1002%${domain_name},[leg_timeout=25]sofia/internal/1003%${domain_name}"/> > > > > Is this fixable? > > > On Mon, Aug 9, 2010 at 11:46 AM, Robert Hadley wrote: > >> Try >> >> >> >> ------------------------------ >> >> *From:* paul gore [mailto:paul.gore.j at gmail.com] >> *Sent:* Sunday, August 08, 2010 7:39 PM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Leg_timeout question >> >> >> >> Hi there, >> I am trying to accomplish a pretty simple thing in FS dialplan - ring one >> or multiple destinations with individual timeout per destination, if nobody >> answers - forward to a voicemail. >> Here is dial plan I use: >> >> > continue="false">> expression="^(\d{7,20})$"/> >> > expression="^(call_route_16099860919)$"> >> >> >> >> >> >> >> > data="[leg_timeout=20]sofia/internal/1002%${domain_name},[leg_timeout=25]sofia/internal/1003%${domain_name}"/> >> > data="voicemail_greeting_number=1"/>> data="default $${domain} 1002"/> >> >> What happens is incoming call from my provider rings ext 1002, but instead >> of going to the VM after 20 sec and no answer FS sends 602 (alloted_timeout) >> and the ring repeats again and so 3 times, after which call dies. But never >> goes to voice mail. >> What I am doing wrong? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/e888bef9/attachment.html From vmknott at gmail.com Tue Aug 10 08:02:51 2010 From: vmknott at gmail.com (VM Knott) Date: Tue, 10 Aug 2010 11:02:51 -0400 Subject: [Freeswitch-users] Unexpected silence on the call Message-ID: Note: My skills with FS are mediocre at best, and my SIP knowledge is very green, so please bear with me. We have been experiencing an inconsistent problem?with our FS conference server. On occassion, a person on the call will go silent. They remain connected to the call, but they have no voice (their channel does not appear muted in the CLI). They can hear everyone else on the call. If they hangup and reconnect to the call everything is okay. After performing some digging on a recent occurance of this behavior, it appeared to coincide with re-invites being sent from our platform back through the SIP Tandem. >From my understanding, the re-invite is initiated because of the session_expire parameter having been set to 15 minutes (1800 seconds). Our current assumption is that the 1800 seconds is expiring, and we are attempting to initiate the re-invite to the carrier, and the routing is not executing properly, perhaps? Has anyone experienced this problem? Is there a way to override the request to set the session_expire, or something? VMK From brian at freeswitch.org Tue Aug 10 08:07:13 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Aug 2010 10:07:13 -0500 Subject: [Freeswitch-users] Unexpected silence on the call In-Reply-To: References: Message-ID: <90371BB1-B717-4D5F-A6E9-73D5131A3052@freeswitch.org> Without a sip trace we can only guess. /b On Aug 10, 2010, at 10:02 AM, VM Knott wrote: > Has anyone experienced this problem? > Is there a way to override the request to set the session_expire, or something? > > VMK From tculjaga at gmail.com Tue Aug 10 08:21:59 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 10 Aug 2010 17:21:59 +0200 Subject: [Freeswitch-users] read function returns corrupted values In-Reply-To: References: Message-ID: i think i found the source of a problem, its when playing chained prompts for the 1st time in DP. Any other usage of chained prompts further in DP results in no DTMF inputs messup. what i did was to combine prompts to be played into a single file and served that to read function. ... after that any other attempt to collect digits was fine ... T. On Tue, Aug 10, 2010 at 2:34 PM, Tihomir Culjaga wrote: > hello guys, > > i got a problem with read function ( > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read) in dialplan. > > > > > > > > > > > data="LANG_RETRIES=${expr(${LANG_RETRIES}+1)}"/> > > > > > > > > > > > data="sound_prefix=$${sounds_dir}/hr/HR/teta1"/> > > > > > > > data="sound_prefix=$${sounds_dir}/en/us/callie"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > when this part of dialplan executes and i enter the DTMF digit immediately > on 1st letter, the collected DTMF digits are not consistent... as there is a > memory overwrite somewhere .... > > > > > 2010-08-10 15:27:22.056491 [INFO] mod_dptools.c:946 ################# > LangSel ################\n > 2010-08-10 15:27:22.056491 [NOTICE] switch_core_session.c:1949 Execute > set(WELCOME_PR=${DEFAULT_LANG_PATH}ivr/welcome.wav) > EXECUTE sofia/external/38516659280 at 195.88.212.41set(WELCOME_PR=/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav) > 2010-08-10 15:27:22.056491 [DEBUG] mod_dptools.c:816 sofia/external/ > 38516659280 at 195.88.212.41 SET > [WELCOME_PR]=[/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav] > 2010-08-10 15:27:22.056491 [NOTICE] switch_core_session.c:1949 Execute > execute_extension(LangSelInput XML NXIVR) > EXECUTE sofia/external/38516659280 at 195.88.212.41execute_extension(LangSelInput XML NXIVR) > 2010-08-10 15:27:22.056491 [INFO] mod_dialplan_xml.c:418 Processing > 38516659280->LangSelInput in context NXIVR > Dialplan: sofia/external/38516659280 at 195.88.212.41 parsing > [NXIVR->LangSelInput] continue=false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) > [LangSelInput] destination_number(LangSelInput) =~ /^LangSelInput$/ > break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action > set(playback_delimiter=!) > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action > set(playback_terminators=#*0123456789) > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(myLANG=) > INLINE > EXECUTE sofia/external/38516659280 at 195.88.212.41 set(myLANG=) > 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ > 38516659280 at 195.88.212.41 SET [myLANG]=[UNDEF] > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action read(0 1 > ${WELCOME_PR}!${ERR_PR}!${LANG_PROMPT_STRING} myLANG ${LANG_TIMEOUT} ) > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action > set(LANG_RETRIES=${expr(${LANG_RETRIES}+1)}) INLINE > EXECUTE sofia/external/38516659280 at 195.88.212.41 set(LANG_RETRIES=1) > 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ > 38516659280 at 195.88.212.41 SET [LANG_RETRIES]=[1] > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action > execute_extension(LangSel XML NXIVR) > 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute > set(playback_delimiter=!) > EXECUTE sofia/external/38516659280 at 195.88.212.41 set(playback_delimiter=!) > 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ > 38516659280 at 195.88.212.41 SET [playback_delimiter]=[!] > 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute > set(playback_terminators=#*0123456789) > EXECUTE sofia/external/38516659280 at 195.88.212.41set(playback_terminators=#*0123456789) > 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ > 38516659280 at 195.88.212.41 SET [playback_terminators]=[#*0123456789] > 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute > read(0 1 ${WELCOME_PR}!${ERR_PR}!${LANG_PROMPT_STRING} myLANG > ${LANG_TIMEOUT} ) > EXECUTE sofia/external/38516659280 at 195.88.212.41 read(0 1 > /usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav!!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/for_croatian.wav!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/press.wav!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/one.wav!/usr/local/freeswitch/sounds/en/us/callie/ivr/for_en_press2.wav!/usr/local/freeswitch/sounds/de/de/helge/ivr/for_german_press3.wav!/usr/local/freeswitch/sounds/it/it/ambra/ivr/for_italian_press4.wav!/usr/local/freeswitch/sounds/fr/fr/celine/ivr/for_french_press5.wav > myLANG 10000 ) > 2010-08-10 15:27:22.059474 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-08-10 15:27:22.872525 [DEBUG] switch_ivr_play_say.c:1444 done playing > file > 2010-08-10 15:27:22.934526 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 1:752 > 2010-08-10 15:27:22.934526 [ERR] mod_native_file.c:74 Error opening > /usr/local/freeswitch/sounds/en/us/callie/.PCMA > 2010-08-10 15:27:23.892521 [DEBUG] switch_ivr_play_say.c:1444 done playing > file > 2010-08-10 15:27:25.012528 [DEBUG] switch_ivr_play_say.c:1444 done playing > file > 2010-08-10 15:27:25.832521 [DEBUG] switch_ivr_play_say.c:1444 done playing > file > 2010-08-10 15:27:29.792528 [DEBUG] switch_ivr_play_say.c:1444 done playing > file > 2010-08-10 15:27:31.872524 [DEBUG] switch_ivr_play_say.c:1444 done playing > file > 2010-08-10 15:27:34.492524 [DEBUG] switch_ivr_play_say.c:1444 done playing > file > 2010-08-10 15:27:37.152526 [DEBUG] switch_ivr_play_say.c:1444 done playing > file > 2010-08-10 15:27:37.152526 [NOTICE] switch_core_session.c:1949 Execute > execute_extension(LangSel XML NXIVR) > EXECUTE sofia/external/38516659280 at 195.88.212.41 execute_extension(LangSel > XML NXIVR) > 2010-08-10 15:27:37.152526 [INFO] mod_dialplan_xml.c:418 Processing > 38516659280->LangSel in context NXIVR > Dialplan: sofia/external/38516659280 at 195.88.212.41 parsing > [NXIVR->LangSel] continue=false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > destination_number(LangSel) =~ /^LangSel$/ break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] > ${myLANG}(1^?@? ) =~ /^1$/ break=on-true > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > destination_number(LangSel) =~ /^LangSel$/ break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] > ${myLANG}(1^?@? ) =~ /^2$/ break=on-true > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > destination_number(LangSel) =~ /^LangSel$/ break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] > ${myLANG}(1^?@? ) =~ /^3$/ break=on-true > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > destination_number(LangSel) =~ /^LangSel$/ break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] > ${myLANG}(1^?@? ) =~ /^4$/ break=on-true > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > destination_number(LangSel) =~ /^LangSel$/ break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] > ${myLANG}(1^?@? ) =~ /^5$/ break=on-true > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > destination_number(LangSel) =~ /^LangSel$/ break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] > ${myLANG}(1^?@? ) =~ /^[06789]$/ break=on-true > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > destination_number(LangSel) =~ /^LangSel$/ break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] > ${myLANG}(1^?@? ) =~ /^$/ break=on-true > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > destination_number(LangSel) =~ /^LangSel$/ break=on-false > Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] > ${myLANG}(1^?@? ) =~ /^.*$/ break=on-true > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action log(INFO > ################# WRONG LANG SEL ################\n) > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(language=hr) > > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action > set(ERR_PR=${WRONG_LANG_PROMPT}) > Dialplan: sofia/external/38516659280 at 195.88.212.41 Action > execute_extension(LangRetries XML NXIVR) > 2010-08-10 15:27:37.154487 [NOTICE] switch_core_session.c:1949 Execute > log(INFO ################# WRONG LANG SEL ################\n) > EXECUTE sofia/external/38516659280 at 195.88.212.41 log(INFO > ################# WRONG LANG SEL ################\n) > 2010-08-10 15:27:37.154487 [INFO] mod_dptools.c:946 ################# WRONG > LANG SEL ################\n > 2010-08-10 15:27:37.154487 [NOTICE] switch_core_session.c:1949 Execute > set(language=hr) > EXECUTE sofia/external/38516659280 at 195.88.212.41 set(language=hr) > > > you can see myLANG variable holds "1^?@? " ... of course i dialed 1 as seen > in the log (switch_rtp.c:2428 RTP RECV DTMF 1:752). > > > now, i did some extra testing and found that it happens only when i use a > playback_separator to append multiple files to be played.... > > i checked switch_ivr_read and switch_ivr_collect_digits_count functions > but everything seems to be ok there... > > > > can anyone help locating the problem ? > > > > > Ps: im running > > > freeswitch at cxss01> version > > FreeSWITCH Version 1.0.6 (svn-exported) > > tried with Revision: 17032 before this one but seems to be the same.... > > > > [tculjaga at cxss01 src]$ cat /etc/issue > CentOS release 5.4 (Final) > Kernel \r on an \m > > [tculjaga at cxss01 src]$ > > > > [tculjaga at cxss01 src]$ uname -a > Linux cxss01 2.6.18-164.el5 #1 SMP Thu Sep 3 03:28:30 EDT 2009 x86_64 > x86_64 x86_64 GNU/Linux > [tculjaga at cxss01 src]$ > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/65be693f/attachment-0001.html From tgraziano at myitdepartment.net Tue Aug 10 08:26:48 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Tue, 10 Aug 2010 11:26:48 -0400 Subject: [Freeswitch-users] Unexpected silence on the call In-Reply-To: References: Message-ID: a quick way to have the caller's UA send the reinvite might be to have them place the call on hold and then resume the call. If this works, it's pretty apparent you do indeed have a reinvite issue though. The carrier might be expecting an empty packet for the rtp keepalive. It might be good to ask the carrier what they would prefer in the following: keepalive interval, method for sip keepalive, method for rtp keepalive and compare that with your settings (as well as a trace). > > Our current assumption is that the 1800 seconds is expiring, and we > are attempting to initiate the re-invite to the carrier, and the > routing is not executing properly, perhaps? > > Has anyone experienced this problem? > Is there a way to override the request to set the session_expire, or > something? > > VMK > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/d661ebf5/attachment.html From kris at kriskinc.com Tue Aug 10 08:29:10 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 10 Aug 2010 11:29:10 -0400 Subject: [Freeswitch-users] Question about srtp secured B-Leg Message-ID: <77a8d8b649751c98e3c9daaacf39afb3@mail.gmail.com> It's probably not as clean as you'd like and there may be a better way to do it but why don't you just try calling the device twice, the first time with SRTP/SDES and the second (failover) time without? I'd think this would be compatible with the largest number of devices (whether using SRTP or not). -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Tue Aug 10 09:21:06 2010 Subject: [Freeswitch-users] Question about srtp secured B-Leg -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have late codec negotiation enabled and using Snom devices for encrypted calls (inernal only). When sdp of A-leg (Snom) has a=crypto I export sip_secure_media=true in my dialplan, so that B-leg (Snom) is also encrypted via srtp. That works nicely. When the B-Leg doesn't support srtp I got an "incompatible_device" error. This is also ok so far. Unfortunately this way you can't use Snom's optional srtp mechanism which consists simply of two media profiles in A-Leg's SDP (first RTP/SAVP then RTP/AVP) so the target can choose to encrypt or not. I don't know whether this is RFC conform or not nor if this makes really sense. I would like to be able to call a non encrypting ATA for checking for eaxample fax devices by simply calling them. My question is now, is there a way to tell FS sending outbound calls also with a non crypto offer within same sdp to B-Leg, so that it can choose to encrypt or not? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMYVJB4tZeNddg3dwRApBGAJ9d4keb5RQqMlS5sJ6jfN3VLAQkhwCeO3/x QlwcZMCW2ipsAIjV7cet2aU= =qzpR -----END PGP SIGNATURE----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From robert.hadley at teotech.com Tue Aug 10 08:51:01 2010 From: robert.hadley at teotech.com (Robert Hadley) Date: Tue, 10 Aug 2010 08:51:01 -0700 Subject: [Freeswitch-users] Leg_timeout question In-Reply-To: References: <97D5CE44FD7A4B7F9D58FC89BF18809F@greyhawk.tonecommander.com> Message-ID: <929233C2F3794B7B9A6D9DD2DC548B32@greyhawk.tonecommander.com> Hi Paul, I would agree with what's been mentioned so far about what should work :-) The hangup and the hangup-after-bridge should have worked, I would suspect (but don't anything about) an interaction with this statement-try removing it: Another thing to try: To hangup when the bridge fails you set this back to false. Have fun, Robert _____ From: paul gore [mailto:paul.gore.j at gmail.com] Sent: Tuesday, August 10, 2010 4:53 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Leg_timeout question I should clarify myself, I talk about "no answer" scenario here, so wouldn't have any effect, correct? So in this case I assume should properly hangup the call, but it does not. On Mon, Aug 9, 2010 at 10:53 PM, paul gore wrote: Thank you, that worked! But I have another problem now. If I don't want to forward to voice mail at the end, but just hangup, it does not work - call repeats 3 times and no hangup gets send to the call originator, even with "hangup" action at the end. Like this: Is this fixable? On Mon, Aug 9, 2010 at 11:46 AM, Robert Hadley wrote: Try _____ From: paul gore [mailto:paul.gore.j at gmail.com] Sent: Sunday, August 08, 2010 7:39 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Leg_timeout question Hi there, I am trying to accomplish a pretty simple thing in FS dialplan - ring one or multiple destinations with individual timeout per destination, if nobody answers - forward to a voicemail. Here is dial plan I use: What happens is incoming call from my provider rings ext 1002, but instead of going to the VM after 20 sec and no answer FS sends 602 (alloted_timeout) and the ring repeats again and so 3 times, after which call dies. But never goes to voice mail. What I am doing wrong? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/c4681bb4/attachment-0001.html From chris.veazey at gmail.com Tue Aug 10 09:05:22 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Tue, 10 Aug 2010 11:05:22 -0500 Subject: [Freeswitch-users] FS processing 302 Message-ID: <4508448D788749F6B25886D626C17C28@left> Freeswitch via the external profile(port 5080) is sending an Invite and receiving a 302 response. Should Freeswitch by default be able to process the maddr of a 302 redirect? SIP/2.0 302 Moved temporarily Via:SIP/2.0/UDP 10.11.0.80:5080;branch=z9hG4bKZ1H8r8c2rmHpc;rport From:"9995551000000";tag=ZXj rggj4HymDe To:;tag=1812200698-1281407077651 Call-ID:43a1c9c6-1ec9-122e-b19d-0018512d505c CSeq:367474 INVITE Contact:;q=0.5;ton=PUBLIC;ct=NIL;cat=OTHER Content-Length:0 Thanks Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/a2697e7f/attachment.html From brian at freeswitch.org Tue Aug 10 09:22:25 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Aug 2010 11:22:25 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <4508448D788749F6B25886D626C17C28@left> References: <4508448D788749F6B25886D626C17C28@left> Message-ID: <436A91D0-DADF-4BD3-A5D1-AF0D6EA02CB6@freeswitch.org> Not sure... isn't that multicast? On Aug 10, 2010, at 11:05 AM, Chris Veazey wrote: > Freeswitch via the external profile(port 5080) is sending an Invite and receiving a 302 response. > > Should Freeswitch by default be able to process the maddr of a 302 redirect? From kris at kriskinc.com Tue Aug 10 09:35:16 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 10 Aug 2010 12:35:16 -0400 Subject: [Freeswitch-users] FS processing 302 Message-ID: It's supposed to be but I've seen implementations use it when attempting to indicate a specific address to signal to. Freeswitch did "the right thing" and used the address from maddr in my tests so it should work for you. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Tue Aug 10 12:22:25 2010 Subject: Re: [Freeswitch-users] FS processing 302 Not sure... isn't that multicast? On Aug 10, 2010, at 11:05 AM, Chris Veazey wrote: > Freeswitch via the external profile(port 5080) is sending an Invite and > receiving a 302 response. > > Should Freeswitch by default be able to process the maddr of a 302 > redirect? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From tculjaga at gmail.com Tue Aug 10 09:36:58 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 10 Aug 2010 18:36:58 +0200 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <436A91D0-DADF-4BD3-A5D1-AF0D6EA02CB6@freeswitch.org> References: <4508448D788749F6B25886D626C17C28@left> <436A91D0-DADF-4BD3-A5D1-AF0D6EA02CB6@freeswitch.org> Message-ID: On Tue, Aug 10, 2010 at 6:22 PM, Brian West wrote: > Not sure... isn't that multicast? > > On Aug 10, 2010, at 11:05 AM, Chris Veazey wrote: > > > Freeswitch via the external profile(port 5080) is sending an Invite and > receiving a 302 response. > > > > Should Freeswitch by default be able to process the maddr of a 302 > redirect? > > > maddr has precedence. If its present in a contact header, any subsequent message needs to be sent to maddr address. T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/fbfbfba5/attachment.html From errotan at elder.hu Tue Aug 10 09:37:38 2010 From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=) Date: Tue, 10 Aug 2010 18:37:38 +0200 Subject: [Freeswitch-users] SPA8000 In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C55A1@server1.st.local> References: <81C2CEF80046FB4F863A60D4347DD33A0C559E@server1.st.local> <81C2CEF80046FB4F863A60D4347DD33A0C55A1@server1.st.local> Message-ID: <201008101837.38236.errotan@elder.hu> Hi. SPA8000 works perfectly for me. Do you have NAT involved ? If so have you set NAT mapping and NAT keep alive on ? Please tell how you configured the SPA8000 because without info we can't help. 2010. augusztus 10. 13:00:49 d?tummal Erkan ?nl? az al?bbiakat ?rta: > I think the problem is in FreeSwitch. > > If I use other voip server that works. > > > > For example VoipSwitch > > > > > > The SPA8000 is registered. But also by VoipSwitch as normal client don't > work. You must select Common Client > > What is common client??? What kind of type is that? Is there exist also in > FreeSwitch? > > Can I create a user in different type or forms. > > > > > > Look also at "Sip Messages Recv" and "Sip Bytes Recv:" > > If I used the freeswitch account they stay everytime on zero (0). > > So I think the freeswitch can not handle it to send the messages to the > SPA8000. > > So the SPA8000 get nothing from FreeSwitch. > > Maybe a NAT setting in my FreeSwitch is not correct? > > But Softphones are worked. > > > > Hope the information's give you more ideas. > > > > Thank you for the help. > > > > Kind regards > > Erkan > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erkan > ?nl? Sent: Tuesday, August 10, 2010 12:09 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] SPA8000 > > > > Hi all, > > > > I have a Linksys SPA8000 but I can't register a line. > > With softphones the user is work. > > But with SPA8000 don't register. > > > > The register to other provider is worked. So I can configure the SPA8000 > correctly. > > > > > > The sample user can be registered with softphone and some hardphones but > with SPA8000 can't register. > > Maybe I must change the XML in the directory folder. > > > > Everybody have idea? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > From chris.veazey at gmail.com Tue Aug 10 09:48:05 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Tue, 10 Aug 2010 11:48:05 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: References: <4508448D788749F6B25886D626C17C28@left> <436A91D0-DADF-4BD3-A5D1-AF0D6EA02CB6@freeswitch.org> Message-ID: <397A2B08-88FB-453F-A0B5-C32D5549873A@gmail.com> That seems to be the issue. It is not sending a new Invitei to the maddr address contained in the 302 contact On Aug 10, 2010, at 11:36 AM, Tihomir Culjaga wrote: > > > On Tue, Aug 10, 2010 at 6:22 PM, Brian West wrote: > Not sure... isn't that multicast? > > On Aug 10, 2010, at 11:05 AM, Chris Veazey wrote: > > > Freeswitch via the external profile(port 5080) is sending an Invite and receiving a 302 response. > > > > Should Freeswitch by default be able to process the maddr of a 302 redirect? > > > > maddr has precedence. If its present in a contact header, any subsequent message needs to be sent to maddr address. > > T. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/aa2a42f2/attachment.html From tomabroad at gmail.com Tue Aug 10 10:15:06 2010 From: tomabroad at gmail.com (tom) Date: Tue, 10 Aug 2010 13:15:06 -0400 Subject: [Freeswitch-users] q: originating a call via http Message-ID: hi is that possible to have just a reg html link to FS to place a call? at best, it would ring first my extension, and when i take the hearer then it should ring the destination. thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/7704459a/attachment.html From stephen at mymessage.us Tue Aug 10 11:05:30 2010 From: stephen at mymessage.us (Stephen Cattaneo) Date: Tue, 10 Aug 2010 14:05:30 -0400 Subject: [Freeswitch-users] mod_voipcodecs build time Message-ID: When ever i build freeswitch on centos 5.5 i have to manually compile and move mod_voipcodecs. The make file exists in the directory it just never gets compiled and installed. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/88f8a934/attachment.html From vizentini at hotmail.com Tue Aug 10 11:07:07 2010 From: vizentini at hotmail.com (Paulo Vicentini) Date: Tue, 10 Aug 2010 18:07:07 +0000 Subject: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking In-Reply-To: References: , Message-ID: Hi,Basically:con.api("originate" ,"{ignore_early_media=true,absolute_codec_string=\'PCMA\'}sofia/test/phone at IP &park ");con.execute("play_and_get_digits", "3 5 3 5000 # welcome.wav sorry.wav myDigits \\d+",uuid);and then uuid_getvar uuid myDigits But con.execute("play_and_get_digits" is not blocking so that I can't grab myDigits.I put a breakpoint at switch_play_and_get_digits (session=0x90d5c18, min_digits=3,.. nevertheless con.execute("play_and_get_digits" returned and with extra debugging I realized that switch_api_execute (cmd=0xb76e2b5c "uuid_getvar"...happens before than#3 0x00cf2d08 in switch_core_session_execute_application_get_flags ( session=0xb76b5530, app=0xb76793b0 "play_and_get_digits", arg=0xb764ef68 "3 15 3 5000 # ivr-enter_ext_pound.wav sorry.wav myDigits \\d+", flags=0x0) at src/switch_core_session.c:1780#4 0x00d43d73 in switch_ivr_parse_event (session=0xb76b5530, event=0xb7652198) So that it seems a kind of race condition between ESL and the core core event system PV Date: Mon, 9 Aug 2010 17:42:42 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking On Mon, Aug 9, 2010 at 5:24 PM, Paulo Vicentini wrote: Hi, With ESL: this->execute ( 'play_and_get_digits',... $var .. $digits = $this->api_uuid_getvar ( $uuid, $var ); Paste your entire script, or at least a simple example that demonstrates the issue so that we can try it ourselves. Personally, I've not experienced this issue so I would like to know more about the circumstances where you're seeing this. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/bbef501c/attachment-0001.html From juanito1982 at gmail.com Tue Aug 10 11:10:55 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Tue, 10 Aug 2010 20:10:55 +0200 Subject: [Freeswitch-users] Different versions, different cpu load In-Reply-To: References: Message-ID: Latest revisions seem to solve this issue. I'll continue with the tests... 2010/8/6 Rupa Schomaker > Not really. If you have a test that shows "good/bad" we (devs) can find > the reason for a problem. git handles choosing versions for testing and you > handle the "good/bad" decision. The end result is "here is the git version > where performance changes" which tony or whomever can then look at the patch > and decide what he wants to do about it. > > 2010/8/6 Juan Antonio Iba?ez Santorum > > I thinkg it is very difficult to find it without a good knowledge of FS >> internals. Do you know if there is any way to make a profiling which could >> help? >> >> Regards >> >> 2010/8/5 Rupa Schomaker >> >>> sounds like a good opportunity to try git-bisect to locate where the >>> performance change occurs. >>> >>> >>> On Thu, Aug 5, 2010 at 10:37 AM, Eliot Gable < >>> egable+freeswitch at gmail.com > wrote: >>> >>>> My most recent testing as of about two weeks ago showed a 25% >>>> performance drop between the old SVN version I was running and the new >>>> GIT version. Initially, it seemed closer to your reported performance >>>> drop, but after moving the database to a ramdisk, it went to 25%. I am >>>> unsure of whether my initial testing was using the db on a ramdisk, so >>>> the drop could be higher if my previous testing did not have the db on >>>> a ramdisk. >>>> >>>> 2010/8/3 Juan Antonio Iba?ez Santorum : >>>> > No one experiment this same issue? >>>> > >>>> > I also noticed a high cpu consume on call hangup. >>>> > >>>> > Regards >>>> > >>>> > 2010/8/2 Juan Antonio Iba?ez Santorum >>>> >> >>>> >> Hello! >>>> >> >>>> >> I was been some test using one FS tarball version downloaded some >>>> weeks >>>> >> ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some >>>> problems with >>>> >> odbc connections managemens. Now I'm using a git version (FreeSWITCH >>>> Version >>>> >> 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve >>>> this >>>> >> problem but I noticed it has a high cpu load comparing with svn >>>> version. >>>> >> While I could manage more than 200 calls with a 75% CPU load into a >>>> dual >>>> >> core server using svn version, now, 50 calls consume this 75% cpu. I >>>> can see >>>> >> same modules are loaded (except new hash module needed in git >>>> version) and >>>> >> same scenario is used. I am not be able to find why now it uses more >>>> cpu >>>> >> than before. Any idea? >>>> >> Each testing call is a simple bridge to an external sip provider. >>>> >> >>>> >> Regards >>>> > >>>> > >>>> > _______________________________________________ >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> >>>> >>>> -- >>>> Eliot Gable >>>> >>>> "We do not inherit the Earth from our ancestors: we borrow it from our >>>> children." ~David Brower >>>> >>>> "I decided the words were too conservative for me. We're not borrowing >>>> from our children, we're stealing from them--and it's not even >>>> considered to be a crime." ~David Brower >>>> >>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to >>>> live; not live to eat.) ~Marcus Tullius Cicero >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> -Rupa >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/0d8ff28b/attachment.html From brian at freeswitch.org Tue Aug 10 11:13:49 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Aug 2010 13:13:49 -0500 Subject: [Freeswitch-users] mod_voipcodecs build time In-Reply-To: References: Message-ID: <33AD71CF-04F0-4889-B499-DC79A2C66053@freeswitch.org> Because you don't need tha tmodule anymore you need mod_spandsp. /b On Aug 10, 2010, at 1:05 PM, Stephen Cattaneo wrote: > When ever i build freeswitch on centos 5.5 i have to manually compile and move mod_voipcodecs. The make file exists in the directory it just never gets compiled and installed. > > > Thanks, > Stephen C > -All of my email addresses go to the same place > -Save Paper, think before you print > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris.veazey at gmail.com Tue Aug 10 11:19:23 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Tue, 10 Aug 2010 13:19:23 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: References: Message-ID: <563B3EF6EAFE41BC8506DDA53B45F7F8@left> Interesting that it worked for you. Currently my setup will not send a new invite to the maddr address in the 302 contact. Do you make any changes to get your setup to accept the 302 and send to the maddr address? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Tuesday, August 10, 2010 11:35 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS processing 302 It's supposed to be but I've seen implementations use it when attempting to indicate a specific address to signal to. Freeswitch did "the right thing" and used the address from maddr in my tests so it should work for you. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Tue Aug 10 12:22:25 2010 Subject: Re: [Freeswitch-users] FS processing 302 Not sure... isn't that multicast? On Aug 10, 2010, at 11:05 AM, Chris Veazey wrote: > Freeswitch via the external profile(port 5080) is sending an Invite and > receiving a 302 response. > > Should Freeswitch by default be able to process the maddr of a 302 > redirect? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From tculjaga at gmail.com Tue Aug 10 11:35:50 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 10 Aug 2010 20:35:50 +0200 Subject: [Freeswitch-users] q: originating a call via http In-Reply-To: References: Message-ID: On Tue, Aug 10, 2010 at 7:15 PM, tom wrote: > hi is that possible to have just a reg html link to FS to place a call? at > best, it would ring first my extension, and when i take the hearer then it > should ring the destination. > thx > use this to send commands to your FS via a web server: http://wiki.freeswitch.org/wiki/Mod_xml_rpc ... after that its just a dialplan thing. cheers! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/e56fde51/attachment.html From kris at kriskinc.com Tue Aug 10 11:42:46 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 10 Aug 2010 14:42:46 -0400 Subject: [Freeswitch-users] FS processing 302 Message-ID: No, it "just worked". -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: 'FreeSWITCH Users Help' Sent: Tue Aug 10 14:19:23 2010 Subject: Re: [Freeswitch-users] FS processing 302 Interesting that it worked for you. Currently my setup will not send a new invite to the maddr address in the 302 contact. Do you make any changes to get your setup to accept the 302 and send to the maddr address? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Tuesday, August 10, 2010 11:35 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS processing 302 It's supposed to be but I've seen implementations use it when attempting to indicate a specific address to signal to. Freeswitch did "the right thing" and used the address from maddr in my tests so it should work for you. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Tue Aug 10 12:22:25 2010 Subject: Re: [Freeswitch-users] FS processing 302 Not sure... isn't that multicast? On Aug 10, 2010, at 11:05 AM, Chris Veazey wrote: > Freeswitch via the external profile(port 5080) is sending an Invite and > receiving a 302 response. > > Should Freeswitch by default be able to process the maddr of a 302 > redirect? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From peter.olsson at visionutveckling.se Tue Aug 10 11:44:35 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 10 Aug 2010 20:44:35 +0200 Subject: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking In-Reply-To: References: , , Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC058102@cooper> Are you using sockets in asynchronous mode? If not, execute() will just queue up to the internal queue and then return imediately. In that case you will need to get the event when the execution has actually been done. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Paulo Vicentini [vizentini at hotmail.com] Skickat: den 10 augusti 2010 20:07 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking Hi, Basically: con.api("originate" ,"{ignore_early_media=true,absolute_codec_string=\'PCMA\'}sofia/test/phone at IP &park "); con.execute("play_and_get_digits", "3 5 3 5000 # welcome.wav sorry.wav myDigits \\d+",uuid); and then uuid_getvar uuid myDigits But con.execute(" play_and_get_digits" is not blocking so that I can't grab myDigits. I put a breakpoint at switch_play_and_get_digits (session=0x90d5c18, min_digits=3,.. nevertheless con.execute("play_and_get_digits" returned and with extra debugging I realized that switch_api_execute (cmd=0xb76e2b5c "uuid_getvar"...happens before than #3 0x00cf2d08 in switch_core_session_execute_application_get_flags ( session=0xb76b5530, app=0xb76793b0 "play_and_get_digits", arg=0xb764ef68 "3 15 3 5000 # ivr-enter_ext_pound.wav sorry.wav myDigits \\d+", flags=0x0) at src/switch_core_session.c:1780 #4 0x00d43d73 in switch_ivr_parse_event (session=0xb76b5530, event=0xb7652198) So that it seems a kind of race condition between ESL and the core core event system PV ________________________________ Date: Mon, 9 Aug 2010 17:42:42 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking On Mon, Aug 9, 2010 at 5:24 PM, Paulo Vicentini > wrote: Hi, With ESL: this->execute ( 'play_and_get_digits',... $var .. $digits = $this->api_uuid_getvar ( $uuid, $var ); Paste your entire script, or at least a simple example that demonstrates the issue so that we can try it ourselves. Personally, I've not experienced this issue so I would like to know more about the circumstances where you're seeing this. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c61978b32931921810807! From macedoslm at gmail.com Tue Aug 10 11:47:40 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Tue, 10 Aug 2010 15:47:40 -0300 Subject: [Freeswitch-users] Video Support In-Reply-To: References: <725A1E83-CC4E-482B-B20F-BBD913381DB2@avgs.ca> Message-ID: I've done another test. This time I'm trying to record my video. "" FS sent to me a 200 OK answer with a strange line in the SDP. "m=video 0 RTP/AVP 19" What does this mean? I'm sending the network trace. Thanks, -- Samuel Macedo On 10 August 2010 10:00, Samuel Macedo wrote: > I have an IP phone and support for H.263 codec. First of all I want to make > video calls using this IP phones, than I will try the video conferences. I > had succeeded doing video calls with bypass_media flag enabled. I couldn't > do the same with bypass_media disabled, my FS can't negotiate the video > codecs. > > I checked the "mod_h26x" and "mod_fsv" and they are enabled. I am also > sending the network trace of my tests. > > Is there any config in FS to make the video calls? > > Thanks, > -- > Samuel Macedo > > > > On 9 August 2010 20:44, Mathieu Rene wrote: > >> You can look at mod_fsv, you can record and play videos provided you call >> with an IP phone. mod_conference also will automatically switch the video >> stream to the one of the person speaking. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 2010-08-09, at 2:47 PM, Samuel Macedo wrote: >> >> > Hi, >> > >> > I've already read that FS supports video, but I've never seen anyone >> using or talking about how to use it. >> > Does FS support video conferences? Or only video calls? Is there any >> documentation about this subject? >> > >> > Thanks, >> > -- >> > Samuel Macedo >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/0a3ce50a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: FSV_record.pcap Type: application/octet-stream Size: 155432 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/0a3ce50a/attachment-0001.obj From brian at freeswitch.org Tue Aug 10 11:51:24 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Aug 2010 13:51:24 -0500 Subject: [Freeswitch-users] Video Support In-Reply-To: References: <725A1E83-CC4E-482B-B20F-BBD913381DB2@avgs.ca> Message-ID: On Aug 10, 2010, at 1:47 PM, Samuel Macedo wrote: > > FS sent to me a 200 OK answer with a strange line in the SDP. > "m=video 0 RTP/AVP 19" > > What does this mean? From brian at freeswitch.org Tue Aug 10 11:51:27 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Aug 2010 13:51:27 -0500 Subject: [Freeswitch-users] Video Support In-Reply-To: References: <725A1E83-CC4E-482B-B20F-BBD913381DB2@avgs.ca> Message-ID: <36D6F460-1942-477B-9C37-EAA7C04C41D6@freeswitch.org> it means it rejected the media for that m line. /b On Aug 10, 2010, at 1:47 PM, Samuel Macedo wrote: > > FS sent to me a 200 OK answer with a strange line in the SDP. > "m=video 0 RTP/AVP 19" > > What does this mean? From peter.olsson at visionutveckling.se Tue Aug 10 12:00:31 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 10 Aug 2010 21:00:31 +0200 Subject: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57DC058102@cooper> References: , , , <549CFEF87AEDE841A38E9D15EAB4C04C57DC058102@cooper> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC058103@cooper> Sorry - I meant synchronous mode... sync mode waits for the command to complete. async mode returns imediately and you will have to wait for the right event to show up. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Peter Olsson [peter.olsson at visionutveckling.se] Skickat: den 10 augusti 2010 20:44 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking Are you using sockets in asynchronous mode? If not, execute() will just queue up to the internal queue and then return imediately. In that case you will need to get the event when the execution has actually been done. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Paulo Vicentini [vizentini at hotmail.com] Skickat: den 10 augusti 2010 20:07 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking Hi, Basically: con.api("originate" ,"{ignore_early_media=true,absolute_codec_string=\'PCMA\'}sofia/test/phone at IP &park "); con.execute("play_and_get_digits", "3 5 3 5000 # welcome.wav sorry.wav myDigits \\d+",uuid); and then uuid_getvar uuid myDigits But con.execute(" play_and_get_digits" is not blocking so that I can't grab myDigits. I put a breakpoint at switch_play_and_get_digits (session=0x90d5c18, min_digits=3,.. nevertheless con.execute("play_and_get_digits" returned and with extra debugging I realized that switch_api_execute (cmd=0xb76e2b5c "uuid_getvar"...happens before than #3 0x00cf2d08 in switch_core_session_execute_application_get_flags ( session=0xb76b5530, app=0xb76793b0 "play_and_get_digits", arg=0xb764ef68 "3 15 3 5000 # ivr-enter_ext_pound.wav sorry.wav myDigits \\d+", flags=0x0) at src/switch_core_session.c:1780 #4 0x00d43d73 in switch_ivr_parse_event (session=0xb76b5530, event=0xb7652198) So that it seems a kind of race condition between ESL and the core core event system PV ________________________________ Date: Mon, 9 Aug 2010 17:42:42 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking On Mon, Aug 9, 2010 at 5:24 PM, Paulo Vicentini > wrote: Hi, With ESL: this->execute ( 'play_and_get_digits',... $var .. $digits = $this->api_uuid_getvar ( $uuid, $var ); Paste your entire script, or at least a simple example that demonstrates the issue so that we can try it ourselves. Personally, I've not experienced this issue so I would like to know more about the circumstances where you're seeing this. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c619f6232936662915155! From macedoslm at gmail.com Tue Aug 10 12:18:20 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Tue, 10 Aug 2010 16:18:20 -0300 Subject: [Freeswitch-users] Video Support In-Reply-To: <36D6F460-1942-477B-9C37-EAA7C04C41D6@freeswitch.org> References: <725A1E83-CC4E-482B-B20F-BBD913381DB2@avgs.ca> <36D6F460-1942-477B-9C37-EAA7C04C41D6@freeswitch.org> Message-ID: I update vars.xml with: And it's working fine. I'll try the video conference now. Thanks, -- Samuel Macedo On 10 August 2010 15:51, Brian West wrote: > it means it rejected the media for that m line. > > /b > > On Aug 10, 2010, at 1:47 PM, Samuel Macedo wrote: > > > > > FS sent to me a 200 OK answer with a strange line in the SDP. > > "m=video 0 RTP/AVP 19" > > > > What does this mean? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/329b7f77/attachment.html From vizentini at hotmail.com Tue Aug 10 12:36:38 2010 From: vizentini at hotmail.com (Paulo Vicentini) Date: Tue, 10 Aug 2010 19:36:38 +0000 Subject: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57DC058103@cooper> References: , , , , , , <549CFEF87AEDE841A38E9D15EAB4C04C57DC058102@cooper>, <549CFEF87AEDE841A38E9D15EAB4C04C57DC058103@cooper> Message-ID: Is there a way to explicitly set synchronous mode?I am just running such commands as shown previously and I think it is already in sync mode. Thanks > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Tue, 10 Aug 2010 21:00:31 +0200 > Subject: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking > > Sorry - I meant synchronous mode... > > sync mode waits for the command to complete. async mode returns imediately and you will have to wait for the right event to show up. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Peter Olsson [peter.olsson at visionutveckling.se] > Skickat: den 10 augusti 2010 20:44 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking > > Are you using sockets in asynchronous mode? If not, execute() will just queue up to the internal queue and then return imediately. In that case you will need to get the event when the execution has actually been done. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Paulo Vicentini [vizentini at hotmail.com] > Skickat: den 10 augusti 2010 20:07 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking > > Hi, > Basically: > con.api("originate" ,"{ignore_early_media=true,absolute_codec_string=\'PCMA\'}sofia/test/phone at IP &park "); > con.execute("play_and_get_digits", "3 5 3 5000 # welcome.wav sorry.wav myDigits \\d+",uuid); > and then uuid_getvar uuid myDigits > > But con.execute(" > play_and_get_digits" is not blocking so that I can't grab > myDigits. > I put a breakpoint at switch_play_and_get_digits (session=0x90d5c18, min_digits=3,.. nevertheless con.execute("play_and_get_digits" returned and with extra debugging I realized that switch_api_execute (cmd=0xb76e2b5c "uuid_getvar"...happens before than > #3 0x00cf2d08 in switch_core_session_execute_application_get_flags ( > session=0xb76b5530, app=0xb76793b0 "play_and_get_digits", > arg=0xb764ef68 "3 15 3 5000 # ivr-enter_ext_pound.wav sorry.wav myDigits \\d+", flags=0x0) at src/switch_core_session.c:1780 > #4 0x00d43d73 in switch_ivr_parse_event (session=0xb76b5530, event=0xb7652198) > > So that it seems a kind of race condition between ESL and the > core > core event system > > PV > > ________________________________ > Date: Mon, 9 Aug 2010 17:42:42 -0700 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking > > > > On Mon, Aug 9, 2010 at 5:24 PM, Paulo Vicentini > wrote: > Hi, With ESL: this->execute ( 'play_and_get_digits',... $var .. $digits = $this->api_uuid_getvar ( $uuid, $var ); > > Paste your entire script, or at least a simple example that demonstrates the issue so that we can try it ourselves. Personally, I've not experienced this issue so I would like to know more about the circumstances where you're seeing this. > -MC > > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4c619f6232936662915155! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/1ca2f3e3/attachment.html From peter.olsson at visionutveckling.se Tue Aug 10 12:57:19 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 10 Aug 2010 21:57:19 +0200 Subject: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking Message-ID: <37281574-2846-4A69-92B6-3B1747C6DC40@visionutveckling.se> In outbound mode it's default sync mode. In inbound mode it's always async, and you will need to handle events. /Peter ----- Reply message ----- Fr?n: "Paulo Vicentini" Datum: tis, aug 10, 2010 21:44 Rubrik: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking Till: "freeswitch-users at lists.freeswitch.org" Is there a way to explicitly set synchronous mode? I am just running such commands as shown previously and I think it is already in sync mode. Thanks > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Tue, 10 Aug 2010 21:00:31 +0200 > Subject: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking > > Sorry - I meant synchronous mode... > > sync mode waits for the command to complete. async mode returns imediately and you will have to wait for the right event to show up. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Peter Olsson [peter.olsson at visionutveckling.se] > Skickat: den 10 augusti 2010 20:44 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking > > Are you using sockets in asynchronous mode? If not, execute() will just queue up to the internal queue and then return imediately. In that case you will need to get the event when the execution has actually been done. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Paulo Vicentini [vizentini at hotmail.com] > Skickat: den 10 augusti 2010 20:07 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking > > Hi, > Basically: > con.api("originate" ,"{ignore_early_media=true,absolute_codec_string=\'PCMA\'}sofia/test/phone at IP &park "); > con.execute("play_and_get_digits", "3 5 3 5000 # welcome.wav sorry.wav myDigits \\d+",uuid); > and then uuid_getvar uuid myDigits > > But con.execute(" > play_and_get_digits" is not blocking so that I can't grab > myDigits. > I put a breakpoint at switch_play_and_get_digits (session=0x90d5c18, min_digits=3,.. nevertheless con.execute("play_and_get_digits" returned and with extra debugging I realized that switch_api_execute (cmd=0xb76e2b5c "uuid_getvar"...happens before than > #3 0x00cf2d08 in switch_core_session_execute_application_get_flags ( > session=0xb76b5530, app=0xb76793b0 "play_and_get_digits", > arg=0xb764ef68 "3 15 3 5000 # ivr-enter_ext_pound.wav sorry.wav myDigits \\d+", flags=0x0) at src/switch_core_session.c:1780 > #4 0x00d43d73 in switch_ivr_parse_event (session=0xb76b5530, event=0xb7652198) > > So that it seems a kind of race condition between ESL and the > core > core event system > > PV > > ________________________________ > Date: Mon, 9 Aug 2010 17:42:42 -0700 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking > > > > On Mon, Aug 9, 2010 at 5:24 PM, Paulo Vicentini > wrote: > Hi, With ESL: this->execute ( 'play_and_get_digits',... $var .. $digits = $this->api_uuid_getvar ( $uuid, $var ); > > Paste your entire script, or at least a simple example that demonstrates the issue so that we can try it ourselves. Personally, I've not experienced this issue so I would like to know more about the circumstances where you're seeing this. > -MC > > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org !DSPAM:4c61ac3032936707717627! From jmesquita at freeswitch.org Tue Aug 10 13:03:35 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 10 Aug 2010 17:03:35 -0300 Subject: [Freeswitch-users] Conference Conrol GUI In-Reply-To: References: Message-ID: If you are interested in developing a tool, I have something partially done that might fit your needs. Contact me off list if you are interested. And just so I don't get ppls minds wondering, I won't charge for it, of course. Regards, Jo?o Mesquita On Tue, Aug 10, 2010 at 5:22 AM, Mike wrote: > Hi, > > I'm looking to evaluate gui's for FreeSWITCH real-time conference contol > (muting of participants, locking the conference etc.) > > I'm currently looking at FusionPBX - can anyone recommend any other > packages worth looking at that support this functionality? > > Mike > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/ded7f807/attachment.html From jmesquita at freeswitch.org Tue Aug 10 13:08:17 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 10 Aug 2010 17:08:17 -0300 Subject: [Freeswitch-users] CentOS 5.x and Python 2.6: segmentation fault importing ESL In-Reply-To: References: Message-ID: Neil, that's because ESL is being compiled against 2.4 and not 2.6. It will use whatever is set on your path. You don't have to change anything on SWIG. You only have to change what is being generated by the python-config script located in ${SRC}/libs/esl/python Hope that helps. Jo?o Mesquita On Tue, Aug 10, 2010 at 4:12 AM, Neil Patel wrote: > Hi All, > > I am using Python 2.6 on my CentOS installation, installed in parallel to > the standard Python 2.4. I've compiled FS's python ESL module using 'make > pymod', but the ESL python module causes a segfault when I try to import it > in Python 2.6 (imports fine in 2.4). I'm guessing this is because the ESL > module was compiled for Python 2.4. How do I change the ESL config/makefiles > to compile for 2.6? I'm not familiar with SWIG, is there some configuration > I need to change with it to use python2.6 executable? > > Thanks, > Neil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/b808bf0d/attachment.html From brian at freeswitch.org Tue Aug 10 13:14:30 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Aug 2010 15:14:30 -0500 Subject: [Freeswitch-users] [freeswitch-users] play_and_get_digits is not blocking In-Reply-To: <37281574-2846-4A69-92B6-3B1747C6DC40@visionutveckling.se> References: <37281574-2846-4A69-92B6-3B1747C6DC40@visionutveckling.se> Message-ID: <247DE761-C3E8-4BC6-A716-A891E00EC682@freeswitch.org> I swear their is a way to do this to force per command or something but I can't recall off the top of my head what it is... I'll have to dig for it. /b On Aug 10, 2010, at 2:57 PM, Peter Olsson wrote: > In outbound mode it's default sync mode. In inbound mode it's always async, and you will need to handle events. > > /Peter From stas at khirman.com Tue Aug 10 13:24:50 2010 From: stas at khirman.com (Stas Khirman) Date: Tue, 10 Aug 2010 13:24:50 -0700 Subject: [Freeswitch-users] First installation help Message-ID: <05cd01cb38ca$40603ff0$c120bfd0$@khirman.com> [first time user beg for help]... Trying my first FreeSwitch installation, I faced two problems : 1.) My host has two IP addresses. SoftPhone (Ekiga) running on the same host successfully registered with a first IP, but refuse to register with the second. Also, registration to 127.0.0.1 failing. How can I instruct FreeSWITCH to use BOTH addresses for registration. 2.) Somehow dialing test numbers (9196, 9199,etc) don't work properly.. FreeSWITCh reports: 2010-08-10 13:17:08.577884 [INFO] mod_dialplan_xml.c:418 Processing 1000->9196 in context default 2010-08-10 13:17:08.619705 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000 at 10.0.2.15 to enum[9196 at default] 2010-08-10 13:17:09.703867 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2010-08-10 13:17:09.703867 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000 at 10.0.2.15 [CS_ROUTING] [NO_ROUTE_DESTINATION] Any help is deeply appreciated! Regards Stas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/bf7e8176/attachment.html From craigesmith at gmail.com Tue Aug 10 14:32:40 2010 From: craigesmith at gmail.com (Craig Smith) Date: Tue, 10 Aug 2010 17:32:40 -0400 Subject: [Freeswitch-users] Features question Message-ID: AASTRA Download user guides at http://www.aastra.com Hold: press HOLD button Retrieve Hold: press blinking line key Transfer (attended): press XFER, dial destination, press DIAL, wait for answer, announce call, hangup to complete transfer Transfer (blind): Press XFER, dial destination, press DIAL, hangup *Retrieve VM: See message wait light; dial *98 *Transfer caller directly to VM: Press XFER, dial **, dial extension number, hangup *Call directly to user`s VM to leave message: Dial **, dial extension number, leave message *Intercom call: Dial **, dial extension number, dial *1 (only works with phones that support hands-free auto-answer) 3-way call: With first call in progress, press CONF, dial 2nd party, press CONF again Would someone be so kind as to explain how to setup the items with a * next to them in FreeSWITCH. Thanks, Craig (Trying hard to be a FreeSWITCH convert) Smith From tomabroad at gmail.com Tue Aug 10 14:48:18 2010 From: tomabroad at gmail.com (tom) Date: Tue, 10 Aug 2010 17:48:18 -0400 Subject: [Freeswitch-users] q: originating a call via http In-Reply-To: References: Message-ID: thx On Tue, Aug 10, 2010 at 2:35 PM, Tihomir Culjaga wrote: > > > On Tue, Aug 10, 2010 at 7:15 PM, tom wrote: > >> hi is that possible to have just a reg html link to FS to place a call? at >> best, it would ring first my extension, and when i take the hearer then it >> should ring the destination. >> thx >> > > > use this to send commands to your FS via a web server: > http://wiki.freeswitch.org/wiki/Mod_xml_rpc ... after that its just a > dialplan thing. > > cheers! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/bfcd4e2a/attachment.html From msc at freeswitch.org Tue Aug 10 15:02:06 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Aug 2010 15:02:06 -0700 Subject: [Freeswitch-users] delay in establishing a call through the event interface In-Reply-To: <4C5F5BB8.8060903@jozep.com.au> References: <4C5F5BB8.8060903@jozep.com.au> Message-ID: On Sun, Aug 8, 2010 at 6:36 PM, marian szczepkowski wrote: > Hi > > I did some testing with freeswitch and I have an appreciable delay > between issueing an originate and the resulting call attempts by the > server. Is there a way to shorten this delay down or is this fixed in > some wierd form? > There shouldn't be a delay. What command, exactly are you sending over the socket? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/54824a3c/attachment-0001.html From msc at freeswitch.org Tue Aug 10 15:06:00 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Aug 2010 15:06:00 -0700 Subject: [Freeswitch-users] read function returns corrupted values In-Reply-To: References: Message-ID: Can you confirm if this behavior occurs on latest git? If so, could you please open a JIRA ticket? Thanks, MC On Tue, Aug 10, 2010 at 8:21 AM, Tihomir Culjaga wrote: > i think i found the source of a problem, > > its when playing > chained prompts for the 1st time in DP. Any other usage of chained prompts > further in DP results in no DTMF inputs messup. > what i did was to combine prompts to be played into a single file and > served that to read function. ... after that any other attempt to collect > digits was fine ... > > T. > > > > On Tue, Aug 10, 2010 at 2:34 PM, Tihomir Culjaga wrote: > >> hello guys, >> >> i got a problem with read function ( >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read) in dialplan. >> >> >> >> >> >> >> >> >> >> >> > data="LANG_RETRIES=${expr(${LANG_RETRIES}+1)}"/> >> >> >> >> >> >> >> >> >> >> >> > data="sound_prefix=$${sounds_dir}/hr/HR/teta1"/> >> >> >> >> >> >> >> > data="sound_prefix=$${sounds_dir}/en/us/callie"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> when this part of dialplan executes and i enter the DTMF digit immediately >> on 1st letter, the collected DTMF digits are not consistent... as there is a >> memory overwrite somewhere .... >> >> >> >> >> 2010-08-10 15:27:22.056491 [INFO] mod_dptools.c:946 ################# >> LangSel ################\n >> 2010-08-10 15:27:22.056491 [NOTICE] switch_core_session.c:1949 Execute >> set(WELCOME_PR=${DEFAULT_LANG_PATH}ivr/welcome.wav) >> EXECUTE sofia/external/38516659280 at 195.88.212.41set(WELCOME_PR=/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav) >> 2010-08-10 15:27:22.056491 [DEBUG] mod_dptools.c:816 sofia/external/ >> 38516659280 at 195.88.212.41 SET >> [WELCOME_PR]=[/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav] >> 2010-08-10 15:27:22.056491 [NOTICE] switch_core_session.c:1949 Execute >> execute_extension(LangSelInput XML NXIVR) >> EXECUTE sofia/external/38516659280 at 195.88.212.41execute_extension(LangSelInput XML NXIVR) >> 2010-08-10 15:27:22.056491 [INFO] mod_dialplan_xml.c:418 Processing >> 38516659280->LangSelInput in context NXIVR >> Dialplan: sofia/external/38516659280 at 195.88.212.41 parsing >> [NXIVR->LangSelInput] continue=false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) >> [LangSelInput] destination_number(LangSelInput) =~ /^LangSelInput$/ >> break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action >> set(playback_delimiter=!) >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action >> set(playback_terminators=#*0123456789) >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action set(myLANG=) >> INLINE >> EXECUTE sofia/external/38516659280 at 195.88.212.41 set(myLANG=) >> 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ >> 38516659280 at 195.88.212.41 SET [myLANG]=[UNDEF] >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action read(0 1 >> ${WELCOME_PR}!${ERR_PR}!${LANG_PROMPT_STRING} myLANG ${LANG_TIMEOUT} ) >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action >> set(LANG_RETRIES=${expr(${LANG_RETRIES}+1)}) INLINE >> EXECUTE sofia/external/38516659280 at 195.88.212.41 set(LANG_RETRIES=1) >> 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ >> 38516659280 at 195.88.212.41 SET [LANG_RETRIES]=[1] >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action >> execute_extension(LangSel XML NXIVR) >> 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute >> set(playback_delimiter=!) >> EXECUTE sofia/external/38516659280 at 195.88.212.41set(playback_delimiter=!) >> 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ >> 38516659280 at 195.88.212.41 SET [playback_delimiter]=[!] >> 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute >> set(playback_terminators=#*0123456789) >> EXECUTE sofia/external/38516659280 at 195.88.212.41set(playback_terminators=#*0123456789) >> 2010-08-10 15:27:22.058522 [DEBUG] mod_dptools.c:816 sofia/external/ >> 38516659280 at 195.88.212.41 SET [playback_terminators]=[#*0123456789] >> 2010-08-10 15:27:22.058522 [NOTICE] switch_core_session.c:1949 Execute >> read(0 1 ${WELCOME_PR}!${ERR_PR}!${LANG_PROMPT_STRING} myLANG >> ${LANG_TIMEOUT} ) >> EXECUTE sofia/external/38516659280 at 195.88.212.41 read(0 1 >> /usr/local/freeswitch/sounds/hr/HR/teta1/ivr/welcome.wav!!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/for_croatian.wav!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/press.wav!/usr/local/freeswitch/sounds/hr/HR/teta1/ivr/one.wav!/usr/local/freeswitch/sounds/en/us/callie/ivr/for_en_press2.wav!/usr/local/freeswitch/sounds/de/de/helge/ivr/for_german_press3.wav!/usr/local/freeswitch/sounds/it/it/ambra/ivr/for_italian_press4.wav!/usr/local/freeswitch/sounds/fr/fr/celine/ivr/for_french_press5.wav >> myLANG 10000 ) >> 2010-08-10 15:27:22.059474 [DEBUG] switch_ivr_play_say.c:1152 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2010-08-10 15:27:22.872525 [DEBUG] switch_ivr_play_say.c:1444 done playing >> file >> 2010-08-10 15:27:22.934526 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 1:752 >> 2010-08-10 15:27:22.934526 [ERR] mod_native_file.c:74 Error opening >> /usr/local/freeswitch/sounds/en/us/callie/.PCMA >> 2010-08-10 15:27:23.892521 [DEBUG] switch_ivr_play_say.c:1444 done playing >> file >> 2010-08-10 15:27:25.012528 [DEBUG] switch_ivr_play_say.c:1444 done playing >> file >> 2010-08-10 15:27:25.832521 [DEBUG] switch_ivr_play_say.c:1444 done playing >> file >> 2010-08-10 15:27:29.792528 [DEBUG] switch_ivr_play_say.c:1444 done playing >> file >> 2010-08-10 15:27:31.872524 [DEBUG] switch_ivr_play_say.c:1444 done playing >> file >> 2010-08-10 15:27:34.492524 [DEBUG] switch_ivr_play_say.c:1444 done playing >> file >> 2010-08-10 15:27:37.152526 [DEBUG] switch_ivr_play_say.c:1444 done playing >> file >> 2010-08-10 15:27:37.152526 [NOTICE] switch_core_session.c:1949 Execute >> execute_extension(LangSel XML NXIVR) >> EXECUTE sofia/external/38516659280 at 195.88.212.41execute_extension(LangSel XML NXIVR) >> 2010-08-10 15:27:37.152526 [INFO] mod_dialplan_xml.c:418 Processing >> 38516659280->LangSel in context NXIVR >> Dialplan: sofia/external/38516659280 at 195.88.212.41 parsing >> [NXIVR->LangSel] continue=false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> destination_number(LangSel) =~ /^LangSel$/ break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] >> ${myLANG}(1^?@? ) =~ /^1$/ break=on-true >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> destination_number(LangSel) =~ /^LangSel$/ break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] >> ${myLANG}(1^?@? ) =~ /^2$/ break=on-true >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> destination_number(LangSel) =~ /^LangSel$/ break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] >> ${myLANG}(1^?@? ) =~ /^3$/ break=on-true >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> destination_number(LangSel) =~ /^LangSel$/ break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] >> ${myLANG}(1^?@? ) =~ /^4$/ break=on-true >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> destination_number(LangSel) =~ /^LangSel$/ break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] >> ${myLANG}(1^?@? ) =~ /^5$/ break=on-true >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> destination_number(LangSel) =~ /^LangSel$/ break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] >> ${myLANG}(1^?@? ) =~ /^[06789]$/ break=on-true >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> destination_number(LangSel) =~ /^LangSel$/ break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (FAIL) [LangSel] >> ${myLANG}(1^?@? ) =~ /^$/ break=on-true >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> destination_number(LangSel) =~ /^LangSel$/ break=on-false >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Regex (PASS) [LangSel] >> ${myLANG}(1^?@? ) =~ /^.*$/ break=on-true >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action log(INFO >> ################# WRONG LANG SEL ################\n) >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action >> set(language=hr) >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action >> set(ERR_PR=${WRONG_LANG_PROMPT}) >> Dialplan: sofia/external/38516659280 at 195.88.212.41 Action >> execute_extension(LangRetries XML NXIVR) >> 2010-08-10 15:27:37.154487 [NOTICE] switch_core_session.c:1949 Execute >> log(INFO ################# WRONG LANG SEL ################\n) >> EXECUTE sofia/external/38516659280 at 195.88.212.41 log(INFO >> ################# WRONG LANG SEL ################\n) >> 2010-08-10 15:27:37.154487 [INFO] mod_dptools.c:946 ################# >> WRONG LANG SEL ################\n >> 2010-08-10 15:27:37.154487 [NOTICE] switch_core_session.c:1949 Execute >> set(language=hr) >> EXECUTE sofia/external/38516659280 at 195.88.212.41 set(language=hr) >> >> >> you can see myLANG variable holds "1^?@? " ... of course i dialed 1 as >> seen in the log (switch_rtp.c:2428 RTP RECV DTMF 1:752). >> >> >> now, i did some extra testing and found that it happens only when i use a >> playback_separator to append multiple files to be played.... >> >> i checked switch_ivr_read and switch_ivr_collect_digits_count functions >> but everything seems to be ok there... >> >> >> >> can anyone help locating the problem ? >> >> >> >> >> Ps: im running >> >> >> freeswitch at cxss01> version >> >> FreeSWITCH Version 1.0.6 (svn-exported) >> >> tried with Revision: 17032 before this one but seems to be the same.... >> >> >> >> [tculjaga at cxss01 src]$ cat /etc/issue >> CentOS release 5.4 (Final) >> Kernel \r on an \m >> >> [tculjaga at cxss01 src]$ >> >> >> >> [tculjaga at cxss01 src]$ uname -a >> Linux cxss01 2.6.18-164.el5 #1 SMP Thu Sep 3 03:28:30 EDT 2009 x86_64 >> x86_64 x86_64 GNU/Linux >> [tculjaga at cxss01 src]$ >> >> >> >> >> >> >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/65d2ac3b/attachment-0001.html From msc at freeswitch.org Tue Aug 10 15:11:12 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Aug 2010 15:11:12 -0700 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <563B3EF6EAFE41BC8506DDA53B45F7F8@left> References: <563B3EF6EAFE41BC8506DDA53B45F7F8@left> Message-ID: On Tue, Aug 10, 2010 at 11:19 AM, Chris Veazey wrote: > Interesting that it worked for you. Currently my setup will not send a new > invite to the maddr address in the 302 contact. > > Do you make any changes to get your setup to accept the 302 and send to the > maddr address? > Are you on the latest git? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/23337717/attachment.html From mranga at gmail.com Tue Aug 10 15:11:47 2010 From: mranga at gmail.com (M. Ranganathan) Date: Tue, 10 Aug 2010 18:11:47 -0400 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> <1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> <2F9511642CCD492CA057DB89F09BF420@MOBILEE1705> Message-ID: On Mon, Aug 9, 2010 at 4:43 PM, Michael Collins wrote: > > > On Mon, Aug 9, 2010 at 9:59 AM, Madovsky wrote: >> >> its' also an excellent way for big companies to avoid >> to pay employees to make all the dev work also.... > > Indeed it is. Most of us in the OSS world say, "So what?" We've given our > work away for "free" in return for other considerations: free advertising, > free distribution (via Internet downloads), bragging rights, and growing a > software-based ecosystem that allows us to tap into other revenue streams > like private consulting or even writing a book. If big companies "take" our > stuff and use it then they're growing the ecosystem. The choice of OSS > licenses available to us gives us the necessary protection from large > corporations hijacking our stuff. (This includes things like CC for > documents, photos, sounds/music, etc.) Many big companies ( IBM for example ) pay employees to contribute to open source. I am an employee of a company that has paid me to contribute to open source. I have also given away many of my contributions in the past and am very glad to report that they are alive and well. That is a terrific feeling. On the other hand, if only I had a penny for each closed source byte I have contributed to the big bit bucket in the sky.... Further, OSS is really the ONLY way for little guys to compete. Linux would have never become what it is if it were a bit company closed source project. -- M. Ranganathan From msc at freeswitch.org Tue Aug 10 15:16:42 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Aug 2010 15:16:42 -0700 Subject: [Freeswitch-users] First installation help In-Reply-To: <05cd01cb38ca$40603ff0$c120bfd0$@khirman.com> References: <05cd01cb38ca$40603ff0$c120bfd0$@khirman.com> Message-ID: Stas, I guess we didn't scare you away from FS when you were at ClueCon! :P I'm not sure about your first question, but your second question seems to indicate that you have an older dialplan. You can test it out by dialing "9996" instead of "9196" or "9999" instead of "9664". If the 99xx extensions work then you have an older dialplan, in which case you should save any modifications that you've made and then delete conf/dialplan/default.xml. Then run "make samples" in your fs src directory to get the most recent default.xml dialplan file. -MC On Tue, Aug 10, 2010 at 1:24 PM, Stas Khirman wrote: > [first time user beg for help]?.. > > > > Trying my first FreeSwitch installation, I faced two problems : > > > > 1.) My host has two IP addresses. SoftPhone (Ekiga) running on the same > host successfully registered with a first IP, but refuse to register with > the second. Also, registration to 127.0.0.1 failing? How can I instruct > FreeSWITCH to use BOTH addresses for registration? > > 2.) Somehow dialing test numbers (9196, 9199,etc) don?t work properly.. > FreeSWITCh reports: > > 2010-08-10 13:17:08.577884 [INFO] mod_dialplan_xml.c:418 Processing > 1000->9196 in context default > > 2010-08-10 13:17:08.619705 [NOTICE] switch_ivr.c:1447 Transfer > sofia/internal/1000 at 10.0.2.15 to enum[9196 at default] > > 2010-08-10 13:17:09.703867 [INFO] switch_core_state_machine.c:142 No Route, > Aborting > > 2010-08-10 13:17:09.703867 [NOTICE] switch_core_state_machine.c:143 Hangup > sofia/internal/1000 at 10.0.2.15 [CS_ROUTING] [NO_ROUTE_DESTINATION] > > > > > > > > Any help is deeply appreciated! > > > > Regards > > Stas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/4ec8d7f7/attachment.html From msc at freeswitch.org Tue Aug 10 15:19:24 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Aug 2010 15:19:24 -0700 Subject: [Freeswitch-users] q: originating a call via http In-Reply-To: References: Message-ID: On Tue, Aug 10, 2010 at 11:35 AM, Tihomir Culjaga wrote: > > > On Tue, Aug 10, 2010 at 7:15 PM, tom wrote: > >> hi is that possible to have just a reg html link to FS to place a call? at >> best, it would ring first my extension, and when i take the hearer then it >> should ring the destination. >> thx >> > > > use this to send commands to your FS via a web server: > http://wiki.freeswitch.org/wiki/Mod_xml_rpc ... after that its just a > dialplan thing. > > cheers! > Don't forget this page: http://wiki.freeswitch.org/wiki/Webapi I wrote this one before I grokked that mod_xml_rpc and the webapi stuff were all part of the same module. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/12cc5113/attachment.html From craigesmith at gmail.com Tue Aug 10 15:34:06 2010 From: craigesmith at gmail.com (Craig Smith) Date: Tue, 10 Aug 2010 18:34:06 -0400 Subject: [Freeswitch-users] Small office configuration Message-ID: Does anyone have a complete small office set of configs that they would be willing to share? From brian at freeswitch.org Tue Aug 10 15:44:34 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Aug 2010 17:44:34 -0500 Subject: [Freeswitch-users] Small office configuration In-Reply-To: References: Message-ID: <5876167D-BBE6-443E-B7FB-193853D6BA06@freeswitch.org> The default config makes a great small office config set...what exactly are you looking for feature wise? /b On Aug 10, 2010, at 5:34 PM, Craig Smith wrote: > Does anyone have a complete small office set of configs that they > would be willing to share? From msc at freeswitch.org Tue Aug 10 15:47:21 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Aug 2010 15:47:21 -0700 Subject: [Freeswitch-users] Small office configuration In-Reply-To: References: Message-ID: On Tue, Aug 10, 2010 at 3:34 PM, Craig Smith wrote: > Does anyone have a complete small office set of configs that they > would be willing to share? > > Hmm... The default configs are very "small office friendly" already. Are there some features that you need that aren't already available in the default configs? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/adeaa092/attachment.html From 12ukwn at gmail.com Tue Aug 10 15:50:10 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 11 Aug 2010 00:50:10 +0200 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> <1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> <2F9511642CCD492CA057DB89F09BF420@MOBILEE1705> Message-ID: <20100811005010.47fc80d2@anubis.defcon1> Le Tue, 10 Aug 2010 18:11:47 -0400, "M. Ranganathan" a ?crit : ... > Many big companies ( IBM for example ) pay employees to contribute to > open source. I am an employee of a company that has paid me to > contribute to open source. I have also given away many of my > contributions in the past and am very glad to report that they are > alive and well. That is a terrific feeling. On the other hand, if only > I had a penny for each closed source byte I have contributed to the > big bit bucket in the sky.... > > Further, OSS is really the ONLY way for little guys to compete. Linux > would have never become what it is if it were a bit company closed > source project. Not only! A small anecdote: in france, there are about 400 theaters of "arts & essais" (mean they project unknown and low budgets films usually from not so well known directors that would never be in the regular circuit), they all use the same software which was first a closed source. The software company bankrupted and the former CEO refused to release the source until a wise judge ordered him to do so. This withholding put theaters in troubles because of law changes that add some mandatory printing on the tickets and because of some weird bugs too. The company that was asked to continue the work took serious decisions: a total rewrite into java for people that wanted to use another OS than w$ (these theaters have very little money, so a w$ license is a hi spending for them) and they put the software into open-source. At this time, some theaters' guys with programming skills are cooperating with the software company to improve the software, theaters are now sure whatever will happen to the company they won't ever be blocked and the quality of the software is now so high that you can leave the control computer on 24/7. So, advantages aren't reserved to the programmer, they're also shared by the client. JY -- If men could get pregnant, abortion would be a sacrament. From stas at khirman.com Tue Aug 10 15:50:37 2010 From: stas at khirman.com (Stas Khirman) Date: Tue, 10 Aug 2010 15:50:37 -0700 Subject: [Freeswitch-users] First installation help In-Reply-To: References: <05cd01cb38ca$40603ff0$c120bfd0$@khirman.com> Message-ID: <061501cb38de$74b81f70$5e285e50$@khirman.com> Michael, Yes, you absolutely right - 999x are working. BTW, I took my code from http://files.freeswitch.org/freeswitch-1.0.6.tar.gz - i was sure that it is a latest version.Maybe it make sense to repackage it to remove confusion. On the first question - it seems that in multi interface configuration FreeSWITCH is explicitly bind() to the first interface only . Is any configuration parameters to specify what interfaces to bind / Regards Stas From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, August 10, 2010 3:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] First installation help Stas, I guess we didn't scare you away from FS when you were at ClueCon! :P I'm not sure about your first question, but your second question seems to indicate that you have an older dialplan. You can test it out by dialing "9996" instead of "9196" or "9999" instead of "9664". If the 99xx extensions work then you have an older dialplan, in which case you should save any modifications that you've made and then delete conf/dialplan/default.xml. Then run "make samples" in your fs src directory to get the most recent default.xml dialplan file. -MC On Tue, Aug 10, 2010 at 1:24 PM, Stas Khirman wrote: [first time user beg for help]... Trying my first FreeSwitch installation, I faced two problems : 1.) My host has two IP addresses. SoftPhone (Ekiga) running on the same host successfully registered with a first IP, but refuse to register with the second. Also, registration to 127.0.0.1 failing. How can I instruct FreeSWITCH to use BOTH addresses for registration. 2.) Somehow dialing test numbers (9196, 9199,etc) don't work properly.. FreeSWITCh reports: 2010-08-10 13:17:08.577884 [INFO] mod_dialplan_xml.c:418 Processing 1000->9196 in context default 2010-08-10 13:17:08.619705 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000 at 10.0.2.15 to enum[9196 at default] 2010-08-10 13:17:09.703867 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2010-08-10 13:17:09.703867 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000 at 10.0.2.15 [CS_ROUTING] [NO_ROUTE_DESTINATION] Any help is deeply appreciated! Regards Stas _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/4ad273bb/attachment-0001.html From brian at freeswitch.org Tue Aug 10 15:52:43 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Aug 2010 17:52:43 -0500 Subject: [Freeswitch-users] Small office configuration In-Reply-To: References: Message-ID: if thats the case we might want to consider some modifications to the defaults ;) /b On Aug 10, 2010, at 5:47 PM, Michael Collins wrote: > Hmm... The default configs are very "small office friendly" already. Are there some features that you need that aren't already available in the default configs? > > -MC > From msc at freeswitch.org Tue Aug 10 16:09:30 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Aug 2010 16:09:30 -0700 Subject: [Freeswitch-users] First installation help In-Reply-To: <061501cb38de$74b81f70$5e285e50$@khirman.com> References: <05cd01cb38ca$40603ff0$c120bfd0$@khirman.com> <061501cb38de$74b81f70$5e285e50$@khirman.com> Message-ID: On Tue, Aug 10, 2010 at 3:50 PM, Stas Khirman wrote: > Michael, > > > > Yes, you absolutely right ? 999x are working? BTW, I took my code from > http://files.freeswitch.org/freeswitch-1.0.6.tar.gz - i was sure that it > is a latest version?Maybe it make sense to repackage it to remove confusion? > > > 1.0.6 is the latest tarball but you can use the git repo to get the latest and greatest code which is recommended for initial installs. http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code > On the first question ? it seems that in multi interface configuration > FreeSWITCH is explicitly bind() to the first interface only . Is any > configuration parameters to specify what interfaces to bind / > Look for conf/sip_profiles/internal.xml and external.xml These files control your SIP profiles, which are just SIP UAs Look for these lines: You can specify an IP address there. Each SIP profile can bind to a particular IP address and port. The port param is like this: Also, look in conf/vars.xml to see where many of these variables are defined. Last item: go buy our FreeSWITCH book! We've fallen to #3 on Packt's top books list so we need a boost! :P Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/cf70a8fe/attachment.html From msc at freeswitch.org Tue Aug 10 16:25:13 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Aug 2010 16:25:13 -0700 Subject: [Freeswitch-users] Features question In-Reply-To: References: Message-ID: On Tue, Aug 10, 2010 at 2:32 PM, Craig Smith wrote: > AASTRA > Download user guides at http://www.aastra.com > > Hold: press HOLD button > > Retrieve Hold: press blinking line key > > Transfer (attended): press XFER, dial destination, press DIAL, wait > for answer, announce call, hangup to complete transfer > > Transfer (blind): Press XFER, dial destination, press DIAL, hangup > > *Retrieve VM: See message wait light; dial *98 > *98 is already in the default dialplan. It is just like dialing 4000 or dialing one's own extension > > *Transfer caller directly to VM: Press XFER, dial **, dial extension > number, hangup > **XXXX currently is used for intercept (i.e. call pickup) However, I threw this extension together quickly for a proof of concept: Use ## + extension number to dial straight to VM > > *Call directly to user`s VM to leave message: Dial **, dial extension > number, leave message > Same as above > > *Intercom call: Dial **, dial extension number, dial *1 (only works > with phones that support hands-free auto-answer) > This is totally doable but I don't see any examples on the wiki. (However, I know it's doable because we do it in the CudaTel.) I will research and let you know what I find. -MC P.S. - The best way to be a FreeSWITCH convert is to buy our new book! :P -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/89e0d692/attachment.html From tculjaga at gmail.com Tue Aug 10 16:40:41 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 11 Aug 2010 01:40:41 +0200 Subject: [Freeswitch-users] q: originating a call via http In-Reply-To: References: Message-ID: On Wed, Aug 11, 2010 at 12:19 AM, Michael Collins wrote: > > > On Tue, Aug 10, 2010 at 11:35 AM, Tihomir Culjaga wrote: > >> >> >> On Tue, Aug 10, 2010 at 7:15 PM, tom wrote: >> >>> hi is that possible to have just a reg html link to FS to place a call? >>> at best, it would ring first my extension, and when i take the hearer then >>> it should ring the destination. >>> thx >>> >> >> >> use this to send commands to your FS via a web server: >> http://wiki.freeswitch.org/wiki/Mod_xml_rpc ... after that its just a >> dialplan thing. >> >> cheers! >> > > Don't forget this page: > http://wiki.freeswitch.org/wiki/Webapi > > I wrote this one before I grokked that mod_xml_rpc and the webapi stuff > were all part of the same module. :) > -MC > > this is even better :) T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/77b6b5f0/attachment.html From chris.veazey at gmail.com Tue Aug 10 20:42:53 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Tue, 10 Aug 2010 22:42:53 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: References: <563B3EF6EAFE41BC8506DDA53B45F7F8@left> Message-ID: <44EF839FFFD0456EA2A1DEA11BC11BDC@left> Updated to latest git. Now its sending out a 410 Gone and kicking this out in the logs 2010-08-11 03:39:31.911514 [DEBUG] mod_sofia.c:453 Channel sofia/external/9995552000000 hanging up, cause: REDIRECTION_TO_NEW_DESTINATION _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, August 10, 2010 5:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS processing 302 On Tue, Aug 10, 2010 at 11:19 AM, Chris Veazey wrote: Interesting that it worked for you. Currently my setup will not send a new invite to the maddr address in the 302 contact. Do you make any changes to get your setup to accept the 302 and send to the maddr address? Are you on the latest git? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100810/22bb04aa/attachment.html From lloyd.aloysius at gmail.com Tue Aug 10 21:05:59 2010 From: lloyd.aloysius at gmail.com (Aloysius Lloyd) Date: Wed, 11 Aug 2010 00:05:59 -0400 Subject: [Freeswitch-users] Small office configuration In-Reply-To: References: Message-ID: Hi Brian/Michael, >From my experience the following features need for the small business pbx 1. Call Forward All Activate ( *72) Please enter the extension number then ask the forwarding number. 2. Call Forward All Deactivate ( *73) Deactivate the above 3. Follow-Me ( Find-ME) 4. Block Caller ID ( *67+ number) 5. Day/Night Service Toggle. Say Daytime human answer Night time IVR. Using a feature code to toggle this feature. 6. Voice Mail Transfer Thanks Lloyd On Tue, Aug 10, 2010 at 6:52 PM, Brian West wrote: > if thats the case we might want to consider some modifications to the > defaults ;) > > /b > > On Aug 10, 2010, at 5:47 PM, Michael Collins wrote: > > > Hmm... The default configs are very "small office friendly" already. Are > there some features that you need that aren't already available in the > default configs? > > > > -MC > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/14370d39/attachment-0001.html From 12ukwn at gmail.com Tue Aug 10 23:18:33 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 11 Aug 2010 08:18:33 +0200 Subject: [Freeswitch-users] Small office configuration In-Reply-To: References: Message-ID: <20100811081833.51224ed8@anubis.defcon1> Le Tue, 10 Aug 2010 17:52:43 -0500, Brian West a ?crit : > if thats the case we might want to consider some modifications to the > defaults ;) I don't know if it is possible but the possibility, by group or extension, to have a caller beeing redirected to MOH if callee's line busy AND automatically reconnected to this same group/extension when line becomes ready would be nice (in an orderly fashion.) I used that in a former live with asterisk, and had a regular announcement (every 30 sec) in MOH to tell people on queue that they were able to leave a message by hitting the star key, which redirected to my VM. JY -- They can always run stderr through uniq. :-) -- Larry Wall in <199704012331.PAA16535 at wall.org> From ben at langfeld.co.uk Wed Aug 11 02:09:17 2010 From: ben at langfeld.co.uk (Ben Langfeld) Date: Wed, 11 Aug 2010 10:09:17 +0100 Subject: [Freeswitch-users] Small office configuration In-Reply-To: <20100811081833.51224ed8@anubis.defcon1> References: <20100811081833.51224ed8@anubis.defcon1> Message-ID: mod_fifo is what you need here. Regards, Ben Langfeld On Wed, Aug 11, 2010 at 7:18 AM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote: > Le Tue, 10 Aug 2010 17:52:43 -0500, > Brian West a ?crit : > > > if thats the case we might want to consider some modifications to the > > defaults ;) > > I don't know if it is possible but the possibility, by group or > extension, to have a caller beeing redirected to MOH if callee's line > busy AND automatically reconnected to this same group/extension when line > becomes ready would be nice (in an orderly fashion.) > > I used that in a former live with asterisk, and had a regular announcement > (every 30 sec) in MOH to tell people on queue that they were able to leave > a message by hitting the star key, which redirected to my VM. > > JY > -- > They can always run stderr through uniq. :-) > -- Larry Wall in <199704012331.PAA16535 at wall.org> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/50d84aba/attachment.html From xanlich at gmail.com Wed Aug 11 02:11:00 2010 From: xanlich at gmail.com (Chia-Yen Wu) Date: Wed, 11 Aug 2010 17:11:00 +0800 Subject: [Freeswitch-users] Sending Fax problem Message-ID: hello, i am trying to send fax via PSTN by VOIP gateway original way: originate sofia/gateway//12345678 &txfax(/path_to_fax_file) but i must set up my gateway as "Call a user as a gateway" i set VOIP gateway to user 9999, and i want to fax a document to 12345678 (pstn number) but how can i input the fax number? i cant find a place to input pstn number, thank you like: originate user/9999 &txfax(/path_to_fax_file) -- Best regards, Ted -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/908d6869/attachment.html From 12ukwn at gmail.com Wed Aug 11 04:04:08 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 11 Aug 2010 13:04:08 +0200 Subject: [Freeswitch-users] Small office configuration In-Reply-To: References: <20100811081833.51224ed8@anubis.defcon1> Message-ID: <20100811130408.5dd7c3d2@anubis.defcon1> Le Wed, 11 Aug 2010 10:09:17 +0100, Ben Langfeld a ?crit : > mod_fifo is what you need here. > > Regards, > Ben Langfeld I wasn't sure, thanks Ben. -- Things past redress and now with me past care. -- William Shakespeare, "Richard II" From chaitanya at vivainfomedia.com Wed Aug 11 04:25:50 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Wed, 11 Aug 2010 16:55:50 +0530 Subject: [Freeswitch-users] bgapi playback not working Message-ID: Hey I am using Freeswitch ESL librarary & commands with help of IVR.pm(provided in perl esl module). When i trying "execute('playback','file.wav')" i am getting call properly but i want to play this file in background, so i am trying to use bgapi('playback','file.wav') but i am not getting call properly. I tried "api('playback','file.wav')" as well, but this also not working. Can you please guide me how can i resolve this ? Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/53a9158f/attachment.html From ken at ukgb.net Wed Aug 11 04:31:10 2010 From: ken at ukgb.net (Ken Gillett) Date: Wed, 11 Aug 2010 12:31:10 +0100 Subject: [Freeswitch-users] Modems In-Reply-To: References: <81802D63-8635-465F-B478-34C846F4E61F@ukgb.net> <197186CA-C15C-4F73-82A0-FB219FF9E2EA@ukgb.net> Message-ID: <99790B7E-FF97-40AC-AFC4-EE736D319331@ukgb.net> That is very interesting and I will almost certainly be trying it out with an SPA3102, but I still don't have a complete grasp of VOIP -> PSTN bridging, whatever the devices. A SIP client normally makes a call by contacting a SIP server somewhere and telling it the number to be called. But how can it do this when it needs to be done through a gateway onto PSTN? The gateway is not a SIP server of any sort, just another client, so how can the calling SIP client/device contact the gateway and say "call this number"? It is this basic process flow that I want to get my head around. Once I have that basic understanding I am in a better position to try and configure the various devices. Without that, I might as well be pushing buttons at random. So I'd appreciate it if someone could please explain this process and enlighten me. On 10 Aug 2010, at 13:09, Rupa Schomaker wrote: > http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo > >> On Tue, Aug 10, 2010 at 2:17 AM, Ken Gillett wrote: >> Well I did try a Linksys/Cisco/Sipura 3102 but it's a configuration nightmare. The problem mainly is as I said, I cannot figure how VOIP -> PSTN bridging can be achieved whatever the gateway device. >> >> A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the call. From that piece of data it gets the SIP domain and the ID of the user. So how is the gateway able to be inserted in this process. The SIP client has no knowledge of this device and no way to include that third piece of data into its process. >> >> Obviously such gateways do exist, but I do not yet understand how the process works. Some advice on the basic process flow would therefore assist me to set up what I need irrespective of what devices I am using. Sorry to be so ignorant about this, but anyone able and willing to help with an explanation? >> >> >> On 10 Aug 2010, at 04:23, Rupa Schomaker wrote: >> >> > I'm pretty sure the zoom does not support sip originated calls to the FXO port. It's FXO port is strictly used as failover or selectable via dialplan when the call originates from the FXS port (eg: dial 9 first to get FXO). >> > >> > Try: >> > >> > cisco 3102 >> > audiocodes >> > grandstream >> > >> > for atas that support full FXS/FXO operation. >> > >> > On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett wrote: >> > I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I've even thought about bringing FreeSwitch into the equation. > Ken G i l l e t t _/_/_/_/_/_/_/_/ From peter.olsson at visionutveckling.se Wed Aug 11 05:23:02 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 11 Aug 2010 14:23:02 +0200 Subject: [Freeswitch-users] bgapi playback not working In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC078E03@cooper> You should use execute(), but make sure that the socket is in async mode. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chaitanya Bhatt // Viva Skickat: den 11 augusti 2010 13:26 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] bgapi playback not working Hey I am using Freeswitch ESL librarary & commands with help of IVR.pm(provided in perl esl module). When i trying "execute('playback','file.wav')" i am getting call properly but i want to play this file in background, so i am trying to use bgapi('playback','file.wav') but i am not getting call properly. I tried "api('playback','file.wav')" as well, but this also not working. Can you please guide me how can i resolve this ? Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 !DSPAM:4c628ac532931275669086! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/b8e0fbb9/attachment-0001.html From abu.4000 at gmail.com Wed Aug 11 05:24:59 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Wed, 11 Aug 2010 17:54:59 +0530 Subject: [Freeswitch-users] bgapi playback not working In-Reply-To: References: Message-ID: playback is an application , you cannot execute it using api or bgapi , U can try uuid_broadcast. -USAGE: [aleg|bleg|both] On Wed, Aug 11, 2010 at 4:55 PM, Chaitanya Bhatt // Viva < chaitanya at vivainfomedia.com> wrote: > Hey > > I am using Freeswitch ESL librarary & commands with help of IVR.pm(provided > in perl esl module). > When i trying "execute('playback','file.wav')" i am getting call properly > but i want to play this file in background, so i am trying to use > bgapi('playback','file.wav') but i am not getting call properly. > I tried "api('playback','file.wav')" as well, but this also not working. > Can you please guide me how can i resolve this ? > > Incase of any further queries, Please feel free to mail me or contact me on > the numbers provided below. > > Thanks & Regards, > Chaitanya Bhatt > Software Engineer. > > Viva Infomedia Pvt. Ltd. > 242, Oshiwara Industrial Centre, > New Link Road, Opp. Oshiwara Bus Depot, > Goregaon West, Mumbai 400104. > > Direct: +91.22.40310356 > Board: +91.22.40310310 > Email : chaitanya at vivainfomedia.com > > Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India > Awards 2009 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Best Regards, **Abubacker systems engineer bk systems (p) ltd** * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/10acf3f1/attachment.html From stephen at stephenjc.com Wed Aug 11 06:12:12 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Wed, 11 Aug 2010 09:12:12 -0400 Subject: [Freeswitch-users] mod_voipcodecs build time In-Reply-To: <33AD71CF-04F0-4889-B499-DC79A2C66053@freeswitch.org> References: <33AD71CF-04F0-4889-B499-DC79A2C66053@freeswitch.org> Message-ID: thanks; i forgot about that and that mod_limit was replaced when i copied my configs over. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print On Tue, Aug 10, 2010 at 2:13 PM, Brian West wrote: > Because you don't need tha tmodule anymore you need mod_spandsp. > > /b > > On Aug 10, 2010, at 1:05 PM, Stephen Cattaneo wrote: > > > When ever i build freeswitch on centos 5.5 i have to manually compile and > move mod_voipcodecs. The make file exists in the directory it just never > gets compiled and installed. > > > > > > Thanks, > > Stephen C > > -All of my email addresses go to the same place > > -Save Paper, think before you print > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/e180e43b/attachment.html From chris.veazey at gmail.com Wed Aug 11 07:00:42 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Wed, 11 Aug 2010 09:00:42 -0500 Subject: [Freeswitch-users] FS processing 302 References: <563B3EF6EAFE41BC8506DDA53B45F7F8@left> Message-ID: <396D3026B6034431B93A6B3DC0803461@left> 2010-08-11 13:58:35.195809 [DEBUG] switch_channel.c:2309 (sofia/external/9995551000001) Callstate Change RINGING -> HANGUP 2010-08-11 13:58:35.195809 [NOTICE] sofia.c:3998 Hangup sofia/external/9995551000001 [CS_CONSUME_MEDIA] [REDIRECTION_TO_NEW_DESTINATION] 2010-08-11 13:58:35.196806 [DEBUG] switch_channel.c:2325 Send signal sofia/external/9995551000001 [KILL] 2010-08-11 13:58:35.196806 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/9995551000001 [BREAK] 2010-08-11 13:58:35.196806 [DEBUG] switch_core_state_machine.c:314 (sofia/external/9995551000001) Running State Change CS_HANGUP 2010-08-11 13:58:35.196806 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 23 [REDIRECTION_TO_NEW_DESTINATION] 2010-08-11 13:58:35.196806 [INFO] mod_dptools.c:2393 Originate Failed. Cause: REDIRECTION_TO_NEW_DESTINATION _____ From: Chris Veazey [mailto:chris.veazey at gmail.com] Sent: Tuesday, August 10, 2010 10:43 PM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] FS processing 302 Updated to latest git. Now its sending out a 410 Gone and kicking this out in the logs 2010-08-11 03:39:31.911514 [DEBUG] mod_sofia.c:453 Channel sofia/external/9995552000000 hanging up, cause: REDIRECTION_TO_NEW_DESTINATION _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, August 10, 2010 5:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS processing 302 On Tue, Aug 10, 2010 at 11:19 AM, Chris Veazey wrote: Interesting that it worked for you. Currently my setup will not send a new invite to the maddr address in the 302 contact. Do you make any changes to get your setup to accept the 302 and send to the maddr address? Are you on the latest git? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/4ebdfbc7/attachment.html From esamuels777 at gmail.com Tue Aug 10 19:41:51 2010 From: esamuels777 at gmail.com (Errol Samuels) Date: Wed, 11 Aug 2010 03:41:51 +0100 Subject: [Freeswitch-users] Nextone one-way Audio issue Message-ID: Dear All, One of my carrier partners uses Nextone SBC and we can send traffic to them from Freeswitch without any issues. However, when they send traffic to our Freeswitch box they are getting one-way audio. The Nextone SBC and our Freeswitch are both on Public IPs. FreeSWITCH Version: 1.0.head (git-910729b 2010-07-29 12-09-49 +0200) I have to wait until tomorrow for the carrier to re-test and I will get a SIP Trace then. So for the moment does anyone have any ideas? Regards, Errol -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/f44d6076/attachment.html From fdelawarde at wirelessmundi.com Wed Aug 11 02:05:42 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 11 Aug 2010 11:05:42 +0200 Subject: [Freeswitch-users] Freeswitch and OCS Message-ID: <1281517542.28815.193.camel@luna.tc.commsmundi.com> Hello, Anyone has experience or feedback about the Office Communications Server product from MS and its compatibility with Freeswitch (presence, chat, voice/video, ...). Should I avoid this product? Thanks, Fran?ois. From shamun.toha at gmail.com Wed Aug 11 06:46:08 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Wed, 11 Aug 2010 15:46:08 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype conference Message-ID: Hi, I have a call like following: Which B have to make. 1. Reference: ========== A0 = first caller B = Operator C = Support Engineer A1, A2, A3 is invited in the conference call for that session, which is made by B, only B gets out when all of them are talking 2. Draw it: ======== A0->B->C | / +----------/ +A1 +A2 +A3 3. Result: ======= A0 + A1 + A2 + A3 getting Support by C as a conference call. B made this connections and gets out when all are connected. Is that possible using FreeSwitch? Thank you Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/e5a05a3f/attachment-0001.html From freeswitch at aastral.net Wed Aug 11 08:19:05 2010 From: freeswitch at aastral.net (Bill W) Date: Wed, 11 Aug 2010 11:19:05 -0400 Subject: [Freeswitch-users] A/B Leg nibble billing and CDR Message-ID: <4C62BF69.8030207@aastral.net> Hi Everybody! In some (but not all) cases, I need to nibble bill on both the A and the B legs, and I need to log those CDR records appropriately. So my questions are: 1. Is it possible to nibble on A and B legs of the same call, to different nibble accounts and at different rates? Could I simply do: (assuming I have LCR export a different nibble_rate to the B-leg) 2. If I do this, will the appropriate nibble_rate and nibble_total_billed be set correctly in the respective A and B legs, assuming I'm logging both a and b legs in mod_cdr_csv? 3. Is there some channel variable I can log that shows which A-leg-cdr-record a B-leg record belongs to? Thanks! Bill From peter.olsson at visionutveckling.se Wed Aug 11 08:20:08 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 11 Aug 2010 17:20:08 +0200 Subject: [Freeswitch-users] Freeswitch and OCS In-Reply-To: <1281517542.28815.193.camel@luna.tc.commsmundi.com> References: <1281517542.28815.193.camel@luna.tc.commsmundi.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC078EA3@cooper> We use FreeSWITCH as a trunk to PSTN (SIP over TCP), it works quite nice - but I have no other expeciance than this... /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Fran?ois Delawarde Skickat: den 11 augusti 2010 11:06 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Freeswitch and OCS Hello, Anyone has experience or feedback about the Office Communications Server product from MS and its compatibility with Freeswitch (presence, chat, voice/video, ...). Should I avoid this product? Thanks, Fran?ois. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c62bdb832931062522955! From 12ukwn at gmail.com Wed Aug 11 08:21:40 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 11 Aug 2010 17:21:40 +0200 Subject: [Freeswitch-users] Nextone one-way Audio issue In-Reply-To: References: Message-ID: <20100811172140.602cdd0d@anubis.defcon1> Le Wed, 11 Aug 2010 03:41:51 +0100, Errol Samuels a ?crit : ... > them from Freeswitch without any issues. However, when they send traffic > to our Freeswitch box they are getting one-way audio. The Nextone SBC > and our Freeswitch are both on Public IPs. ... Do you authorize any incoming unprivileged port connection from your providers IP(s)? (looks like an 'established,related' without a connect) -- Q: What's the difference between a sorority girl and a fast car? A: Not everyone's been in a fast car. From jeff at jefflenk.com Wed Aug 11 08:59:58 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 11 Aug 2010 08:59:58 -0700 (PDT) Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <396D3026B6034431B93A6B3DC0803461@left> References: <4508448D788749F6B25886D626C17C28@left> <563B3EF6EAFE41BC8506DDA53B45F7F8@left> <396D3026B6034431B93A6B3DC0803461@left> Message-ID: <1281542398575-5412733.post@n2.nabble.com> This is caused by commit dfa5439937c7197e2bc108ed14b87d5924f76710 * fix potential excess cpu usage during originate Comment out that line to correct 302 redirect I am still trying to determine what is the correct fix here but this will temporarily fix the problem -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p5412733.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gmaruzz at celliax.org Wed Aug 11 09:13:01 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 11 Aug 2010 18:13:01 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype conference In-Reply-To: References: Message-ID: On Wed, Aug 11, 2010 at 3:46 PM, Shamun toha md wrote: > Is that possible using FreeSwitch? Yes Have a look on the wiki on how to operate conferences. -giovanni > > > Thank you > Best regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From sameer2k3t at gmail.com Wed Aug 11 09:30:42 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Wed, 11 Aug 2010 21:30:42 +0500 Subject: [Freeswitch-users] Hi All.. dingaling help Message-ID: Hello everyone I configured dingaling two days ago i am using it in client mode with gmail id. when i configured it, it was sending invites to users that it couldn't find in its address list. but when i changed the login id it is failing to do that... any suggestion -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/9577044f/attachment.html From craigesmith at gmail.com Wed Aug 11 09:34:03 2010 From: craigesmith at gmail.com (Craig Smith) Date: Wed, 11 Aug 2010 12:34:03 -0400 Subject: [Freeswitch-users] Small office configuration Message-ID: Thanks for responding. I think what I am looking for is the FreeSWITCH equivalents to the Asterisk features below: Transfer: blind or consult Call forwarding Pickup groups: local or explicit Intercom Paging Call parking Thanks again, Craig -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/4b8b1c71/attachment.html From msc at freeswitch.org Wed Aug 11 09:39:19 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Aug 2010 09:39:19 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly! Message-ID: Hello all, Please be sure to join the conf call. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_08_11 It is light today but that will give us time to recap some of the fun stuff that happened at ClueCon. :P Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/4abb59ad/attachment.html From chris.veazey at gmail.com Wed Aug 11 09:42:17 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Wed, 11 Aug 2010 11:42:17 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <1281542398575-5412733.post@n2.nabble.com> References: <4508448D788749F6B25886D626C17C28@left><563B3EF6EAFE41BC8506DDA53B45F7F8@left><396D3026B6034431B93A6B3DC0803461@left> <1281542398575-5412733.post@n2.nabble.com> Message-ID: <55D75CA26DF8494B9FD88112BEAE8211@left> Specifically which line are you saying needs to be commented? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Wednesday, August 11, 2010 11:00 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS processing 302 This is caused by commit dfa5439937c7197e2bc108ed14b87d5924f76710 * fix potential excess cpu usage during originate Comment out that line to correct 302 redirect I am still trying to determine what is the correct fix here but this will temporarily fix the problem -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p54 12733.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sos at sokhapkin.dyndns.org Wed Aug 11 09:45:58 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 11 Aug 2010 12:45:58 -0400 Subject: [Freeswitch-users] Small office configuration In-Reply-To: References: Message-ID: <201008111245.58798.sos@sokhapkin.dyndns.org> Most of features you mention are not stock asterisk features, but addons provided by freepbx. On Wednesday 11 August 2010, Craig Smith wrote: > Thanks for responding. > > I think what I am looking for is the FreeSWITCH equivalents to the Asterisk > features below: > > Transfer: blind or consult > Call forwarding > Pickup groups: local or explicit > Intercom > Paging > Call parking > > Thanks again, > > Craig > From brian at freeswitch.org Wed Aug 11 09:50:38 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Aug 2010 11:50:38 -0500 Subject: [Freeswitch-users] Small office configuration In-Reply-To: <201008111245.58798.sos@sokhapkin.dyndns.org> References: <201008111245.58798.sos@sokhapkin.dyndns.org> Message-ID: Or just config options... most of what you want isn't even the job of FreeSWITCH... like blind and consult transfers... thats your phone... Intercom and paging are features of your phone... Parking we have ... Pickup is already in the defaults check intercept... call forwarding again a feature of your phone. /b On Aug 11, 2010, at 11:45 AM, Sergey Okhapkin wrote: > Most of features you mention are not stock asterisk features, but addons > provided by freepbx. > > On Wednesday 11 August 2010, Craig Smith wrote: >> Thanks for responding. >> >> I think what I am looking for is the FreeSWITCH equivalents to the Asterisk >> features below: >> >> Transfer: blind or consult >> Call forwarding >> Pickup groups: local or explicit >> Intercom >> Paging >> Call parking >> >> Thanks again, >> >> Craig > From msc at freeswitch.org Wed Aug 11 09:54:19 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Aug 2010 09:54:19 -0700 Subject: [Freeswitch-users] Small office configuration In-Reply-To: References: Message-ID: On Wed, Aug 11, 2010 at 9:34 AM, Craig Smith wrote: > Thanks for responding. > > I think what I am looking for is the FreeSWITCH equivalents to the Asterisk > features below: > > Transfer: blind or consult > If you have Polycoms then this is easy. If not, then get Polycoms. :P > Call forwarding > Right now we don't have a call-forwarding feature built into the default dialplan. Most phones already support this feature so if we added it then really it would be just for kicks. > Pickup groups: local or explicit > This currently exists in the default dialplan. See this page for more info: http://wiki.freeswitch.org/wiki/Default_Dialplan_QRF > Intercom See "extension-intercom" in the QRF > Paging > We haven't added this to the default dp yet > > Call parking > This is available also in the default dialplan. You can use FIFO parking or valet parking. See the above QRF link -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/0dd143c9/attachment-0001.html From msc at freeswitch.org Wed Aug 11 09:57:00 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Aug 2010 09:57:00 -0700 Subject: [Freeswitch-users] Sending Fax problem In-Reply-To: References: Message-ID: On Wed, Aug 11, 2010 at 2:11 AM, Chia-Yen Wu wrote: > hello, > > i am trying to send fax via PSTN by VOIP gateway > > original way: > originate sofia/gateway//12345678 &txfax(/path_to_fax_file) > > but i must set up my gateway as "Call a user as a gateway" > Why? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/d4e10e63/attachment.html From ben at langfeld.co.uk Wed Aug 11 10:37:05 2010 From: ben at langfeld.co.uk (Ben Langfeld) Date: Wed, 11 Aug 2010 18:37:05 +0100 Subject: [Freeswitch-users] FreeSWITCH Book Has Arrived! In-Reply-To: References: <4C514B79.9050003@xpirio.com> <4C5420A1.80207@gmx.net> Message-ID: Does anyone know when the book will be available in ePub? It's difficult to read on an iPhone without the ability to increase the font size. Regards, Ben Langfeld On Sat, Jul 31, 2010 at 3:59 PM, Nyamul Hassan wrote: > On Sat, Jul 31, 2010 at 19:09, Peter P GMX wrote: > >> Can we buy the Book at the Cluecon also (besides the chance to win a >> giveaway)? >> I would definitely buy one. >> >> Best regards >> Peter >> >> Tony Graziano schrieb: >> > Er, I wanted the ebook and I kept trying to make me pay in Euro's, >> > until after the price went up, then it decided I could pay in USD. Go >> > figure. >> > >> > >> > >> > On Fri, Jul 30, 2010 at 4:06 PM, eman > > > wrote: >> > >> > Yeah save $9 for pre-ordering or save $30 after it is released. I >> > guess they didn't sell enough copies after 2 days. >> > >> > >> > On Fri, Jul 30, 2010 at 3:32 PM, Tom Carlson > > > wrote: >> > >> > I am, however, disappointed that the ebook is PDF. I wish it >> > was epub. I tried to use a converter program to convert it to >> > epub, but I think the XML examples in the book were not >> > something the converter program could handle. >> > >> > >> > On Fri, Jul 30, 2010 at 12:28 PM, Tom Carlson >> > > wrote: >> > >> > I ordered the set, (paper + ebook). I expected to pay $65 >> > and change, I think they said the combo was, but when I >> > went to check out, the price was only $40 and change. >> > They had only charged me $4.80 for the ebook. The catch >> > is, you have to buy the paper book too. >> > >> > I just went back to the site, and now, when you click on >> > "Book and Ebook", they tell you that it's $40.79. >> > >> > >> > On Fri, Jul 30, 2010 at 8:32 AM, William Suffill >> > > > > wrote: >> > >> > UPS just dropped off my dead tree copy. >> > >> > -- W >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > ====================== >> > Tony Graziano, Manager >> > Telephone: 434.984.8430 >> > sip: tgraziano at voice.myitdepartment.net >> > >> > Fax: 434.984.8431 >> > >> > Email: tgraziano at myitdepartment.net > tgraziano at myitdepartment.net> >> > >> > LAN/Telephony/Security and Control Systems Helpdesk: >> > Telephone: 434.984.8426 >> > sip: helpdesk at voice.myitdepartment.net >> > >> > Fax: 434.984.8427 >> > >> > Helpdesk Contract Customers: >> > http://www.myitdepartment.net/gethelp/ >> > >> > Why do mathematicians always confuse Halloween and Christmas? >> > Because 31 Oct = 25 Dec. >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > Sorry to post this message here, but it seems my email ID is not authorized > to send emails to the list without moderation. > > I had sent two separate emails yesterday to the list, and they are still > not showing up here. Can someone please help? > > Regards > HASSAN > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/c55017f6/attachment.html From jeff at jefflenk.com Wed Aug 11 12:15:34 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 11 Aug 2010 12:15:34 -0700 (PDT) Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <55D75CA26DF8494B9FD88112BEAE8211@left> References: <4508448D788749F6B25886D626C17C28@left> <563B3EF6EAFE41BC8506DDA53B45F7F8@left> <396D3026B6034431B93A6B3DC0803461@left> <1281542398575-5412733.post@n2.nabble.com> <55D75CA26DF8494B9FD88112BEAE8211@left> Message-ID: <1281554134702-5413561.post@n2.nabble.com> Line 3998 of Sofia.c switch_channel_hangup(channel, SWITCH_CAUSE_REDIRECTION_TO_NEW_DESTINATION); -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p5413561.html Sent from the freeswitch-users mailing list archive at Nabble.com. From stephen at stephenjc.com Wed Aug 11 12:28:37 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Wed, 11 Aug 2010 15:28:37 -0400 Subject: [Freeswitch-users] cepstral problem Message-ID: I'm working with cepstral support and they seem to have no idea. Does any one know why cepstral might produce no audio on a cent os 5.5 machine. Its works fine on another box of mine. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print From john at area-europa.net Wed Aug 11 12:15:49 2010 From: john at area-europa.net (Area Europa) Date: Wed, 11 Aug 2010 21:15:49 +0200 Subject: [Freeswitch-users] SIP CANCEL Problems Message-ID: <4C62F6E5.4070205@area-europa.net> Hi List I'm having problems when a calling extension hangs up before called extension answers. Called extension keeps ringing for quite a while after calling party hangs up, giving called party plenty of time to pick up and listen to dead audio. Here's the set-up Ext 1000 ---- | |---- NAT Router ----- FS on Public IP | Ext 1002 ---- Both Extensions are on same LAN behind same NAT router, using different SIP ports: Ext 1000: 41000 Ext 1002: 41002 FreeSwitch is installed from GIT and conf files are basically unchanged, except IPs. I've pasted a complete trace of transaction below, but I think of relevance is that FreeSwitch receives 5 CANCEL packets from Ext 1000 over a 7 second period before it decides to send the CANCEL on to ext 1002. Seven seconds is a long time for a phone to be ringing. Any ideas on what / where to tweak to get FreeSwitch to respond after first CANCEL would be appreciated. Thanks in advance John Complete trace of transaction follows: freeswitch at internal> recv 993 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:03.892139: ------------------------------------------------------------------------ INVITE sip:1002 at 93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-141b2a92 From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Remote-Party-ID: "1000" ;screen=yes;party=calling Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 101 INVITE Max-Forwards: 70 Contact: "1000" Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 423343 423343 IN IP4 95.xxx.yyy.231 s=- c=IN IP4 95.xxx.yyy.231 t=0 0 m=audio 16384 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 318 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:03.892641: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-141b2a92 From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-08-11 21:01:03.892755 [DEBUG] sofia.c:6000 IP 95.xxx.yyy.231 Rejected by acl "domains". Falling back to Digest auth. send 805 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:03.901957: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-141b2a92 From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=mr2Kt3U0cB2FF Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="93.aaa.bbb.186", nonce="ca0e3c26-a57a-11df-8498-6130e0d7e4c0", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 401 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:03.981956: ------------------------------------------------------------------------ ACK sip:1002 at 93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-141b2a92 From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=mr2Kt3U0cB2FF Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 101 ACK Max-Forwards: 70 Contact: "1000" User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ recv 1229 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:04.018603: ------------------------------------------------------------------------ INVITE sip:1002 at 93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Remote-Party-ID: "1000" ;screen=yes;party=calling Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="93.aaa.bbb.186",nonce="ca0e3c26-a57a-11df-8498-6130e0d7e4c0",uri="sip:1002 at 93.aaa.bbb.186",algorithm=MD5,response="100a38311f3ee9f12c5fb78e61fdf14d",qop=auth,nc=00000001,cnonce="54f72ed1" Contact: "1000" Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 423343 423343 IN IP4 95.xxx.yyy.231 s=- c=IN IP4 95.xxx.yyy.231 t=0 0 m=audio 16384 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 318 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:04.019014: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-08-11 21:01:04.018556 [DEBUG] sofia.c:6000 IP 95.xxx.yyy.231 Rejected by acl "domains". Falling back to Digest auth. 2010-08-11 21:01:04.021204 [NOTICE] switch_channel.c:779 New Channel sofia/internal/1000 at 93.aaa.bbb.186 [ca21d98e-a57a-11df-8499-6130e0d7e4c0] 2010-08-11 21:01:04.022284 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 93.aaa.bbb.186) Running State Change CS_NEW 2010-08-11 21:01:04.022284 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 93.aaa.bbb.186) State NEW 2010-08-11 21:01:04.023433 [DEBUG] sofia.c:4318 Channel sofia/internal/1000 at 93.aaa.bbb.186 entering state [received][100] 2010-08-11 21:01:04.023433 [DEBUG] sofia.c:4329 Remote SDP: v=0 o=- 423343 423343 IN IP4 95.xxx.yyy.231 s=- c=IN IP4 95.xxx.yyy.231 t=0 0 m=audio 16384 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3847 Audio Codec Compare [PCMU:0:8000:30]/[G729:18:8000:20] 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3847 Audio Codec Compare [PCMU:0:8000:30]/[GSM:3:8000:20] 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3847 Audio Codec Compare [PCMU:0:8000:30]/[G7221:115:32000:20] 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3847 Audio Codec Compare [PCMU:0:8000:30]/[G7221:107:16000:20] 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3847 Audio Codec Compare [PCMU:0:8000:30]/[G722:9:8000:20] 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3847 Audio Codec Compare [PCMU:0:8000:30]/[PCMU:0:8000:20] 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3847 Audio Codec Compare [PCMU:0:8000:30]/[PCMA:8:8000:20] 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3895 Substituting codec PCMU at 30i@8000h 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:2444 Set Codec sofia/internal/1000 at 93.aaa.bbb.186 PCMU/8000 30 ms 240 samples 2010-08-11 21:01:04.023433 [DEBUG] sofia_glue.c:3943 Set 2833 dtmf send/recv payload to 101 2010-08-11 21:01:04.023433 [DEBUG] sofia.c:4476 (sofia/internal/1000 at 93.aaa.bbb.186) State Change CS_NEW -> CS_INIT 2010-08-11 21:01:04.023433 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/1000 at 93.aaa.bbb.186 [BREAK] 2010-08-11 21:01:04.024600 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 93.aaa.bbb.186) Running State Change CS_INIT 2010-08-11 21:01:04.024600 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 93.aaa.bbb.186) State INIT 2010-08-11 21:01:04.024600 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 93.aaa.bbb.186 SOFIA INIT 2010-08-11 21:01:04.024600 [DEBUG] mod_sofia.c:119 (sofia/internal/1000 at 93.aaa.bbb.186) State Change CS_INIT -> CS_ROUTING 2010-08-11 21:01:04.024600 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/1000 at 93.aaa.bbb.186 [BREAK] 2010-08-11 21:01:04.024600 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 93.aaa.bbb.186) State INIT going to sleep 2010-08-11 21:01:04.024600 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 93.aaa.bbb.186) Running State Change CS_ROUTING 2010-08-11 21:01:04.024600 [DEBUG] switch_channel.c:1512 (sofia/internal/1000 at 93.aaa.bbb.186) Callstate Change DOWN -> RINGING 2010-08-11 21:01:04.024600 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 93.aaa.bbb.186) State ROUTING 2010-08-11 21:01:04.024600 [DEBUG] mod_sofia.c:142 sofia/internal/1000 at 93.aaa.bbb.186 SOFIA ROUTING 2010-08-11 21:01:04.024600 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000 at 93.aaa.bbb.186 Standard ROUTING 2010-08-11 21:01:04.024600 [INFO] mod_dialplan_xml.c:331 Processing 1000->1002 in context default Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->unloop] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->tod_example] continue=true Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Date/Time Match (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [global-intercept] destination_number(1002) =~ /^886$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [group-intercept] destination_number(1002) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [intercept-ext] destination_number(1002) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->redial] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [redial] destination_number(1002) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->global] continue=true Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Absolute Condition [global] Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [snom-demo-2] destination_number(1002) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [snom-demo-1] destination_number(1002) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [eavesdrop] destination_number(1002) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [eavesdrop] destination_number(1002) =~ /^779$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->call_return] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [call_return] destination_number(1002) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->del-group] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [del-group] destination_number(1002) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->add-group] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [add-group] destination_number(1002) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [call-group-simo] destination_number(1002) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [call-group-order] destination_number(1002) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (FAIL) [extension-intercom] destination_number(1002) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Regex (PASS) [Local_Extension] destination_number(1002) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action set(dialed_extension=1002) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action export(dialed_extension=1002) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action set(ringback=${us-ring}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action set(call_timeout=30) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action set(continue_on_fail=true) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action answer() Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action sleep(1000) Dialplan: sofia/internal/1000 at 93.aaa.bbb.186 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2010-08-11 21:01:04.025632 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000 at 93.aaa.bbb.186) State Change CS_ROUTING -> CS_EXECUTE 2010-08-11 21:01:04.025632 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/1000 at 93.aaa.bbb.186 [BREAK] 2010-08-11 21:01:04.025632 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 93.aaa.bbb.186) State ROUTING going to sleep 2010-08-11 21:01:04.025632 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 93.aaa.bbb.186) Running State Change CS_EXECUTE 2010-08-11 21:01:04.025632 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 93.aaa.bbb.186) State EXECUTE 2010-08-11 21:01:04.025632 [DEBUG] mod_sofia.c:235 sofia/internal/1000 at 93.aaa.bbb.186 SOFIA EXECUTE 2010-08-11 21:01:04.025632 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000 at 93.aaa.bbb.186 Standard EXECUTE EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 hash(insert/93.aaa.bbb.186-spymap/1000/ca21d98e-a57a-11df-8499-6130e0d7e4c0) EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 hash(insert/93.aaa.bbb.186-last_dial/1000/1002) EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 hash(insert/93.aaa.bbb.186-last_dial/global/ca21d98e-a57a-11df-8499-6130e0d7e4c0) EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 set(dialed_extension=1002) 2010-08-11 21:01:04.026839 [DEBUG] mod_dptools.c:854 sofia/internal/1000 at 93.aaa.bbb.186 SET [dialed_extension]=[1002] EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 export(dialed_extension=1002) 2010-08-11 21:01:04.026839 [DEBUG] mod_dptools.c:938 EXPORT [dialed_extension]=[1002] EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 bind_meta_app(1 b s execute_extension::dx XML features) 2010-08-11 21:01:04.027981 [INFO] switch_ivr_async.c:2464 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1000.2010-08-11-21-01-04.wav) 2010-08-11 21:01:04.027981 [INFO] switch_ivr_async.c:2464 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2010-08-11-21-01-04.wav EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 bind_meta_app(3 b s execute_extension::cf XML features) 2010-08-11 21:01:04.027981 [INFO] switch_ivr_async.c:2464 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 set(ringback=%(2000,4000,440.0,480.0)) 2010-08-11 21:01:04.027981 [DEBUG] mod_dptools.c:854 sofia/internal/1000 at 93.aaa.bbb.186 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 set(transfer_ringback=local_stream://moh) 2010-08-11 21:01:04.029110 [DEBUG] mod_dptools.c:854 sofia/internal/1000 at 93.aaa.bbb.186 SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 set(call_timeout=30) 2010-08-11 21:01:04.029110 [DEBUG] mod_dptools.c:854 sofia/internal/1000 at 93.aaa.bbb.186 SET [call_timeout]=[30] EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 set(hangup_after_bridge=true) 2010-08-11 21:01:04.029110 [DEBUG] mod_dptools.c:854 sofia/internal/1000 at 93.aaa.bbb.186 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 set(continue_on_fail=true) 2010-08-11 21:01:04.029110 [DEBUG] mod_dptools.c:854 sofia/internal/1000 at 93.aaa.bbb.186 SET [continue_on_fail]=[true] EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 hash(insert/93.aaa.bbb.186-call_return/1002/1000) EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 hash(insert/93.aaa.bbb.186-last_dial_ext/1002/ca21d98e-a57a-11df-8499-6130e0d7e4c0) EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 set(called_party_callgroup=techsupport) 2010-08-11 21:01:04.030434 [DEBUG] mod_dptools.c:854 sofia/internal/1000 at 93.aaa.bbb.186 SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 hash(insert/93.aaa.bbb.186-last_dial/techsupport/ca21d98e-a57a-11df-8499-6130e0d7e4c0) EXECUTE sofia/internal/1000 at 93.aaa.bbb.186 bridge(user/1002 at 93.aaa.bbb.186) 2010-08-11 21:01:04.032457 [DEBUG] switch_ivr_originate.c:1979 variable string 0 = [presence_id=1002 at 93.aaa.bbb.186] 2010-08-11 21:01:04.033462 [NOTICE] switch_channel.c:779 New Channel sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [ca23b600-a57a-11df-849a-6130e0d7e4c0] 2010-08-11 21:01:04.040579 [DEBUG] mod_sofia.c:3892 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State Change CS_NEW -> CS_INIT 2010-08-11 21:01:04.040579 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [BREAK] 2010-08-11 21:01:04.040579 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Running State Change CS_INIT 2010-08-11 21:01:04.041592 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State INIT 2010-08-11 21:01:04.041592 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 SOFIA INIT 2010-08-11 21:01:04.041592 [DEBUG] mod_sofia.c:119 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State Change CS_INIT -> CS_ROUTING 2010-08-11 21:01:04.041592 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [BREAK] 2010-08-11 21:01:04.041592 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State INIT going to sleep 2010-08-11 21:01:04.041592 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Running State Change CS_ROUTING 2010-08-11 21:01:04.041592 [DEBUG] switch_channel.c:1512 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Callstate Change DOWN -> RINGING 2010-08-11 21:01:04.041592 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State ROUTING 2010-08-11 21:01:04.041592 [DEBUG] mod_sofia.c:142 sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 SOFIA ROUTING 2010-08-11 21:01:04.041592 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-08-11 21:01:04.041592 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [BREAK] 2010-08-11 21:01:04.041592 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State ROUTING going to sleep 2010-08-11 21:01:04.041592 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Running State Change CS_CONSUME_MEDIA 2010-08-11 21:01:04.041592 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State CONSUME_MEDIA 2010-08-11 21:01:04.041592 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State CONSUME_MEDIA going to sleep send 1219 bytes to udp/[95.xxx.yyy.231]:41002 at 19:01:04.042665: ------------------------------------------------------------------------ INVITE sip:1002 at 95.xxx.yyy.231:41002 SIP/2.0 Via: SIP/2.0/UDP 93.aaa.bbb.186;rport;branch=z9hG4bKyvX4D3Z0yHZ5e Max-Forwards: 69 From: "Extension 1000" ;tag=paN5XSX76veNp To: Call-ID: a1834f63-201d-122e-719b-a4badbdfa48e CSeq: 440568 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 318 X-FS-Support: update_display Remote-Party-ID: "Extension 1000" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1281531490 1281531491 IN IP4 93.aaa.bbb.186 s=FreeSWITCH c=IN IP4 93.aaa.bbb.186 t=0 0 m=audio 21774 RTP/AVP 0 18 9 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30 ------------------------------------------------------------------------ 2010-08-11 21:01:04.041592 [DEBUG] sofia.c:4318 Channel sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 entering state [calling][0] recv 301 bytes from udp/[95.xxx.yyy.231]:41002 at 19:01:04.120370: ------------------------------------------------------------------------ SIP/2.0 100 Trying To: From: "Extension 1000" ;tag=paN5XSX76veNp Call-ID: a1834f63-201d-122e-719b-a4badbdfa48e CSeq: 440568 INVITE Via: SIP/2.0/UDP 93.aaa.bbb.186;branch=z9hG4bKyvX4D3Z0yHZ5e Server: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ recv 397 bytes from udp/[95.xxx.yyy.231]:41002 at 19:01:04.264793: ------------------------------------------------------------------------ SIP/2.0 180 Ringing To: ;tag=dd77407622bd885i0 From: "Extension 1000" ;tag=paN5XSX76veNp Call-ID: a1834f63-201d-122e-719b-a4badbdfa48e CSeq: 440568 INVITE Via: SIP/2.0/UDP 93.aaa.bbb.186;branch=z9hG4bKyvX4D3Z0yHZ5e Server: Linksys/SPA941-5.1.8 Remote-Party-ID: "1002" ;screen=yes;party=called Content-Length: 0 ------------------------------------------------------------------------ 2010-08-11 21:01:04.264258 [INFO] sofia.c:662 sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 Update Callee ID to "1002" <1002> 2010-08-11 21:01:04.273760 [DEBUG] sofia.c:4318 Channel sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 entering state [proceeding][180] 2010-08-11 21:01:04.273760 [NOTICE] sofia.c:4390 Ring-Ready sofia/internal/sip:1002 at 95.xxx.yyy.231:41002! 2010-08-11 21:01:04.276790 [INFO] switch_ivr_originate.c:1079 Sending early media 2010-08-11 21:01:04.276790 [DEBUG] sofia_glue.c:2684 AUDIO RTP [sofia/internal/1000 at 93.aaa.bbb.186] 93.aaa.bbb.186 port 29380 -> 95.xxx.yyy.231 port 16384 codec: 0 ms: 30 2010-08-11 21:01:04.276790 [DEBUG] switch_rtp.c:1413 Starting timer [soft] 240 bytes per 30ms 2010-08-11 21:01:04.278977 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send payload to 101 2010-08-11 21:01:04.278977 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive payload to 101 2010-08-11 21:01:04.278977 [DEBUG] mod_sofia.c:2144 Ring SDP: v=0 o=FreeSWITCH 1281523884 1281523885 IN IP4 93.aaa.bbb.186 s=FreeSWITCH c=IN IP4 93.aaa.bbb.186 t=0 0 m=audio 29380 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv 2010-08-11 21:01:04.278977 [NOTICE] mod_sofia.c:2147 Pre-Answer sofia/internal/1000 at 93.aaa.bbb.186! 2010-08-11 21:01:04.278977 [DEBUG] switch_channel.c:2397 (sofia/internal/1000 at 93.aaa.bbb.186) Callstate Change RINGING -> EARLY 2010-08-11 21:01:04.278977 [DEBUG] switch_core_session.c:658 Send signal sofia/internal/1000 at 93.aaa.bbb.186 [BREAK] 2010-08-11 21:01:04.278977 [DEBUG] switch_ivr_originate.c:1128 Raw Codec Activation Success L16 at 8000hz 1 channel 30ms 2010-08-11 21:01:04.278977 [DEBUG] switch_core_codec.c:116 sofia/internal/1000 at 93.aaa.bbb.186 Push codec L16:10 2010-08-11 21:01:04.278977 [DEBUG] switch_ivr_originate.c:1193 Play Ringback Tone [%(2000,4000,440.0,480.0)] send 1118 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:04.279808: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 Remote-Party-ID: "1002" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1281523884 1281523885 IN IP4 93.aaa.bbb.186 s=FreeSWITCH c=IN IP4 93.aaa.bbb.186 t=0 0 m=audio 29380 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 ------------------------------------------------------------------------ 2010-08-11 21:01:04.280443 [DEBUG] sofia.c:4313 Channel sofia/internal/1000 at 93.aaa.bbb.186 skipping state [early][183] 2010-08-11 21:01:04.405197 [DEBUG] switch_rtp.c:2527 Correct ip/port confirmed. recv 578 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:07.276699: ------------------------------------------------------------------------ CANCEL sip:1002 at 93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:4100000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 CANCEL Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="93.aaa.bbb.186",nonce="ca0e3c26-a57a-11df-8498-6130e0d7e4c0",uri="sip:1002 at 93.aaa.bbb.186",algorithm=MD5,response="314faae6a65e754939bf9f5678a2fd60",qop=auth,nc=00000002,cnonce="54f72ed1" User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ recv 578 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:07.764599: ------------------------------------------------------------------------ CANCEL sip:1002 at 93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:4100000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 CANCEL Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="93.aaa.bbb.186",nonce="ca0e3c26-a57a-11df-8498-6130e0d7e4c0",uri="sip:1002 at 93.aaa.bbb.186",algorithm=MD5,response="314faae6a65e754939bf9f5678a2fd60",qop=auth,nc=00000002,cnonce="54f72ed1" User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ recv 578 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:08.765910: ------------------------------------------------------------------------ CANCEL sip:1002 at 93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:4100000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 CANCEL Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="93.aaa.bbb.186",nonce="ca0e3c26-a57a-11df-8498-6130e0d7e4c0",uri="sip:1002 at 93.aaa.bbb.186",algorithm=MD5,response="314faae6a65e754939bf9f5678a2fd60",qop=auth,nc=00000002,cnonce="54f72ed1" User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ recv 578 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:10.764272: ------------------------------------------------------------------------ CANCEL sip:1002 at 93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:4100000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 CANCEL Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="93.aaa.bbb.186",nonce="ca0e3c26-a57a-11df-8498-6130e0d7e4c0",uri="sip:1002 at 93.aaa.bbb.186",algorithm=MD5,response="314faae6a65e754939bf9f5678a2fd60",qop=auth,nc=00000002,cnonce="54f72ed1" User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ recv 576 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:14.762786: ------------------------------------------------------------------------ CANCEL sip:1002 at 93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 CANCEL Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="93.aaa.bbb.186",nonce="ca0e3c26-a57a-11df-8498-6130e0d7e4c0",uri="sip:1002 at 93.aaa.bbb.186",algorithm=MD5,response="314faae6a65e754939bf9f5678a2fd60",qop=auth,nc=00000002,cnonce="54f72ed1" User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ send 284 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:14.763018: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 CANCEL Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:14.763162: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2010-08-11 21:01:14.762892 [DEBUG] sofia.c:4318 Channel sofia/internal/1000 at 93.aaa.bbb.186 entering state [terminated][487] 2010-08-11 21:01:14.762892 [DEBUG] switch_channel.c:2309 (sofia/internal/1000 at 93.aaa.bbb.186) Callstate Change EARLY -> HANGUP 2010-08-11 21:01:14.762892 [NOTICE] sofia.c:4932 Hangup sofia/internal/1000 at 93.aaa.bbb.186 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2010-08-11 21:01:14.762892 [DEBUG] switch_channel.c:2325 Send signal sofia/internal/1000 at 93.aaa.bbb.186 [KILL] 2010-08-11 21:01:14.762892 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/1000 at 93.aaa.bbb.186 [BREAK] 2010-08-11 21:01:14.785078 [DEBUG] switch_core_codec.c:140 sofia/internal/1000 at 93.aaa.bbb.186 Restore previous codec PCMU:0. 2010-08-11 21:01:14.785078 [DEBUG] switch_channel.c:2309 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Callstate Change RINGING -> HANGUP 2010-08-11 21:01:14.785078 [NOTICE] switch_ivr_originate.c:3282 Hangup sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2010-08-11 21:01:14.785078 [DEBUG] switch_channel.c:2325 Send signal sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [KILL] 2010-08-11 21:01:14.785078 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [BREAK] 2010-08-11 21:01:14.785078 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Running State Change CS_HANGUP 2010-08-11 21:01:14.785078 [DEBUG] switch_ivr_originate.c:3425 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2010-08-11 21:01:14.785078 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State HANGUP 2010-08-11 21:01:14.785078 [DEBUG] mod_sofia.c:447 sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 Overriding SIP cause 487 with 487 from the other leg 2010-08-11 21:01:14.785078 [DEBUG] mod_sofia.c:453 Channel sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 hanging up, cause: ORIGINATOR_CANCEL 2010-08-11 21:01:14.785078 [ERR] switch_ivr_originate.c:2623 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-08-11 21:01:14.785078 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-08-11 21:01:14.785078 [INFO] mod_dptools.c:2393 Originate Failed. Cause: ORIGINATOR_CANCEL 2010-08-11 21:01:14.788208 [DEBUG] switch_core_session.c:1905 sofia/internal/1000 at 93.aaa.bbb.186 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2010-08-11 21:01:14.788208 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 93.aaa.bbb.186) State EXECUTE going to sleep 2010-08-11 21:01:14.788208 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 93.aaa.bbb.186) Running State Change CS_HANGUP 2010-08-11 21:01:14.788208 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1000 at 93.aaa.bbb.186) State HANGUP 2010-08-11 21:01:14.788208 [DEBUG] mod_sofia.c:447 sofia/internal/1000 at 93.aaa.bbb.186 Overriding SIP cause 487 with 487 from the other leg 2010-08-11 21:01:14.788208 [DEBUG] mod_sofia.c:453 Channel sofia/internal/1000 at 93.aaa.bbb.186 hanging up, cause: ORIGINATOR_CANCEL 2010-08-11 21:01:14.797311 [DEBUG] mod_sofia.c:506 Sending CANCEL to sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 Standard HANGUP, cause: ORIGINATOR_CANCEL 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State HANGUP going to sleep 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State Change CS_HANGUP -> CS_REPORTING 2010-08-11 21:01:14.797311 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [BREAK] 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Running State Change CS_REPORTING 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State REPORTING 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 Standard REPORTING, cause: ORIGINATOR_CANCEL 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State REPORTING going to sleep 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State Change CS_REPORTING -> CS_DESTROY 2010-08-11 21:01:14.797311 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [BREAK] 2010-08-11 21:01:14.797311 [DEBUG] switch_core_session.c:1202 Session 16 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Locked, Waiting on external entities 2010-08-11 21:01:14.797311 [NOTICE] switch_core_session.c:1220 Session 16 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Ended 2010-08-11 21:01:14.797311 [NOTICE] switch_core_session.c:1222 Close Channel sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 [CS_DESTROY] 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Callstate Change HANGUP -> DOWN 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) Running State Change CS_DESTROY 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State DESTROY 2010-08-11 21:01:14.797311 [DEBUG] mod_sofia.c:358 sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 SOFIA DESTROY 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1002 at 95.xxx.yyy.231:41002 Standard DESTROY 2010-08-11 21:01:14.797311 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:1002 at 95.xxx.yyy.231:41002) State DESTROY going to sleep send 376 bytes to udp/[95.xxx.yyy.231]:41002 at 19:01:14.798471: ------------------------------------------------------------------------ CANCEL sip:1002 at 95.xxx.yyy.231:41002 SIP/2.0 Via: SIP/2.0/UDP 93.aaa.bbb.186;rport;branch=z9hG4bKyvX4D3Z0yHZ5e Max-Forwards: 69 From: "Extension 1000" ;tag=paN5XSX76veNp To: Call-ID: a1834f63-201d-122e-719b-a4badbdfa48e CSeq: 440568 CANCEL Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 ------------------------------------------------------------------------ 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000 at 93.aaa.bbb.186 Standard HANGUP, cause: ORIGINATOR_CANCEL 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1000 at 93.aaa.bbb.186) State HANGUP going to sleep 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1000 at 93.aaa.bbb.186) State Change CS_HANGUP -> CS_REPORTING 2010-08-11 21:01:14.801620 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/1000 at 93.aaa.bbb.186 [BREAK] 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 93.aaa.bbb.186) Running State Change CS_REPORTING 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/1000 at 93.aaa.bbb.186) State REPORTING 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1000 at 93.aaa.bbb.186 Standard REPORTING, cause: ORIGINATOR_CANCEL 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/1000 at 93.aaa.bbb.186) State REPORTING going to sleep 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1000 at 93.aaa.bbb.186) State Change CS_REPORTING -> CS_DESTROY 2010-08-11 21:01:14.801620 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/1000 at 93.aaa.bbb.186 [BREAK] 2010-08-11 21:01:14.801620 [DEBUG] switch_core_session.c:1202 Session 15 (sofia/internal/1000 at 93.aaa.bbb.186) Locked, Waiting on external entities 2010-08-11 21:01:14.801620 [NOTICE] switch_core_session.c:1220 Session 15 (sofia/internal/1000 at 93.aaa.bbb.186) Ended 2010-08-11 21:01:14.801620 [NOTICE] switch_core_session.c:1222 Close Channel sofia/internal/1000 at 93.aaa.bbb.186 [CS_DESTROY] 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/1000 at 93.aaa.bbb.186) Callstate Change HANGUP -> DOWN 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/1000 at 93.aaa.bbb.186) Running State Change CS_DESTROY 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at 93.aaa.bbb.186) State DESTROY 2010-08-11 21:01:14.801620 [DEBUG] mod_sofia.c:358 sofia/internal/1000 at 93.aaa.bbb.186 SOFIA DESTROY 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000 at 93.aaa.bbb.186 Standard DESTROY 2010-08-11 21:01:14.801620 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/1000 at 93.aaa.bbb.186) State DESTROY going to sleep recv 335 bytes from udp/[95.xxx.yyy.231]:41002 at 19:01:14.875581: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated To: ;tag=dd77407622bd885i0 From: "Extension 1000" ;tag=paN5XSX76veNp Call-ID: a1834f63-201d-122e-719b-a4badbdfa48e CSeq: 440568 INVITE Via: SIP/2.0/UDP 93.aaa.bbb.186;branch=z9hG4bKyvX4D3Z0yHZ5e Server: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ send 337 bytes to udp/[95.xxx.yyy.231]:41002 at 19:01:14.875798: ------------------------------------------------------------------------ ACK sip:1002 at 95.xxx.yyy.231:41002 SIP/2.0 Via: SIP/2.0/UDP 93.aaa.bbb.186;rport;branch=z9hG4bKyvX4D3Z0yHZ5e Max-Forwards: 69 From: "Extension 1000" ;tag=paN5XSX76veNp To: ;tag=dd77407622bd885i0 Call-ID: a1834f63-201d-122e-719b-a4badbdfa48e CSeq: 440568 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 319 bytes from udp/[95.xxx.yyy.231]:41002 at 19:01:14.887136: ------------------------------------------------------------------------ SIP/2.0 200 OK To: ;tag=dd77407622bd885i0 From: "Extension 1000" ;tag=paN5XSX76veNp Call-ID: a1834f63-201d-122e-719b-a4badbdfa48e CSeq: 440568 CANCEL Via: SIP/2.0/UDP 93.aaa.bbb.186;branch=z9hG4bKyvX4D3Z0yHZ5e Server: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:15.263994: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:16.264001: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:18.264000: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:22.264012: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:26.264005: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:30.264009: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 331 bytes from udp/[95.xxx.yyy.231]:41002 at 19:01:30.347502: ------------------------------------------------------------------------ NOTIFY sip:93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:41002;branch=z9hG4bK-adf19f8b From: "1002" ;tag=6c504ae2eadd96bfo0 To: Call-ID: bf54057e-64adbcd3 at 192.168.1.40 CSeq: 70 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ send 637 bytes to udp/[95.xxx.yyy.231]:41002 at 19:01:30.347847: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 95.xxx.yyy.231:41002;branch=z9hG4bK-adf19f8b From: "1002" ;tag=6c504ae2eadd96bfo0 To: ;tag=QKeyZmeB4547H Call-ID: bf54057e-64adbcd3 at 192.168.1.40 CSeq: 70 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:34.268007: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:38.268010: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 331 bytes from udp/[95.xxx.yyy.231]:41000 at 19:01:39.777149: ------------------------------------------------------------------------ NOTIFY sip:93.aaa.bbb.186 SIP/2.0 Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-4494dffc From: "1000" ;tag=9ddc56accbaeed50o0 To: Call-ID: 8acf935c-c8ed5860 at 192.168.1.34 CSeq: 71 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 ------------------------------------------------------------------------ send 637 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:39.777522: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-4494dffc From: "1000" ;tag=9ddc56accbaeed50o0 To: ;tag=rv7p1FZe1eUtD Call-ID: 8acf935c-c8ed5860 at 192.168.1.34 CSeq: 71 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:42.268669: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[95.xxx.yyy.231]:41000 at 19:01:46.268661: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 95.xxx.yyy.231:41000;branch=z9hG4bK-bf655f9e From: "1000" ;tag=ae60c1d11550e891o0 To: "Extension 1002" ;tag=N1Ucvyc49Kr2a Call-ID: 97c7aad1-f86f1391 at 192.168.1.34 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ freeswitch at internal> ... From Victor at isptelecom.net Wed Aug 11 12:34:21 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Wed, 11 Aug 2010 15:34:21 -0400 Subject: [Freeswitch-users] FIFO queue callers ID: how to set to a real-life callers ID, not a static entry? Message-ID: <4C62FB3D.4030707@isptelecom.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/53c4a6ee/attachment.html From chris.veazey at gmail.com Wed Aug 11 13:17:40 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Wed, 11 Aug 2010 15:17:40 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <1281554134702-5413561.post@n2.nabble.com> References: <4508448D788749F6B25886D626C17C28@left> <563B3EF6EAFE41BC8506DDA53B45F7F8@left> <396D3026B6034431B93A6B3DC0803461@left> <1281542398575-5412733.post@n2.nabble.com> <55D75CA26DF8494B9FD88112BEAE8211@left> <1281554134702-5413561.post@n2.nabble.com> Message-ID: Recompiled. Now getting a continuous Invite-->302 loop. That eventually gets killed with a 503 by the upstream redirect server. [root at fs1 log]# tail -f freeswitch.log 2010-08-11 20:03:30.027817 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/9995552000000 [BREAK] 2010-08-11 20:03:30.027817 [DEBUG] switch_core_session.c:1202 Session 6 (sofia/external/9995552000000) Locked, Waiting on external entities 2010-08-11 20:03:30.027817 [NOTICE] switch_core_session.c:1220 Session 6 (sofia/external/9995552000000) Ended 2010-08-11 20:03:30.027817 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/9995552000000 [CS_DESTROY] 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:427 (sofia/external/9995552000000) Callstate Change HANGUP -> DOWN 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:430 (sofia/external/9995552000000) Running State Change CS_DESTROY 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:440 (sofia/external/9995552000000) State DESTROY 2010-08-11 20:03:30.027817 [DEBUG] mod_sofia.c:358 sofia/external/9995552000000 SOFIA DESTROY 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:60 sofia/external/9995552000000 Standard DESTROY 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:440 (sofia/external/9995552000000) State DESTROY going to sleep 2010-08-11 20:07:50.601786 [DEBUG] sofia.c:5999 IP 10.14.0.128 Rejected by acl "domains". Falling back to Digest auth. 2010-08-11 20:07:51.179975 [DEBUG] sofia.c:5999 IP 10.14.0.128 Rejected by acl "domains". Falling back to Digest auth. 2010-08-11 20:07:51.186996 [NOTICE] switch_channel.c:779 New Channel sofia/internal/9995551000000 at sip.blinkmind.net[1e96cac0-a584-11df-ba1a-11ad2b71c214] 2010-08-11 20:07:51.187997 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9995551000000 at sip.blinkmind.net) Running State Change CS_NEW 2010-08-11 20:07:51.187997 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/9995551000000 at sip.blinkmind.net) State NEW 2010-08-11 20:07:51.195018 [DEBUG] sofia.c:4317 Channel sofia/internal/ 9995551000000 at sip.blinkmind.net entering state [received][100] 2010-08-11 20:07:51.195018 [DEBUG] sofia.c:4328 Remote SDP: v=0 o=- 0 0 IN IP4 10.14.0.128 s=- c=IN IP4 10.14.0.128 t=0 0 m=audio 20596 RTP/AVP 107 0 101 a=rtpmap:107 CTL-SB-ADPCM/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 21114 RTP/AVP 125 a=rtpmap:125 H264/90000 a=fmtp:125 packetization-mode=1;profile-level-id=42E01F;sprop-parameter-sets=Z0IADPQLBKIA,aM44gAA= 2010-08-11 20:07:51.195018 [DEBUG] sofia.c:4445 (sofia/internal/ 9995551000000 at sip.blinkmind.net) State Change CS_NEW -> CS_INIT 2010-08-11 20:07:51.195018 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9995551000000 at sip.blinkmind.net [BREAK] 2010-08-11 20:07:51.195018 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9995551000000 at sip.blinkmind.net) Running State Change CS_INIT 2010-08-11 20:07:51.195018 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/9995551000000 at sip.blinkmind.net) State INIT 2010-08-11 20:07:51.195018 [DEBUG] mod_sofia.c:83 sofia/internal/ 9995551000000 at sip.blinkmind.net SOFIA INIT 2010-08-11 20:07:51.195018 [DEBUG] mod_sofia.c:119 (sofia/internal/ 9995551000000 at sip.blinkmind.net) State Change CS_INIT -> CS_ROUTING 2010-08-11 20:07:51.195018 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9995551000000 at sip.blinkmind.net [BREAK] 2010-08-11 20:07:51.195018 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/9995551000000 at sip.blinkmind.net) State INIT going to sleep 2010-08-11 20:07:51.195018 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9995551000000 at sip.blinkmind.net) Running State Change CS_ROUTING 2010-08-11 20:07:51.195018 [DEBUG] switch_channel.c:1512 (sofia/internal/ 9995551000000 at sip.blinkmind.net) Callstate Change DOWN -> RINGING 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/9995551000000 at sip.blinkmind.net) State ROUTING 2010-08-11 20:07:51.196022 [DEBUG] mod_sofia.c:142 sofia/internal/ 9995551000000 at sip.blinkmind.net SOFIA ROUTING 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:77 sofia/internal/9995551000000 at sip.blinkmind.net Standard ROUTING 2010-08-11 20:07:51.196022 [INFO] mod_dialplan_xml.c:331 Processing 9995551000000->9995552000000 in context default Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net parsing [default->unloop] continue=false Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net parsing [default->sip_uri] continue=false Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Regex (FAIL) [sip_uri] destination_number(9995552000000) =~ /^sip:(.*)$/ break=on-false Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net parsing [default->catchall] continue=false Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Regex (PASS) [catchall] destination_number(9995552000000) =~ /.*/ break=on-false Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Action bridge(sofia/gateway/sip.blinkmind.net/${destination_number}) 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/9995551000000 at sip.blinkmind.net) State Change CS_ROUTING -> CS_EXECUTE 2010-08-11 20:07:51.196022 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9995551000000 at sip.blinkmind.net [BREAK] 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/9995551000000 at sip.blinkmind.net) State ROUTING going to sleep 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9995551000000 at sip.blinkmind.net) Running State Change CS_EXECUTE 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/9995551000000 at sip.blinkmind.net) State EXECUTE 2010-08-11 20:07:51.196022 [DEBUG] mod_sofia.c:235 sofia/internal/ 9995551000000 at sip.blinkmind.net SOFIA EXECUTE 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:157 sofia/internal/9995551000000 at sip.blinkmind.net Standard EXECUTE EXECUTE sofia/internal/9995551000000 at sip.blinkmind.net bridge(sofia/gateway/ sip.blinkmind.net/9995552000000) 2010-08-11 20:07:51.196022 [NOTICE] switch_channel.c:779 New Channel sofia/external/9995552000000 [1e984e40-a584-11df-ba1b-11ad2b71c214] 2010-08-11 20:07:51.196022 [DEBUG] mod_sofia.c:3892 (sofia/external/9995552000000) State Change CS_NEW -> CS_INIT 2010-08-11 20:07:51.196022 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/9995552000000 [BREAK] 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:314 (sofia/external/9995552000000) Running State Change CS_INIT 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:338 (sofia/external/9995552000000) State INIT 2010-08-11 20:07:51.197023 [DEBUG] mod_sofia.c:83 sofia/external/9995552000000 SOFIA INIT 2010-08-11 20:07:51.197023 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.197023 [DEBUG] mod_sofia.c:119 (sofia/external/9995552000000) State Change CS_INIT -> CS_ROUTING 2010-08-11 20:07:51.197023 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/9995552000000 [BREAK] 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:338 (sofia/external/9995552000000) State INIT going to sleep 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:314 (sofia/external/9995552000000) Running State Change CS_ROUTING 2010-08-11 20:07:51.197023 [DEBUG] switch_channel.c:1512 (sofia/external/9995552000000) Callstate Change DOWN -> RINGING 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:341 (sofia/external/9995552000000) State ROUTING 2010-08-11 20:07:51.197023 [DEBUG] mod_sofia.c:142 sofia/external/9995552000000 SOFIA ROUTING 2010-08-11 20:07:51.197023 [DEBUG] switch_ivr_originate.c:66 (sofia/external/9995552000000) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-08-11 20:07:51.197023 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/9995552000000 [BREAK] 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:341 (sofia/external/9995552000000) State ROUTING going to sleep 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:314 (sofia/external/9995552000000) Running State Change CS_CONSUME_MEDIA 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:360 (sofia/external/9995552000000) State CONSUME_MEDIA 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:360 (sofia/external/9995552000000) State CONSUME_MEDIA going to sleep 2010-08-11 20:07:51.198023 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.201020 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.201020 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.203018 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.204015 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.206022 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.207023 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.209026 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.209026 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.212037 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.212037 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.215047 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.215047 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.217052 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.217052 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.220062 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.220062 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.222070 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.223073 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.225079 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.225079 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.227086 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 2010-08-11 20:07:51.228089 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [calling][0] 2010-08-11 20:07:51.230097 [DEBUG] sofia.c:4317 Channel sofia/external/9995552000000 entering state [terminated][503] 2010-08-11 20:07:51.230097 [DEBUG] switch_channel.c:2309 (sofia/external/9995552000000) Callstate Change RINGING -> HANGUP 2010-08-11 20:07:51.230097 [NOTICE] sofia.c:4931 Hangup sofia/external/9995552000000 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2010-08-11 20:07:51.230097 [DEBUG] switch_channel.c:2325 Send signal sofia/external/9995552000000 [KILL] 2010-08-11 20:07:51.230097 [DEBUG] switch_core_session.c:1039 Send signal sofia/external/9995552000000 [BREAK] 2010-08-11 20:07:51.230097 [DEBUG] switch_core_state_machine.c:314 (sofia/external/9995552000000) Running State Change CS_HANGUP 2010-08-11 20:07:51.230097 [DEBUG] switch_core_state_machine.c:535 (sofia/external/9995552000000) State HANGUP 2010-08-11 20:07:51.230097 [DEBUG] mod_sofia.c:447 sofia/external/9995552000000 Overriding SIP cause 503 with 503 from the other leg On Wed, Aug 11, 2010 at 2:15 PM, Jeff Lenk wrote: > > Line 3998 of Sofia.c > switch_channel_hangup(channel, > SWITCH_CAUSE_REDIRECTION_TO_NEW_DESTINATION); > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p5413561.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/0f4b223a/attachment-0001.html From brian at freeswitch.org Wed Aug 11 13:25:40 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Aug 2010 15:25:40 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: References: <4508448D788749F6B25886D626C17C28@left> <563B3EF6EAFE41BC8506DDA53B45F7F8@left> <396D3026B6034431B93A6B3DC0803461@left> <1281542398575-5412733.post@n2.nabble.com> <55D75CA26DF8494B9FD88112BEAE8211@left> <1281554134702-5413561.post@n2.nabble.com> Message-ID: <43420C41-3D0B-4FF4-9DC6-5880B11BEEB2@freeswitch.org> This line concerns me... care to show me what variables you're setting. Some will cause the proxy to lock on and never change. /b On Aug 11, 2010, at 3:17 PM, Chris Veazey wrote: > 2010-08-11 20:07:51.206022 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/9995552000000 From jeff at jefflenk.com Wed Aug 11 14:25:04 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 11 Aug 2010 14:25:04 -0700 (PDT) Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <43420C41-3D0B-4FF4-9DC6-5880B11BEEB2@freeswitch.org> References: <4508448D788749F6B25886D626C17C28@left> <563B3EF6EAFE41BC8506DDA53B45F7F8@left> <396D3026B6034431B93A6B3DC0803461@left> <1281542398575-5412733.post@n2.nabble.com> <55D75CA26DF8494B9FD88112BEAE8211@left> <1281554134702-5413561.post@n2.nabble.com> <43420C41-3D0B-4FF4-9DC6-5880B11BEEB2@freeswitch.org> Message-ID: <1281561904083-5414031.post@n2.nabble.com> Hi Brian, I have been investigating a problem with my FS gateway to OCS which has been suffering from this same problem. I finally found this line at fault today and am trying to figure out whether that line was added by accident or intentionally. Commenting out that code definitely fixed my problem. Jeff -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p5414031.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Wed Aug 11 14:39:19 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 11 Aug 2010 22:39:19 +0100 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: References: <4508448D788749F6B25886D626C17C28@left> <563B3EF6EAFE41BC8506DDA53B45F7F8@left> <396D3026B6034431B93A6B3DC0803461@left> <1281542398575-5412733.post@n2.nabble.com> <55D75CA26DF8494B9FD88112BEAE8211@left> <1281554134702-5413561.post@n2.nabble.com> Message-ID: AFAIK 302 reenters the dialplan. If the destination number is unchanged, it will execute the same commands after the redirect as it did before the redirect, which would likely cause it to continously bridge, redirect, bridge, redirect.... That could explain what you're experiencing. How do you want to handle the 302? It's advisable to be careful what to do with a 302 which is why it reenters the dialplan. You can try dialing a $0.01 route but have it redirect you to a $1.00 route leaving you sending the call out on a route costing you $1.00 but charging your customer $0.01, or if you do bill them correctly leaving them with a bill of $1.00 when they expected only $0.01. It's possible to handle a 302 manually by sending them to a dialplan context specifically for it, which keeps the redirect logic away from your main dialplan. See http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect -Steve On 11 August 2010 21:17, Chris Veazey wrote: > Recompiled. Now getting a continuous Invite-->302 loop. That eventually > gets killed with a 503 by the upstream redirect server. > > > [root at fs1 log]# tail -f freeswitch.log > 2010-08-11 20:03:30.027817 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/9995552000000 [BREAK] > 2010-08-11 20:03:30.027817 [DEBUG] switch_core_session.c:1202 Session 6 > (sofia/external/9995552000000) Locked, Waiting on external entities > 2010-08-11 20:03:30.027817 [NOTICE] switch_core_session.c:1220 Session 6 > (sofia/external/9995552000000) Ended > 2010-08-11 20:03:30.027817 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/9995552000000 [CS_DESTROY] > 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:427 > (sofia/external/9995552000000) Callstate Change HANGUP -> DOWN > 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:430 > (sofia/external/9995552000000) Running State Change CS_DESTROY > 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/9995552000000) State DESTROY > 2010-08-11 20:03:30.027817 [DEBUG] mod_sofia.c:358 > sofia/external/9995552000000 SOFIA DESTROY > 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:60 > sofia/external/9995552000000 Standard DESTROY > 2010-08-11 20:03:30.027817 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/9995552000000) State DESTROY going to sleep > 2010-08-11 20:07:50.601786 [DEBUG] sofia.c:5999 IP 10.14.0.128 Rejected by > acl "domains". Falling back to Digest auth. > 2010-08-11 20:07:51.179975 [DEBUG] sofia.c:5999 IP 10.14.0.128 Rejected by > acl "domains". Falling back to Digest auth. > 2010-08-11 20:07:51.186996 [NOTICE] switch_channel.c:779 New Channel > sofia/internal/9995551000000 at sip.blinkmind.net[1e96cac0-a584-11df-ba1a-11ad2b71c214] > 2010-08-11 20:07:51.187997 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9995551000000 at sip.blinkmind.net) Running State Change > CS_NEW > 2010-08-11 20:07:51.187997 [DEBUG] switch_core_state_machine.c:320 > (sofia/internal/9995551000000 at sip.blinkmind.net) State NEW > 2010-08-11 20:07:51.195018 [DEBUG] sofia.c:4317 Channel sofia/internal/ > 9995551000000 at sip.blinkmind.net entering state [received][100] > 2010-08-11 20:07:51.195018 [DEBUG] sofia.c:4328 Remote SDP: > v=0 > o=- 0 0 IN IP4 10.14.0.128 > s=- > c=IN IP4 10.14.0.128 > t=0 0 > m=audio 20596 RTP/AVP 107 0 101 > a=rtpmap:107 CTL-SB-ADPCM/16000 > > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > m=video 21114 RTP/AVP 125 > a=rtpmap:125 H264/90000 > a=fmtp:125 > packetization-mode=1;profile-level-id=42E01F;sprop-parameter-sets=Z0IADPQLBKIA,aM44gAA= > > 2010-08-11 20:07:51.195018 [DEBUG] sofia.c:4445 (sofia/internal/ > 9995551000000 at sip.blinkmind.net) State Change CS_NEW -> CS_INIT > 2010-08-11 20:07:51.195018 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9995551000000 at sip.blinkmind.net [BREAK] > 2010-08-11 20:07:51.195018 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9995551000000 at sip.blinkmind.net) Running State Change > CS_INIT > 2010-08-11 20:07:51.195018 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9995551000000 at sip.blinkmind.net) State INIT > 2010-08-11 20:07:51.195018 [DEBUG] mod_sofia.c:83 sofia/internal/ > 9995551000000 at sip.blinkmind.net SOFIA INIT > 2010-08-11 20:07:51.195018 [DEBUG] mod_sofia.c:119 (sofia/internal/ > 9995551000000 at sip.blinkmind.net) State Change CS_INIT -> CS_ROUTING > 2010-08-11 20:07:51.195018 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9995551000000 at sip.blinkmind.net [BREAK] > 2010-08-11 20:07:51.195018 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9995551000000 at sip.blinkmind.net) State INIT going to sleep > 2010-08-11 20:07:51.195018 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9995551000000 at sip.blinkmind.net) Running State Change > CS_ROUTING > 2010-08-11 20:07:51.195018 [DEBUG] switch_channel.c:1512 (sofia/internal/ > 9995551000000 at sip.blinkmind.net) Callstate Change DOWN -> RINGING > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9995551000000 at sip.blinkmind.net) State ROUTING > 2010-08-11 20:07:51.196022 [DEBUG] mod_sofia.c:142 sofia/internal/ > 9995551000000 at sip.blinkmind.net SOFIA ROUTING > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:77 > sofia/internal/9995551000000 at sip.blinkmind.net Standard ROUTING > 2010-08-11 20:07:51.196022 [INFO] mod_dialplan_xml.c:331 Processing > 9995551000000->9995552000000 in context default > Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net parsing > [default->unloop] continue=false > Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Regex (PASS) > [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Regex (FAIL) > [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net parsing > [default->sip_uri] continue=false > Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Regex (FAIL) > [sip_uri] destination_number(9995552000000) =~ /^sip:(.*)$/ break=on-false > Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net parsing > [default->catchall] continue=false > Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Regex (PASS) > [catchall] destination_number(9995552000000) =~ /.*/ break=on-false > Dialplan: sofia/internal/9995551000000 at sip.blinkmind.net Action > bridge(sofia/gateway/sip.blinkmind.net/${destination_number} > ) > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:119 > (sofia/internal/9995551000000 at sip.blinkmind.net) State Change CS_ROUTING > -> CS_EXECUTE > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9995551000000 at sip.blinkmind.net [BREAK] > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9995551000000 at sip.blinkmind.net) State ROUTING going to > sleep > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9995551000000 at sip.blinkmind.net) Running State Change > CS_EXECUTE > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/9995551000000 at sip.blinkmind.net) State EXECUTE > 2010-08-11 20:07:51.196022 [DEBUG] mod_sofia.c:235 sofia/internal/ > 9995551000000 at sip.blinkmind.net SOFIA EXECUTE > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_state_machine.c:157 > sofia/internal/9995551000000 at sip.blinkmind.net Standard EXECUTE > EXECUTE sofia/internal/9995551000000 at sip.blinkmind.netbridge(sofia/gateway/ > sip.blinkmind.net/9995552000000) > 2010-08-11 20:07:51.196022 [NOTICE] switch_channel.c:779 New Channel > sofia/external/9995552000000 [1e984e40-a584-11df-ba1b-11ad2b71c214] > 2010-08-11 20:07:51.196022 [DEBUG] mod_sofia.c:3892 > (sofia/external/9995552000000) State Change CS_NEW -> CS_INIT > 2010-08-11 20:07:51.196022 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/9995552000000 [BREAK] > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/9995552000000) Running State Change CS_INIT > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:338 > (sofia/external/9995552000000) State INIT > 2010-08-11 20:07:51.197023 [DEBUG] mod_sofia.c:83 > sofia/external/9995552000000 SOFIA INIT > 2010-08-11 20:07:51.197023 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.197023 [DEBUG] mod_sofia.c:119 > (sofia/external/9995552000000) State Change CS_INIT -> CS_ROUTING > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/9995552000000 [BREAK] > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:338 > (sofia/external/9995552000000) State INIT going to sleep > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/9995552000000) Running State Change CS_ROUTING > 2010-08-11 20:07:51.197023 [DEBUG] switch_channel.c:1512 > (sofia/external/9995552000000) Callstate Change DOWN -> RINGING > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:341 > (sofia/external/9995552000000) State ROUTING > 2010-08-11 20:07:51.197023 [DEBUG] mod_sofia.c:142 > sofia/external/9995552000000 SOFIA ROUTING > 2010-08-11 20:07:51.197023 [DEBUG] switch_ivr_originate.c:66 > (sofia/external/9995552000000) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/9995552000000 [BREAK] > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:341 > (sofia/external/9995552000000) State ROUTING going to sleep > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/9995552000000) Running State Change CS_CONSUME_MEDIA > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:360 > (sofia/external/9995552000000) State CONSUME_MEDIA > 2010-08-11 20:07:51.197023 [DEBUG] switch_core_state_machine.c:360 > (sofia/external/9995552000000) State CONSUME_MEDIA going to sleep > 2010-08-11 20:07:51.198023 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.201020 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.201020 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.203018 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.204015 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.206022 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.207023 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.209026 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.209026 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.212037 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.212037 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.215047 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.215047 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.217052 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.217052 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.220062 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.220062 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.222070 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.223073 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.225079 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.225079 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.227086 [DEBUG] sofia_glue.c:2036 sip:10.11.0.20 Setting > proxy route to sofia/external/9995552000000 > 2010-08-11 20:07:51.228089 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [calling][0] > 2010-08-11 20:07:51.230097 [DEBUG] sofia.c:4317 Channel > sofia/external/9995552000000 entering state [terminated][503] > 2010-08-11 20:07:51.230097 [DEBUG] switch_channel.c:2309 > (sofia/external/9995552000000) Callstate Change RINGING -> HANGUP > 2010-08-11 20:07:51.230097 [NOTICE] sofia.c:4931 Hangup > sofia/external/9995552000000 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2010-08-11 20:07:51.230097 [DEBUG] switch_channel.c:2325 Send signal > sofia/external/9995552000000 [KILL] > 2010-08-11 20:07:51.230097 [DEBUG] switch_core_session.c:1039 Send signal > sofia/external/9995552000000 [BREAK] > 2010-08-11 20:07:51.230097 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/9995552000000) Running State Change CS_HANGUP > 2010-08-11 20:07:51.230097 [DEBUG] switch_core_state_machine.c:535 > (sofia/external/9995552000000) State HANGUP > 2010-08-11 20:07:51.230097 [DEBUG] mod_sofia.c:447 > sofia/external/9995552000000 Overriding SIP cause 503 with 503 from the > other leg > > > On Wed, Aug 11, 2010 at 2:15 PM, Jeff Lenk wrote: > >> >> Line 3998 of Sofia.c >> switch_channel_hangup(channel, >> SWITCH_CAUSE_REDIRECTION_TO_NEW_DESTINATION); >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p5413561.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/2ba2fef5/attachment-0001.html From pjintheusa at gmail.com Wed Aug 11 15:41:30 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 11 Aug 2010 18:41:30 -0400 Subject: [Freeswitch-users] Enterprise Originate and group_confirm Message-ID: Hi there, I have the following dialplan which uses enterprise originate and group_confirm: The plan is that two numbers are called simutanuously, each number has several gateway to use, incase a gateway fails. Every thing works great in terms of mutiple legs ringing, bridging, hanging up etc. I have one issue though: When either of the numbers, 6095553828 for example, answers the call but does not accept it (i.e. does not press 1) and hangs up, the dialplan tries the next gateway for the same number and therefore 6095553828 ends up getting another call even though he has rejected it. I was expecting that once the call is answered and rejected (not bridged) then that part of the dialplan would stop. Any help appreciated. Thanks Pj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/1922f6b2/attachment.html From chris.veazey at gmail.com Wed Aug 11 16:02:46 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Wed, 11 Aug 2010 18:02:46 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: References: <4508448D788749F6B25886D626C17C28@left><563B3EF6EAFE41BC8506DDA53B45F7F8@left><396D3026B6034431B93A6B3DC0803461@left><1281542398575-5412733.post@n2.nabble.com><55D75CA26DF8494B9FD88112BEAE8211@left><1281554134702-5413561.post@n2.nabble.com> Message-ID: <44682250FFBC45B8B8DBCA85D7D66125@left> I'd like Freeswitch to send a new Invite to the server IP listed in the maddr address in the 302 contact. Checking out the wiki to see what variable options are possible, _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Wednesday, August 11, 2010 4:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS processing 302 AFAIK 302 reenters the dialplan. If the destination number is unchanged, it will execute the same commands after the redirect as it did before the redirect, which would likely cause it to continously bridge, redirect, bridge, redirect.... That could explain what you're experiencing. How do you want to handle the 302? It's advisable to be careful what to do with a 302 which is why it reenters the dialplan. You can try dialing a $0.01 route but have it redirect you to a $1.00 route leaving you sending the call out on a route costing you $1.00 but charging your customer $0.01, or if you do bill them correctly leaving them with a bill of $1.00 when they expected only $0.01. It's possible to handle a 302 manually by sending them to a dialplan context specifically for it, which keeps the redirect logic away from your main dialplan. See http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect -Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/ff2d77da/attachment.html From kris at kriskinc.com Wed Aug 11 17:12:18 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 11 Aug 2010 20:12:18 -0400 Subject: [Freeswitch-users] FS processing 302 Message-ID: Like I said, it just worked for me. Fs parsed the new contact (with maddr) and re-sent the invite to the new contact. I'm on "vacation" so I won't be able to look at this but I will in the next couple of weeks if you're still having problems. -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces at lists.freeswitch.org < freeswitch-users-bounces at lists.freeswitch.org> *To*: 'FreeSWITCH Users Help' *Sent*: Wed Aug 11 19:02:46 2010 *Subject*: Re: [Freeswitch-users] FS processing 302 I?d like Freeswitch to send a new Invite to the server IP listed in the maddr address in the 302 contact. Checking out the wiki to see what variable options are possible, ------------------------------ *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre *Sent:* Wednesday, August 11, 2010 4:39 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] FS processing 302 AFAIK 302 reenters the dialplan. If the destination number is unchanged, it will execute the same commands after the redirect as it did before the redirect, which would likely cause it to continously bridge, redirect, bridge, redirect.... That could explain what you're experiencing. How do you want to handle the 302? It's advisable to be careful what to do with a 302 which is why it reenters the dialplan. You can try dialing a $0.01 route but have it redirect you to a $1.00 route leaving you sending the call out on a route costing you $1.00 but charging your customer $0.01, or if you do bill them correctly leaving them with a bill of $1.00 when they expected only $0.01. It's possible to handle a 302 manually by sending them to a dialplan context specifically for it, which keeps the redirect logic away from your main dialplan. See http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect -Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/27b6f110/attachment.html From macedoslm at gmail.com Wed Aug 11 18:33:28 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Wed, 11 Aug 2010 22:33:28 -0300 Subject: [Freeswitch-users] Increase the recording volume In-Reply-To: References: Message-ID: It's not working. I don't think this example is correct. Maybe it's something like: Any help? Thanks, -- Samuel Macedo On 1 August 2010 03:23, Tony Tin wrote: > Try this > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level > > Tony > > > On Sun, Aug 1, 2010 at 4:47 AM, Samuel Macedo wrote: > >> Hi, >> >> I'm having some problems with the recorded messages - I'm using the >> uuid_record command and ".wav" extension. >> I want to increase the volume of the files, they are so low. >> >> Is there anything that I can do? >> >> Thanks, >> -- >> Samuel Macedo >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/b75778c4/attachment-0001.html From xanlich at gmail.com Wed Aug 11 19:27:12 2010 From: xanlich at gmail.com (Chia-Yen Wu) Date: Thu, 12 Aug 2010 10:27:12 +0800 Subject: [Freeswitch-users] Sending Fax problem In-Reply-To: References: Message-ID: i am using WellGate 2644 voip gateway, it has 4 FXO, i can send fax by original way and it will automaticlly choose which FXO to dial, but if i want to dial by specific FXO, i must register the specific FXO as a user 2010/8/12 Michael Collins > > > On Wed, Aug 11, 2010 at 2:11 AM, Chia-Yen Wu wrote: > >> hello, >> >> i am trying to send fax via PSTN by VOIP gateway >> >> original way: >> originate sofia/gateway//12345678 &txfax(/path_to_fax_file) >> >> but i must set up my gateway as "Call a user as a gateway" >> > Why? > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best regards, Ted -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/1c5de3a0/attachment.html From macedoslm at gmail.com Wed Aug 11 19:35:18 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Wed, 11 Aug 2010 23:35:18 -0300 Subject: [Freeswitch-users] Increase the recording volume In-Reply-To: References: Message-ID: Sorry, it's working fine. This is a command: "set_audio_level". Thanks, -- Samuel Macedo On 11 August 2010 22:33, Samuel Macedo wrote: > It's not working. I don't think this example is correct. > > > > > Maybe it's something like: > > > > > Any help? > > Thanks, > -- > Samuel Macedo > > On 1 August 2010 03:23, Tony Tin wrote: > >> Try this >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level >> >> Tony >> >> >> On Sun, Aug 1, 2010 at 4:47 AM, Samuel Macedo wrote: >> >>> Hi, >>> >>> I'm having some problems with the recorded messages - I'm using the >>> uuid_record command and ".wav" extension. >>> I want to increase the volume of the files, they are so low. >>> >>> Is there anything that I can do? >>> >>> Thanks, >>> -- >>> Samuel Macedo >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/b9a8623b/attachment.html From mike at jerris.com Wed Aug 11 20:57:06 2010 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Aug 2010 23:57:06 -0400 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> <1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> <2F9511642CCD492CA057DB89F09BF420@MOBILEE1705> Message-ID: <7B40A1E5-2CBA-45DD-AEEC-5B225B22AC53@jerris.com> On Aug 10, 2010, at 6:11 PM, M. Ranganathan wrote: > On Mon, Aug 9, 2010 at 4:43 PM, Michael Collins wrote: >> >> >> On Mon, Aug 9, 2010 at 9:59 AM, Madovsky wrote: >>> >>> its' also an excellent way for big companies to avoid >>> to pay employees to make all the dev work also.... >> >> Indeed it is. Most of us in the OSS world say, "So what?" We've given our >> work away for "free" in return for other considerations: free advertising, >> free distribution (via Internet downloads), bragging rights, and growing a >> software-based ecosystem that allows us to tap into other revenue streams >> like private consulting or even writing a book. If big companies "take" our >> stuff and use it then they're growing the ecosystem. The choice of OSS >> licenses available to us gives us the necessary protection from large >> corporations hijacking our stuff. (This includes things like CC for >> documents, photos, sounds/music, etc.) > > Many big companies ( IBM for example ) pay employees to contribute to > open source. I am an employee of a company that has paid me to > contribute to open source. I have also given away many of my > contributions in the past and am very glad to report that they are > alive and well. That is a terrific feeling. On the other hand, if only > I had a penny for each closed source byte I have contributed to the > big bit bucket in the sky.... > > Further, OSS is really the ONLY way for little guys to compete. Linux > would have never become what it is if it were a bit company closed > source project. > > -- > M. Ranganathan Another perfect example of corporate contribution to open source would be FreeSWITCH itself. I am an employee of Barracuda Networks, and part of my job is to contribute to FreeSWITCH. This is also the case for some other members of the core FreeSWITCH team. We spend considerable time working on open source software, but also depend on the contributions of the community at large. Open source and corporate involvement seem to me to be a perfect match with winners on both sides. There will always be those people and companies that will try to take advantage of this without contributing back, but this is nearly impossible. There is inevitably bug reports and fixes that will come from these organizations when they have issues, or contracting revenue to some member of the community when they need help. We all contribute in some way. This is how open source ecosystems work, requiring both selfless personal and corporate involvement. FreeSWITCH was designed from the start in both architecture and licensing to attract the most of both personal and corporate use as a way to build a larger more vibrant community of developers building a wide range of solutions. Over the last year we have seen the release of Cudatel (), of a book (https://www.packtpub.com/), open source web user interfaces such as 2600hz project blue.box (http://www.2600hz.org) and FusionPBX (http://www.fusionpbx.com/). Features added this year such as T.38 support, Broadsoft SCA support, and sip high availability support would all not have happened without corporate support, and FreeTDM is now maintained by Sangoma. The addition and maintenance of many freeswitch modules are now supported entirely by individuals such as mod_vmd and mod_avmd contributed by Eric des Courtis and mod_limit/hash/db by Rupa Schomaker. I'd personally like to thank all members of the FreeSWITCH community for all your ongoing support. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100811/43826352/attachment.html From odermann at googlemail.com Thu Aug 12 01:01:09 2010 From: odermann at googlemail.com (Dennis) Date: Thu, 12 Aug 2010 10:01:09 +0200 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: it sounds as if there is a way to stream audio from fs over red5 to a flash-player. could someone explain how it is done? we do not want any telephony or any bi-directional things. we just need a simple one-way solution to listen to calls. in the moment we stream like this: fs -> icecast2 -> flash audio-player (which makes it a http stream) we would like to go another way: fs -> red5 -> flash audio-player (where the stream to the flash player ist rtmp) it would be great, if someone could explain how it works! thanks dennis From neilp at cs.stanford.edu Thu Aug 12 01:03:07 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 12 Aug 2010 13:33:07 +0530 Subject: [Freeswitch-users] CentOS 5.x and Python 2.6: segmentation fault importing ESL In-Reply-To: References: Message-ID: Hi Jo?o, I am still not able to get it to compile for 2.6. By 'path' I assume you mean whatever version /usr/bin/python points to? I tried aliasing that with 2.6, it still didn't work. Note that it's not recommended to replace 2.6 with 2.4 on CentOS, so I've installed it in parallel. How exactly do I modify python-config? I tried setting the first line to '#! /usr/bin/python2.6', but that didn't work. Also tried changed the 'pyver' variable to '2.6', but then I get a compilation error saying that lib is not found. Thanks, Neil 2010/8/11 Jo?o Mesquita > Neil, that's because ESL is being compiled against 2.4 and not 2.6. It will > use whatever is set on your path. > > You don't have to change anything on SWIG. You only have to change what is > being generated by the python-config script located in > ${SRC}/libs/esl/python > > Hope that helps. > > Jo?o Mesquita > > > On Tue, Aug 10, 2010 at 4:12 AM, Neil Patel wrote: > >> Hi All, >> >> I am using Python 2.6 on my CentOS installation, installed in parallel to >> the standard Python 2.4. I've compiled FS's python ESL module using 'make >> pymod', but the ESL python module causes a segfault when I try to import it >> in Python 2.6 (imports fine in 2.4). I'm guessing this is because the ESL >> module was compiled for Python 2.4. How do I change the ESL config/makefiles >> to compile for 2.6? I'm not familiar with SWIG, is there some configuration >> I need to change with it to use python2.6 executable? >> >> Thanks, >> Neil >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/844209fc/attachment-0001.html From petedao at gmail.com Thu Aug 12 01:11:13 2010 From: petedao at gmail.com (Pete Kay) Date: Thu, 12 Aug 2010 16:11:13 +0800 Subject: [Freeswitch-users] re-invite problem Message-ID: I am running a b2bua with freeswitch. It is fine until a Mitel UAS starts sending INVITE without sdp and ACK with sdp. Freeswitch seems to treat it as another dialog and sends it to dialplan handling. Within the dialplan, how can I recognize request as a re-invite and possibly ignore it? Does anyone know how to resolve this problem? From neilp at cs.stanford.edu Thu Aug 12 01:45:18 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 12 Aug 2010 14:15:18 +0530 Subject: [Freeswitch-users] outbound call with hunting number as the originating caller Message-ID: Hi All, I have an app running on FS making outbound calls over a PRI line. The calls all caller-id at the endpoints with the PRI's pilot number. Say I wanted to change it so that the endpoint receives the call from one of the block of hunting numbers that my line comes with (instead of the primary/pilot number), how do I originate a call to do that? Here's the command on fs_cli that I'm trying, that doesn't seem to work: > originate {destination_number=XXXXXXXXXX}openzap/smg_prid/a/YYYYYYYYYY at g2&echo Not sure which channel var(s) I should be setting, or even if that's the way to make this work. Maybe it's up to my PRI service provider to provide this functionality? In case it matters, I am hosting two PRIs on Sangoma hardware, here is my openzap.conf: [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 trunk_type =>e1 b-channel => 2:1-15 b-channel => 2:17-31 Thanks in advance, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/f405370a/attachment.html From ben at langfeld.co.uk Thu Aug 12 02:10:13 2010 From: ben at langfeld.co.uk (Ben Langfeld) Date: Thu, 12 Aug 2010 10:10:13 +0100 Subject: [Freeswitch-users] FreeSWITCH Book Has Arrived! In-Reply-To: References: <4C514B79.9050003@xpirio.com> <4C5420A1.80207@gmx.net> Message-ID: >From Packt: " We are sorry to know that the epub format of ?FreeSWITCH 1.0.6? is not available. We are looking into this issue. However there is some temporary problem with the epub format of this book and we are looking to resolve this quickly. Do let me know if you have any other query, I will be glad to help you. " Regards, Ben Langfeld On Wed, Aug 11, 2010 at 6:37 PM, Ben Langfeld wrote: > Does anyone know when the book will be available in ePub? It's difficult to > read on an iPhone without the ability to increase the font size. > > Regards, > Ben Langfeld > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/82522d56/attachment.html From dujinfang at gmail.com Thu Aug 12 03:36:05 2010 From: dujinfang at gmail.com (Seven Du) Date: Thu, 12 Aug 2010 18:36:05 +0800 Subject: [Freeswitch-users] Some build 32 bit binary on 64bit Mac Message-ID: I'm on Mac 10.6.4 64bit, a fresh clone of git head. I tried to build FS into 32 bit. I use the following command when configure: CFLAGS="-arch i386" CXXFLAGS="-arch i386" LDFLAGS="-arch i386" ./configure --prefix=/opt/fs-i386 --host=i386 --build=i386 --target=i386 However, when make, it throws some errors, I manually *change* the source like from "ifdef" to "ifndef" to trick it and make further, however, I still couldn't get fully built. some errors below. and ideas? thanks. src/switch_apr.c: In function 'switch_vasprintf': src/switch_apr.c:1024: warning: implicit declaration of function 'vasprintf' src/switch_core_session.c: In function 'switch_core_session_thread': src/switch_core_session.c:1193: warning: format '%d' expects type 'int', but argument 8 has type 'switch_size_t' src/switch_core_session.c:1203: warning: format '%d' expects type 'int', but argument 8 has type 'switch_size_t' src/switch_core_session.c:1221: warning: format '%d' expects type 'int', but argument 8 has type 'switch_size_t' src/switch_core.c: In function 'send_heartbeat': src/switch_core.c:84: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int' src/switch_core.c: In function 'change_user_group': src/switch_core.c:677: warning: implicit declaration of function 'setgroups' src/switch_core.c:704: warning: implicit declaration of function 'initgroups' src/switch_core.c: In function 'send_heartbeat': src/switch_core.c:84: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int' src/switch_core.c: In function 'change_user_group': src/switch_core.c:677: warning: implicit declaration of function 'setgroups' src/switch_core.c:704: warning: implicit declaration of function 'initgroups' src/switch_rtp.c: In function 'rtp_common_write': src/switch_rtp.c:3437: error: 'u_long' undeclared (first use in this function) src/switch_rtp.c:3437: error: (Each undeclared identifier is reported only once src/switch_rtp.c:3437: error: for each function it appears in.) src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 src/switch_utils.c: In function 'switch_build_uri': src/switch_utils.c:1530: error: 'NI_MAXHOST' undeclared (first use in this function) src/switch_utils.c:1530: error: (Each undeclared identifier is reported only once src/switch_utils.c:1530: error: for each function it appears in.) src/switch_utils.c:1530: error: 'NI_MAXSERV' undeclared (first use in this function) more detailed log: http://pastebin.freeswitch.org/13615 -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From steveayre at gmail.com Thu Aug 12 04:33:29 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 12 Aug 2010 12:33:29 +0100 Subject: [Freeswitch-users] Some build 32 bit binary on 64bit Mac In-Reply-To: References: Message-ID: If you're on 64bit, why do you want to compile as 32bit? -Steve On 12 August 2010 11:36, Seven Du wrote: > I'm on Mac 10.6.4 64bit, a fresh clone of git head. I tried to build > FS into 32 bit. > > I use the following command when configure: > > CFLAGS="-arch i386" CXXFLAGS="-arch i386" LDFLAGS="-arch i386" > ./configure --prefix=/opt/fs-i386 --host=i386 --build=i386 > --target=i386 > > > However, when make, it throws some errors, I manually *change* the > source like from "ifdef" to "ifndef" to trick it and make further, > however, I still couldn't get fully built. > > some errors below. and ideas? thanks. > > > > src/switch_apr.c: In function 'switch_vasprintf': > src/switch_apr.c:1024: warning: implicit declaration of function > 'vasprintf' > > > > src/switch_core_session.c: In function 'switch_core_session_thread': > src/switch_core_session.c:1193: warning: format '%d' expects type > 'int', but argument 8 has type 'switch_size_t' > src/switch_core_session.c:1203: warning: format '%d' expects type > 'int', but argument 8 has type 'switch_size_t' > src/switch_core_session.c:1221: warning: format '%d' expects type > 'int', but argument 8 has type 'switch_size_t' > > > src/switch_core.c: In function 'send_heartbeat': > src/switch_core.c:84: warning: format '%d' expects type 'int', but > argument 5 has type 'long unsigned int' > src/switch_core.c: In function 'change_user_group': > src/switch_core.c:677: warning: implicit declaration of function > 'setgroups' > src/switch_core.c:704: warning: implicit declaration of function > 'initgroups' > > > > > src/switch_core.c: In function 'send_heartbeat': > src/switch_core.c:84: warning: format '%d' expects type 'int', but > argument 5 has type 'long unsigned int' > src/switch_core.c: In function 'change_user_group': > src/switch_core.c:677: warning: implicit declaration of function > 'setgroups' > src/switch_core.c:704: warning: implicit declaration of function > 'initgroups' > > > > > > src/switch_rtp.c: In function 'rtp_common_write': > src/switch_rtp.c:3437: error: 'u_long' undeclared (first use in this > function) > src/switch_rtp.c:3437: error: (Each undeclared identifier is reported only > once > src/switch_rtp.c:3437: error: for each function it appears in.) > src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' > src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' > src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' > src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' > src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' > src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' > make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 > > > > > > src/switch_utils.c: In function 'switch_build_uri': > src/switch_utils.c:1530: error: 'NI_MAXHOST' undeclared (first use in > this function) > src/switch_utils.c:1530: error: (Each undeclared identifier is > reported only once > src/switch_utils.c:1530: error: for each function it appears in.) > src/switch_utils.c:1530: error: 'NI_MAXSERV' undeclared (first use in > this function) > > > > more detailed log: > > http://pastebin.freeswitch.org/13615 > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/dae0f81b/attachment.html From dujinfang at gmail.com Thu Aug 12 05:20:22 2010 From: dujinfang at gmail.com (Seven Du) Date: Thu, 12 Aug 2010 20:20:22 +0800 Subject: [Freeswitch-users] Some build 32 bit binary on 64bit Mac In-Reply-To: References: Message-ID: There are many reasons I need a 32bit lib: 1) 64bit app cannot load flash in webkit in QT(at least in my case). I'm try to build a client with libfreeswitch and flash in. 2) I want the lib run in iPhone simulator. Eventually on ARM on iphone. 3) If you release prebuilt desktop binary, you'd better don't forget 32 bit users. 4) perhaps more... On Thu, Aug 12, 2010 at 7:33 PM, Steven Ayre wrote: > If you're on 64bit, why do you want to compile as 32bit? > > -Steve > > > > On 12 August 2010 11:36, Seven Du wrote: >> >> I'm on Mac 10.6.4 64bit, a fresh clone of git head. I tried to build >> FS into 32 bit. >> >> I use the following command when configure: >> >> CFLAGS="-arch i386" CXXFLAGS="-arch i386" LDFLAGS="-arch i386" >> ./configure --prefix=/opt/fs-i386 --host=i386 --build=i386 >> --target=i386 >> >> >> However, when make, it throws some errors, I manually *change* the >> source like from "ifdef" to "ifndef" to trick it and make further, >> however, I still couldn't get fully built. >> >> some errors below. and ideas? thanks. >> >> >> >> src/switch_apr.c: In function 'switch_vasprintf': >> src/switch_apr.c:1024: warning: implicit declaration of function >> 'vasprintf' >> >> >> >> src/switch_core_session.c: In function 'switch_core_session_thread': >> src/switch_core_session.c:1193: warning: format '%d' expects type >> 'int', but argument 8 has type 'switch_size_t' >> src/switch_core_session.c:1203: warning: format '%d' expects type >> 'int', but argument 8 has type 'switch_size_t' >> src/switch_core_session.c:1221: warning: format '%d' expects type >> 'int', but argument 8 has type 'switch_size_t' >> >> >> src/switch_core.c: In function 'send_heartbeat': >> src/switch_core.c:84: warning: format '%d' expects type 'int', but >> argument 5 has type 'long unsigned int' >> src/switch_core.c: In function 'change_user_group': >> src/switch_core.c:677: warning: implicit declaration of function >> 'setgroups' >> src/switch_core.c:704: warning: implicit declaration of function >> 'initgroups' >> >> >> >> >> src/switch_core.c: In function 'send_heartbeat': >> src/switch_core.c:84: warning: format '%d' expects type 'int', but >> argument 5 has type 'long unsigned int' >> src/switch_core.c: In function 'change_user_group': >> src/switch_core.c:677: warning: implicit declaration of function >> 'setgroups' >> src/switch_core.c:704: warning: implicit declaration of function >> 'initgroups' >> >> >> >> >> >> src/switch_rtp.c: In function 'rtp_common_write': >> src/switch_rtp.c:3437: error: 'u_long' undeclared (first use in this >> function) >> src/switch_rtp.c:3437: error: (Each undeclared identifier is reported only >> once >> src/switch_rtp.c:3437: error: for each function it appears in.) >> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >> make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >> >> >> >> >> >> src/switch_utils.c: In function 'switch_build_uri': >> src/switch_utils.c:1530: error: 'NI_MAXHOST' undeclared (first use in >> this function) >> src/switch_utils.c:1530: error: (Each undeclared identifier is >> reported only once >> src/switch_utils.c:1530: error: for each function it appears in.) >> src/switch_utils.c:1530: error: 'NI_MAXSERV' undeclared (first use in >> this function) >> >> >> >> more detailed log: >> >> http://pastebin.freeswitch.org/13615 >> >> >> >> -- >> Blog: http://www.dujinfang.com >> Proj:? http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From babak.freeswitch at gmail.com Thu Aug 12 05:47:20 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Thu, 12 Aug 2010 17:17:20 +0430 Subject: [Freeswitch-users] g729 Message-ID: Hi How many licenses is needed to be able to record 10 concurrent calls between endpoints which all support g729? thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/93d3984b/attachment.html From brian at freeswitch.org Thu Aug 12 05:58:50 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Aug 2010 07:58:50 -0500 Subject: [Freeswitch-users] g729 In-Reply-To: References: Message-ID: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> 20. /b On Aug 12, 2010, at 7:47 AM, babak yakhchali wrote: > Hi > How many licenses is needed to be able to record 10 concurrent calls between endpoints which all support g729? > thanx From brian at freeswitch.org Thu Aug 12 06:00:02 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Aug 2010 08:00:02 -0500 Subject: [Freeswitch-users] re-invite problem In-Reply-To: References: Message-ID: <5693B149-67BD-4675-8254-8DD963E33B42@freeswitch.org> Chances are it is in fact a new dialog have you double checked the to/from tags and callid? Also if its without an SDP you would have to enable 3pcc on the profile to accept it possibly. /b On Aug 12, 2010, at 3:11 AM, Pete Kay wrote: > I am running a b2bua with freeswitch. It is fine until a Mitel UAS > starts sending INVITE without sdp and ACK with sdp. Freeswitch seems > to treat it as another dialog and sends it to dialplan handling. > > Within the dialplan, how can I recognize request as a re-invite and > possibly ignore it? > > Does anyone know how to resolve this problem? From petedao at gmail.com Thu Aug 12 06:25:48 2010 From: petedao at gmail.com (Pete Kay) Date: Thu, 12 Aug 2010 21:25:48 +0800 Subject: [Freeswitch-users] re-invite problem In-Reply-To: <5693B149-67BD-4675-8254-8DD963E33B42@freeswitch.org> References: <5693B149-67BD-4675-8254-8DD963E33B42@freeswitch.org> Message-ID: Hi, I check tags and callid. It is the same dialog. Also, the invite is accepted and I can see UAC does respond 200 OK . The freeswitch sends out BYE after ACK. The problem I am seeing is that the re-invite triggers the dialplan execution which based on its logic is responding with a 488 within the dialplan using the respond app. When freeswitch receives the 488, it can't recognize the dialog so it sends out BYE. Therefore, I think the way to solve this is to configure sofia so that the invite won't trigger the execution of dialplan. Is there anyway to do that? Thanks, P On Thu, Aug 12, 2010 at 9:00 PM, Brian West wrote: > Chances are it is in fact a new dialog have you double checked the to/from tags and callid? ?Also if its without an SDP you would have to enable 3pcc on the profile to accept it possibly. > > /b > > On Aug 12, 2010, at 3:11 AM, Pete Kay wrote: > >> I am running a b2bua with freeswitch. ?It is fine until a Mitel UAS >> starts sending INVITE without sdp and ACK with sdp. Freeswitch seems >> to treat it as another dialog and sends it to dialplan handling. >> >> Within the dialplan, how can I recognize request as a re-invite and >> possibly ignore it? >> >> Does anyone know how to resolve this problem? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From damjan at ecntelecoms.com Thu Aug 12 07:13:44 2010 From: damjan at ecntelecoms.com (Damjan Jovanovic) Date: Thu, 12 Aug 2010 16:13:44 +0200 Subject: [Freeswitch-users] [contrib] a new java ESL inbound/outbound client In-Reply-To: <74a861001001050541v4e02f487xd21fe8fc13e8ed0a@mail.gmail.com> References: <74a861001001050541v4e02f487xd21fe8fc13e8ed0a@mail.gmail.com> Message-ID: <1281622424.3347.334.camel@damjan-laptop> Hi David How do you compile this? I type "mvn clean install" and get errors: [INFO] Scanning for projects... [INFO] ------------------------------------------------------------------------ [ERROR] FATAL ERROR [INFO] ------------------------------------------------------------------------ [INFO] Error building POM (may not be this project's POM). Project ID: null:org.freeswitch.esl.client:bundle:null Reason: Cannot find parent: org.freeswitch.esl.client:esl-client-parent for project: null:org.freeswitch.esl.client:bundle:null for project null:org.freeswitch.esl.client:bundle:null [INFO] ------------------------------------------------------------------------ [INFO] Trace org.apache.maven.reactor.MavenExecutionException: Cannot find parent: org.freeswitch.esl.client:esl-client-parent for project: null:org.freeswitch.esl.client:bundle:null for project null:org.freeswitch.esl.client:bundle:null etc Damjan On Wed, 2010-01-06 at 00:41 +1100, david varnes wrote: > Hi all, > I had a basic java inbound ESL client kicking around that I have used > in a couple of small projects over the last year. I needed something > a little more complete for a new project so I dusted it off and made it > less incomplete. > > It still needs more work and testing, but it certainly is usable right now. > I have tested it against FS 1.0.4 and latest trunk. > > I have put it in my contrib area in svn in hopes that some may find it > useful: > http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-client > I would be interested in any feedback ... > > Features > * Apache License (ASL) version 2 > * Standalone Inbound client > * Framework classes to easily create an Outbound socket client > * based on Netty [1] nio library version 3.1.5.GA (previously was > using Apache MINA, but this is easier) > * logging via slf4j > * only dependencies are slf4j-api and netty (both Apache licensed) > * single jar which is a valid OSGi bundle > * built using maven > * eclipse projects > * reasonable level of java docs > > Still todo > * Docs > * Simple example apps > * .. more in TODO.txt in project root. > > There is no binary jar available right now since I don't know how/if > I can put files up to file.freeswitch.org. > > In the meantime to build you need maven [2] installed. If you are > unfamiliar with maven usage, I can post a simple howto. > > davidv > > [1] http://www.jboss.org/netty/downloads.html > [2] http://maven.apache.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/fd9c70ab/attachment.html From fdelawarde at wirelessmundi.com Thu Aug 12 02:25:24 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Thu, 12 Aug 2010 11:25:24 +0200 Subject: [Freeswitch-users] use of multiple templates and output files with mod_cdr_csv Message-ID: <1281605124.28815.239.camel@luna.tc.commsmundi.com> Hello folks, I would be interested in using mod_cdr_csv to log CDRs in multiple files using different templates for each file. Is this currently possible? It would be used for example to export to more than one database tables having different fields and used for different purposes (billing, debug, ...). Thank you, Fran?ois. From ovvenkatesan at gmail.com Thu Aug 12 06:49:29 2010 From: ovvenkatesan at gmail.com (ovvenkat) Date: Thu, 12 Aug 2010 19:19:29 +0530 Subject: [Freeswitch-users] how to use "nuance ASR" Message-ID: Hi. I wanted to incorporate nuance ASR with freeSwitch. Any one guide me from where I have to start my journey ? -- Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/8ffaf000/attachment.html From jan.berger at video24.no Thu Aug 12 08:18:22 2010 From: jan.berger at video24.no (Jan Berger) Date: Thu, 12 Aug 2010 17:18:22 +0200 Subject: [Freeswitch-users] how to use "nuance ASR" In-Reply-To: References: Message-ID: <7670E6D3205F482190949DDDBAAF45AD@dell9400> Read about MRCP on the wiki. Nuance support MRCP and so does FreeSWITCH _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ovvenkat Sent: 12. august 2010 15:49 To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to use "nuance ASR" Hi. I wanted to incorporate nuance ASR with freeSwitch. Any one guide me from where I have to start my journey ? -- Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/da1d3500/attachment-0001.html From steveayre at gmail.com Thu Aug 12 08:21:02 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 12 Aug 2010 16:21:02 +0100 Subject: [Freeswitch-users] how to use "nuance ASR" In-Reply-To: References: Message-ID: Use mod_unimrcp to communicate with nuance via MRCP. Module documentation (includes example configs for nuance): http://wiki.freeswitch.org/wiki/Mod_unimrcp -Steve On 12 August 2010 14:49, ovvenkat wrote: > Hi. > > I wanted to incorporate nuance ASR with freeSwitch. Any one guide me from > where I have to start my journey ? > > -- > > Regards > Venkatesan OV. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/06168f3d/attachment.html From damjan at ecntelecoms.com Thu Aug 12 08:24:33 2010 From: damjan at ecntelecoms.com (Damjan Jovanovic) Date: Thu, 12 Aug 2010 17:24:33 +0200 Subject: [Freeswitch-users] [contrib] a new java ESL inbound/outbound client In-Reply-To: <1281622424.3347.334.camel@damjan-laptop> References: <74a861001001050541v4e02f487xd21fe8fc13e8ed0a@mail.gmail.com> <1281622424.3347.334.camel@damjan-laptop> Message-ID: <1281626673.3347.341.camel@damjan-laptop> Below is the patch I had to use to get it to compile and run. By the way, your ESL library rules. I tried fseslib (http://versafon.com/versafonweb/Software.jsp) but it's GPL(v3!) and doesn't handle event bodies; even with a hack to it still doesn't provide essential fields. The ESL library for Java built into Freeswitch uses native code and seems to only handle outbound. Only your library provides everything, is 100% Java, and is licensed nicely. Thank you so much Damjan diff --git a/dvarnes/java/esl-client/org.freeswitch.esl.client.example/pom.xml b/dvarnes/java/esl-client/org.freeswitch.esl.client.example/pom.xml index 56c8250..97f03db 100644 --- a/dvarnes/java/esl-client/org.freeswitch.esl.client.example/pom.xml +++ b/dvarnes/java/esl-client/org.freeswitch.esl.client.example/pom.xml @@ -19,7 +19,7 @@ org.freeswitch.esl.client esl-client-parent - 0.0.1-SNAPSHOT + 0.9.0-SNAPSHOT org.freeswitch.esl.client.example FreeSWITCH Event Socket Library - Java Client example usage @@ -27,7 +27,7 @@ org.freeswitch.esl.client org.freeswitch.esl.client - 0.0.1-SNAPSHOT + 0.9.0-SNAPSHOT - \ No newline at end of file + diff --git a/dvarnes/java/esl-client/org.freeswitch.esl.client/pom.xml b/dvarnes/java/esl-client/org.freeswitch.esl.client/pom.xml index cc9d9f1..0093650 100644 --- a/dvarnes/java/esl-client/org.freeswitch.esl.client/pom.xml +++ b/dvarnes/java/esl-client/org.freeswitch.esl.client/pom.xml @@ -19,7 +19,7 @@ org.freeswitch.esl.client esl-client-parent - 0.0.1-SNAPSHOT + 0.9.0-SNAPSHOT org.freeswitch.esl.client FreeSWITCH Event Socket Library - Java Client @@ -43,13 +43,17 @@ * org.freeswitch.esl.client, + org.freeswitch.esl.client.inbound, + org.freeswitch.esl.client.internal, + org.freeswitch.esl.client.internal.debug, + org.freeswitch.esl.client.debug, + org.freeswitch.esl.client.outbound, + org.freeswitch.esl.client.transport, org.freeswitch.esl.client.transport.event, org.freeswitch.esl.client.transport.message, - org.freeswitch.esl.client.internal, - org.freeswitch.esl.client.debug @@ -111,4 +115,4 @@ - \ No newline at end of file + On Thu, 2010-08-12 at 16:13 +0200, Damjan Jovanovic wrote: > Hi David > > How do you compile this? I type "mvn clean install" and get errors: > > [INFO] Scanning for projects... > [INFO] > ------------------------------------------------------------------------ > [ERROR] FATAL ERROR > [INFO] > ------------------------------------------------------------------------ > [INFO] Error building POM (may not be this project's POM). > > > Project ID: null:org.freeswitch.esl.client:bundle:null > > Reason: Cannot find parent: > org.freeswitch.esl.client:esl-client-parent for project: > null:org.freeswitch.esl.client:bundle:null for project > null:org.freeswitch.esl.client:bundle:null > > > [INFO] > ------------------------------------------------------------------------ > [INFO] Trace > org.apache.maven.reactor.MavenExecutionException: Cannot find parent: > org.freeswitch.esl.client:esl-client-parent for project: > null:org.freeswitch.esl.client:bundle:null for project > null:org.freeswitch.esl.client:bundle:null > > etc > > Damjan > > On Wed, 2010-01-06 at 00:41 +1100, david varnes wrote: > > > Hi all, > > I had a basic java inbound ESL client kicking around that I have used > > in a couple of small projects over the last year. I needed something > > a little more complete for a new project so I dusted it off and made it > > less incomplete. > > > > It still needs more work and testing, but it certainly is usable right now. > > I have tested it against FS 1.0.4 and latest trunk. > > > > I have put it in my contrib area in svn in hopes that some may find it > > useful: > > http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-client > > I would be interested in any feedback ... > > > > Features > > * Apache License (ASL) version 2 > > * Standalone Inbound client > > * Framework classes to easily create an Outbound socket client > > * based on Netty [1] nio library version 3.1.5.GA (previously was > > using Apache MINA, but this is easier) > > * logging via slf4j > > * only dependencies are slf4j-api and netty (both Apache licensed) > > * single jar which is a valid OSGi bundle > > * built using maven > > * eclipse projects > > * reasonable level of java docs > > > > Still todo > > * Docs > > * Simple example apps > > * .. more in TODO.txt in project root. > > > > There is no binary jar available right now since I don't know how/if > > I can put files up to file.freeswitch.org. > > > > In the meantime to build you need maven [2] installed. If you are > > unfamiliar with maven usage, I can post a simple howto. > > > > davidv > > > > [1] http://www.jboss.org/netty/downloads.html > > [2] http://maven.apache.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/37c35abd/attachment.html From brian at freeswitch.org Thu Aug 12 08:28:15 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Aug 2010 10:28:15 -0500 Subject: [Freeswitch-users] [contrib] a new java ESL inbound/outbound client In-Reply-To: <1281626673.3347.341.camel@damjan-laptop> References: <74a861001001050541v4e02f487xd21fe8fc13e8ed0a@mail.gmail.com> <1281622424.3347.334.camel@damjan-laptop> <1281626673.3347.341.camel@damjan-laptop> Message-ID: <5472A339-5B7F-46DF-B9F0-16C52CF8BE5B@freeswitch.org> Can you put this on jira please? Thanks, Brian On Aug 12, 2010, at 10:24 AM, Damjan Jovanovic wrote: > Below is the patch I had to use to get it to compile and run. > > By the way, your ESL library rules. I tried fseslib (http://versafon.com/versafonweb/Software.jsp) but it's GPL(v3!) and doesn't handle event bodies; even with a hack to it still doesn't provide essential fields. The ESL library for Java built into Freeswitch uses native code and seems to only handle outbound. Only your library provides everything, is 100% Java, and is licensed nicely. > > Thank you so much > Damjan From jan.berger at video24.no Thu Aug 12 09:09:07 2010 From: jan.berger at video24.no (Jan Berger) Date: Thu, 12 Aug 2010 18:09:07 +0200 Subject: [Freeswitch-users] GPL Wins Again In-Reply-To: <7B40A1E5-2CBA-45DD-AEEC-5B225B22AC53@jerris.com> References: <1D3161664B5A45C7A286AD9225E714B0@MOBILEE1705> <1235E016-2F82-4FC3-BFC5-039E9881F521@freeswitch.org> <42F9C5089CC941D2BF6AD2302724902D@MOBILEE1705> <2F9511642CCD492CA057DB89F09BF420@MOBILEE1705> <7B40A1E5-2CBA-45DD-AEEC-5B225B22AC53@jerris.com> Message-ID: <665A76D6CD674678A8A44B77081B6CD9@dell9400> Many large companies do that. They use OSS and can't contribute OSS back, but are happy with some employees using parts of their time here maintaining the bits they use. Glad it worked out for you Mike. Another perfect example of corporate contribution to open source would be FreeSWITCH itself. I am an employee of Barracuda Networks, and part of my job is to contribute to FreeSWITCH. This is also the case for some other members of the core FreeSWITCH team. We spend considerable time working on open source software, but also depend on the contributions of the community at large. Open source and corporate involvement seem to me to be a perfect match with winners on both sides. There will always be those people and companies that will try to take advantage of this without contributing back, but this is nearly impossible. There is inevitably bug reports and fixes that will come from these organizations when they have issues, or contracting revenue to some member of the community when they need help. We all contribute in some way. This is how open source ecosystems work, requiring both selfless personal and corporate involvement. FreeSWITCH was designed from the start in both architecture and licensing to attract the most of both personal and corporate use as a way to build a larger more vibrant community of developers building a wide range of solutions. Over the last year we have seen the release of Cudatel (), of a book (https://www.packtpub.com/), open source web user interfaces such as 2600hz project blue.box (http://www.2600hz.org) and FusionPBX (http://www.fusionpbx.com/). Features added this year such as T.38 support, Broadsoft SCA support, and sip high availability support would all not have happened without corporate support, and FreeTDM is now maintained by Sangoma. The addition and maintenance of many freeswitch modules are now supported entirely by individuals such as mod_vmd and mod_avmd contributed by Eric des Courtis and mod_limit/hash/db by Rupa Schomaker. I'd personally like to thank all members of the FreeSWITCH community for all your ongoing support. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/51f018d9/attachment-0001.html From odermann at googlemail.com Thu Aug 12 10:02:22 2010 From: odermann at googlemail.com (Dennis) Date: Thu, 12 Aug 2010 19:02:22 +0200 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... Message-ID: hi there, at the moment we have a serious and urgent problem with transmitting dtmf-inputs in the right way from one endpoint through fs to another endpoint. we are not very shure, if the problem is on our or on the carriers side. digits send through fs from one side to another are not received correctly. is there someone who could help us (quite quickly)? we are receiving the dtmf-signals on fs as RFC-2833. our fs-setting are the following: - we are receiving a call and we connect to another side through fs. - the caller enters the following dtmf-digits: 987654321. - fs receives the dtmf-inputs absolutely correctly !!! - the target receives the following dtmf-digits: 98877665543321 it is important to say, that the call comes over the "normal phone-line" (or what ever a phonecall is called, which does not come over voip - landline?). per VOIP (cirpack) our fs-servers receive this call/dtmf-inputs and then we send the call to the target over the "normal phone-line". does anybody have a clue where our problems might be? what could we do to find out more about this probleme? where could we debug? thanks a lot dennis From gmaruzz at celliax.org Thu Aug 12 10:29:01 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 12 Aug 2010 19:29:01 +0200 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: Message-ID: On Thu, Aug 12, 2010 at 7:02 PM, Dennis wrote: > hi there, > > at the moment we have a serious and urgent problem with transmitting While you probably will receive a good answer here on the mailing list, if your problem is urgent (and maybe will require exchanges of questions and answers), I would counsel you to join the #freeswitch IRC channel in the irc.freenode.net IRC server, and pose your question again there, so you'll have a quick interaction with the "gurus". -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From steveayre at gmail.com Thu Aug 12 11:05:47 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 12 Aug 2010 19:05:47 +0100 Subject: [Freeswitch-users] re-invite problem In-Reply-To: References: <5693B149-67BD-4675-8254-8DD963E33B42@freeswitch.org> Message-ID: Do you have a packet trace available? -Steve On 12 August 2010 14:25, Pete Kay wrote: > Hi, > > I check tags and callid. It is the same dialog. Also, the invite is > accepted and I can see UAC does respond 200 OK . The freeswitch sends > out BYE after ACK. > > The problem I am seeing is that the re-invite triggers the dialplan > execution which based on its logic is responding with a 488 within the > dialplan using the respond app. When freeswitch receives the 488, it > can't recognize the dialog so it sends out BYE. > > Therefore, I think the way to solve this is to configure sofia so that > the invite won't trigger the execution of dialplan. > > Is there anyway to do that? > > Thanks, > P > > On Thu, Aug 12, 2010 at 9:00 PM, Brian West wrote: > > Chances are it is in fact a new dialog have you double checked the > to/from tags and callid? Also if its without an SDP you would have to > enable 3pcc on the profile to accept it possibly. > > > > /b > > > > On Aug 12, 2010, at 3:11 AM, Pete Kay wrote: > > > >> I am running a b2bua with freeswitch. It is fine until a Mitel UAS > >> starts sending INVITE without sdp and ACK with sdp. Freeswitch seems > >> to treat it as another dialog and sends it to dialplan handling. > >> > >> Within the dialplan, how can I recognize request as a re-invite and > >> possibly ignore it? > >> > >> Does anyone know how to resolve this problem? > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/143f5243/attachment.html From msc at freeswitch.org Thu Aug 12 11:10:25 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Aug 2010 11:10:25 -0700 Subject: [Freeswitch-users] Help needed: review FS book Message-ID: Hello all! The publishers of the new "Bridge Book" are looking for FS devs/users who are in a position to write a review. If you would like to write a review please contact me off list. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/e1b2b827/attachment.html From msc at freeswitch.org Thu Aug 12 11:56:13 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Aug 2010 11:56:13 -0700 Subject: [Freeswitch-users] use of multiple templates and output files with mod_cdr_csv In-Reply-To: <1281605124.28815.239.camel@luna.tc.commsmundi.com> References: <1281605124.28815.239.camel@luna.tc.commsmundi.com> Message-ID: On Thu, Aug 12, 2010 at 2:25 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > Hello folks, > > I would be interested in using mod_cdr_csv to log CDRs in multiple files > using different templates for each file. Is this currently possible? > > Look at the accountcode channel variable: http://wiki.freeswitch.org/wiki/Mod_cdr_csv#accountcode -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/7d91b666/attachment.html From msc at freeswitch.org Thu Aug 12 12:00:54 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Aug 2010 12:00:54 -0700 Subject: [Freeswitch-users] outbound call with hunting number as the originating caller In-Reply-To: References: Message-ID: On Thu, Aug 12, 2010 at 1:45 AM, Neil Patel wrote: > Hi All, > > I have an app running on FS making outbound calls over a PRI line. The > calls all caller-id at the endpoints with the PRI's pilot number. Say I > wanted to change it so that the endpoint receives the call from one of the > block of hunting numbers that my line comes with (instead of the > primary/pilot number), how do I originate a call to do that? Here's the > command on fs_cli that I'm trying, that doesn't seem to work: > > > originate {destination_number=XXXXXXXXXX}openzap/smg_prid/a/YYYYYYYYYY at g2&echo > > Neil, use "origination_caller_id_number=xxx" like this: > originate {origination_caller_id_number=1234567890,destination_number=XXXXXXXXXX}openzap/smg_prid/a/YYYYYYYYYY at g2&echo NOTE: The telco must support custom sending of "calling party number" sometimes called CPN. It won't hurt to try it, though, so give it a whirl and let us know. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/ecd25002/attachment.html From helmut.kuper at ewetel.de Thu Aug 12 12:07:45 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Aug 2010 21:07:45 +0200 Subject: [Freeswitch-users] Question about originate Message-ID: <4C644681.1050703@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello there, I try to originate a new call in "event socket outbound" mode in an external application. I would like to get originate's result code, but I only got "+OK uuid" on answer, dnd, user busy, cancel ... or "NO_ANSWER" on timeout rather then "+OK uuid" for answer and "-ERR Reason" for everything else. So how can I get the result code of the originate process? In FS console I can see the wanted result codes from target (e.g. "mod_dptools.c:2355 Originate Failed. Cause: USER_BUSY"). best regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMZEaB4tZeNddg3dwRAiLXAJ9wS0pEIe3pagDADOXcx/tNlkBr+QCdEL/S P0OhzYqbWJ54At6ENrcD1q0= =XeB+ -----END PGP SIGNATURE----- From david.ponzone at ipeva.fr Thu Aug 12 12:18:21 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 12 Aug 2010 21:18:21 +0200 Subject: [Freeswitch-users] use of multiple templates and output files with mod_cdr_csv In-Reply-To: References: <1281605124.28815.239.camel@luna.tc.commsmundi.com> Message-ID: This will allow to log to 2 CDR files (Master.csv + one specific to accountcode), no more. Correct me if I am wrong. Fran?ois, what I would do is to log all the fields you need to Master.csv, and then do selective imports of the fields you require depending on the usage. mysqlimport (LOAD DATA INFILE) allows that if you put a dummy variable (@dummy) instead of a column name. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 12/08/2010 ? 20:56, Michael Collins a ?crit : > > On Thu, Aug 12, 2010 at 2:25 AM, Fran?ois Delawarde > wrote: > Hello folks, > > I would be interested in using mod_cdr_csv to log CDRs in multiple > files > using different templates for each file. Is this currently possible? > > Look at the accountcode channel variable: > http://wiki.freeswitch.org/wiki/Mod_cdr_csv#accountcode > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/55b85514/attachment-0001.html From msc at freeswitch.org Thu Aug 12 12:25:13 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Aug 2010 12:25:13 -0700 Subject: [Freeswitch-users] Question about originate In-Reply-To: <4C644681.1050703@ewetel.de> References: <4C644681.1050703@ewetel.de> Message-ID: What is your originate command? Are you ignoring early media? Just curious. -MC On Thu, Aug 12, 2010 at 12:07 PM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello there, > > I try to originate a new call in "event socket outbound" mode in an > external application. I would like to get originate's result code, but I > only got > > "+OK uuid" on answer, dnd, user busy, cancel ... > > or > > "NO_ANSWER" on timeout > > > rather then "+OK uuid" for answer and "-ERR Reason" for everything else. > > So how can I get the result code of the originate process? In FS console > I can see the wanted result codes from target (e.g. "mod_dptools.c:2355 > Originate Failed. Cause: USER_BUSY"). > > > best regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFMZEaB4tZeNddg3dwRAiLXAJ9wS0pEIe3pagDADOXcx/tNlkBr+QCdEL/S > P0OhzYqbWJ54At6ENrcD1q0= > =XeB+ > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/cb95a782/attachment.html From msc at freeswitch.org Thu Aug 12 12:29:43 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Aug 2010 12:29:43 -0700 Subject: [Freeswitch-users] use of multiple templates and output files with mod_cdr_csv In-Reply-To: References: <1281605124.28815.239.camel@luna.tc.commsmundi.com> Message-ID: On Thu, Aug 12, 2010 at 12:18 PM, David Ponzone wrote: > This will allow to log to 2 CDR files (Master.csv + one specific to > accountcode), no more. > Correct me if I am wrong. > Yes, sort of. It does log to Master.csv and ${accountcode}.csv However, if you have a template in cdr_csv.conf.xml with the same name as the value in accountcode then it will use that template when creating the ${accountcode}.csv file. HTH, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/f278378f/attachment.html From Nabble at slickdeals.endjunk.com Thu Aug 12 12:43:35 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Thu, 12 Aug 2010 12:43:35 -0700 (PDT) Subject: [Freeswitch-users] Help needed: review FS book In-Reply-To: References: Message-ID: <1281642215312-5417580.post@n2.nabble.com> I may not be qualified to review any books. However, I just wanted to say that at least more and more books about FS are coming soon and that is certainly a good news for FS community! ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-needed-review-FS-book-tp5417294p5417580.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Thu Aug 12 12:46:43 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 12 Aug 2010 20:46:43 +0100 Subject: [Freeswitch-users] Question about originate In-Reply-To: References: <4C644681.1050703@ewetel.de> Message-ID: Subscribe to events http://wiki.freeswitch.org/wiki/Mod_event_socket#event http://wiki.freeswitch.org/wiki/Event_list You'll then see an event for ringing, answer, hangup, failed etc. Those events will give you the information you need -Steve On 12 August 2010 20:25, Michael Collins wrote: > What is your originate command? Are you ignoring early media? Just curious. > > -MC > > > On Thu, Aug 12, 2010 at 12:07 PM, Helmut Kuper wrote: > >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Hello there, >> >> I try to originate a new call in "event socket outbound" mode in an >> external application. I would like to get originate's result code, but I >> only got >> >> "+OK uuid" on answer, dnd, user busy, cancel ... >> >> or >> >> "NO_ANSWER" on timeout >> >> >> rather then "+OK uuid" for answer and "-ERR Reason" for everything else. >> >> So how can I get the result code of the originate process? In FS console >> I can see the wanted result codes from target (e.g. "mod_dptools.c:2355 >> Originate Failed. Cause: USER_BUSY"). >> >> >> best regards >> Helmut >> -----BEGIN PGP SIGNATURE----- >> Version: GnuPG v1.4.7 (MingW32) >> >> iD8DBQFMZEaB4tZeNddg3dwRAiLXAJ9wS0pEIe3pagDADOXcx/tNlkBr+QCdEL/S >> P0OhzYqbWJ54At6ENrcD1q0= >> =XeB+ >> -----END PGP SIGNATURE----- >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/a9ef3b5e/attachment.html From steveayre at gmail.com Thu Aug 12 12:48:39 2010 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 12 Aug 2010 20:48:39 +0100 Subject: [Freeswitch-users] Question about originate In-Reply-To: References: <4C644681.1050703@ewetel.de> Message-ID: For instance CHANNEL_HANGUP will contain the hangup cause. -Steve On 12 August 2010 20:46, Steven Ayre wrote: > Subscribe to events > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > You'll then see an event for ringing, answer, hangup, failed etc. > > Those events will give you the information you need > > -Steve > > > On 12 August 2010 20:25, Michael Collins wrote: > >> What is your originate command? Are you ignoring early media? Just >> curious. >> -MC >> >> >> On Thu, Aug 12, 2010 at 12:07 PM, Helmut Kuper wrote: >> >>> -----BEGIN PGP SIGNED MESSAGE----- >>> Hash: SHA1 >>> >>> Hello there, >>> >>> I try to originate a new call in "event socket outbound" mode in an >>> external application. I would like to get originate's result code, but I >>> only got >>> >>> "+OK uuid" on answer, dnd, user busy, cancel ... >>> >>> or >>> >>> "NO_ANSWER" on timeout >>> >>> >>> rather then "+OK uuid" for answer and "-ERR Reason" for everything else. >>> >>> So how can I get the result code of the originate process? In FS console >>> I can see the wanted result codes from target (e.g. "mod_dptools.c:2355 >>> Originate Failed. Cause: USER_BUSY"). >>> >>> >>> best regards >>> Helmut >>> -----BEGIN PGP SIGNATURE----- >>> Version: GnuPG v1.4.7 (MingW32) >>> >>> iD8DBQFMZEaB4tZeNddg3dwRAiLXAJ9wS0pEIe3pagDADOXcx/tNlkBr+QCdEL/S >>> P0OhzYqbWJ54At6ENrcD1q0= >>> =XeB+ >>> -----END PGP SIGNATURE----- >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/6db72f0a/attachment.html From helmut.kuper at ewetel.de Thu Aug 12 12:50:11 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Aug 2010 21:50:11 +0200 Subject: [Freeswitch-users] Question about originate In-Reply-To: References: <4C644681.1050703@ewetel.de> Message-ID: <4C645073.3090006@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Michael, for example this: /log console +OK log level console [0] freeswitch at 85.16.246.6@internal> originate sofia/internal/2850 at 85.16.246.6 &park +OK 9d345a98-a64a-11df-82f4-09e441f35e18 freeswitch at 85.16.246.6@internal> on 2850 I simply pressed cancel button to reject call with user_busy regards Helmut Am 12.08.2010 21:25, schrieb Michael Collins: > What is your originate command? Are you ignoring early media? Just curious. > -MC > > On Thu, Aug 12, 2010 at 12:07 PM, Helmut Kuper wrote: > > Hello there, > > I try to originate a new call in "event socket outbound" mode in an > external application. I would like to get originate's result code, but I > only got > > "+OK uuid" on answer, dnd, user busy, cancel ... > > or > > "NO_ANSWER" on timeout > > > rather then "+OK uuid" for answer and "-ERR Reason" for everything else. > > So how can I get the result code of the originate process? In FS console > I can see the wanted result codes from target (e.g. "mod_dptools.c:2355 > Originate Failed. Cause: USER_BUSY"). > > > best regards > Helmut >> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMZFBz4tZeNddg3dwRAl/IAJ9ac73EfpXMkwdZdDy4tgUCM716FgCggrOB QtQdplpKD1nc+k8RBMF5ESQ= =yYFh -----END PGP SIGNATURE----- From tculjaga at gmail.com Thu Aug 12 13:13:16 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 12 Aug 2010 22:13:16 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> Message-ID: On Thu, Aug 12, 2010 at 2:58 PM, Brian West wrote: > 20. > > i never heard of a softswitch that needs double codec license than supported simultaneous calls. This is like you say you need 2000 licenses is needed if you want to switch 1000 simultaneous calls.... strange! something is fishy :) T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/3bc69183/attachment.html From brian at freeswitch.org Thu Aug 12 13:24:10 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Aug 2010 15:24:10 -0500 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> Message-ID: <3D922F11-C874-4D4B-893F-3B449A55EB02@freeswitch.org> Nothing at all is fishy. He said he wanted to transcode and he wanted to record. That would require more licenses. I specifically answered the question he asked. /b On Aug 12, 2010, at 3:13 PM, Tihomir Culjaga wrote: > > > On Thu, Aug 12, 2010 at 2:58 PM, Brian West wrote: > 20. > > > i never heard of a softswitch that needs double codec license than supported simultaneous calls. This is like you say you need 2000 licenses is needed if you want to switch 1000 simultaneous calls.... strange! > > something is fishy :) > > T. From helmut.kuper at ewetel.de Thu Aug 12 13:26:19 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Aug 2010 22:26:19 +0200 Subject: [Freeswitch-users] Question about originate In-Reply-To: <4C645073.3090006@ewetel.de> References: <4C644681.1050703@ewetel.de> <4C645073.3090006@ewetel.de> Message-ID: <4C6458EB.1030100@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, ok, solved it. I had a transfer in my dialplan when bridge fails. The transfer diaplan played media debening on answer code ... thx and sorry Helmut Am 12.08.2010 21:50, schrieb Helmut Kuper: > Hi Michael, > > for example this: > > /log console > > +OK log level console [0] > freeswitch at 85.16.246.6@internal> originate > sofia/internal/2850 at 85.16.246.6 &park > +OK 9d345a98-a64a-11df-82f4-09e441f35e18 > > freeswitch at 85.16.246.6@internal> > > > on 2850 I simply pressed cancel button to reject call with user_busy > > regards > Helmut > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMZFjr4tZeNddg3dwRAnvfAKCq+dbjCM1/NtpM84NM8wtkTIYUlgCdHeJV s/nIiatwn70fVeOhTktgFGs= =lweV -----END PGP SIGNATURE----- From david.ponzone at gmail.com Thu Aug 12 13:28:02 2010 From: david.ponzone at gmail.com (David Ponzone) Date: Thu, 12 Aug 2010 22:28:02 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> Message-ID: No, if you need to switch one call from G729 to G711, you need one license. But recording a call between 2 G729 endpoints require to record both legs, so you need to decode 2 streams, so you need 2 licenses. I am pretty sure it's the same on all commercial platforms, but of course, they won't mention that in big letters on their marketing papers... David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 12/08/2010 ? 22:13, Tihomir Culjaga a ?crit : > > > On Thu, Aug 12, 2010 at 2:58 PM, Brian West > wrote: > 20. > > > i never heard of a softswitch that needs double codec license than > supported simultaneous calls. This is like you say you need 2000 > licenses is needed if you want to switch 1000 simultaneous calls.... > strange! > > something is fishy :) > > T. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/ef3cb52b/attachment.html From helmut.kuper at ewetel.de Thu Aug 12 13:48:47 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Aug 2010 22:48:47 +0200 Subject: [Freeswitch-users] A further question about originate Message-ID: <4C645E2F.3040608@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to set a custom channel variable within originate command as described here: http://wiki.freeswitch.org/wiki/Mod_commands [...] Note: you can set any channel variable, even custom ones. Use single quotes to enclose values with spaces, commas, etc. originate {my_own_var=my_value}sofia/mydomain.com/that.ext at 1.2.3.4 15555551212 originate {my_own_var='my value'}sofia/mydomain.com/that.ext at 1.2.3.4 15555551212 [...] My command is like this: originate {my_var=my_data}sofia/internal/2850 at 85.16.246.6 &park When I call info in dialplan, the variable is missing When I do originate {my_var=my_data}user/2850 at 85.16.246.6 &park the variable is there! Any reasons for this? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMZF4v4tZeNddg3dwRAiwOAJoCGHZEZCrGfDDUPDB10p4p9GCzAQCfbtKS o+9fmLXxswvtTFebbnoWbuM= =A/i/ -----END PGP SIGNATURE----- From brian at freeswitch.org Thu Aug 12 13:53:26 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Aug 2010 15:53:26 -0500 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> Message-ID: <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> You do make an excellent point too... I was thinking 711 to 729 plus recording... it gets interesting when you toss stereo in it. /b On Aug 12, 2010, at 3:28 PM, David Ponzone wrote: > No, if you need to switch one call from G729 to G711, you need one license. > But recording a call between 2 G729 endpoints require to record both legs, so you need to decode 2 streams, so you need 2 licenses. > I am pretty sure it's the same on all commercial platforms, but of course, they won't mention that in big letters on their marketing papers... > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > From mnhassan at usa.net Thu Aug 12 14:06:31 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Fri, 13 Aug 2010 03:06:31 +0600 Subject: [Freeswitch-users] g729 In-Reply-To: <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> Message-ID: What if we record the session in encoded format, and use licenses when we need to play in a different format? Is this possible? What if the user called in from an G.729 enabled endpoint, would a license be needed to play an encoded stream? Regards HASSAN On 2010-08-13, Brian West wrote: > You do make an excellent point too... I was thinking 711 to 729 plus > recording... it gets interesting when you toss stereo in it. > > /b > > On Aug 12, 2010, at 3:28 PM, David Ponzone wrote: > >> No, if you need to switch one call from G729 to G711, you need one >> license. >> But recording a call between 2 G729 endpoints require to record both legs, >> so you need to decode 2 streams, so you need 2 licenses. >> I am pretty sure it's the same on all commercial platforms, but of course, >> they won't mention that in big letters on their marketing papers... >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message >> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de >> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From brian at freeswitch.org Thu Aug 12 14:15:37 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Aug 2010 16:15:37 -0500 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> Message-ID: <872A78F1-916B-4237-8550-2CD536EFD823@freeswitch.org> it should. /b On Aug 12, 2010, at 4:06 PM, Nyamul Hassan wrote: > What if we record the session in encoded format, and use licenses when > we need to play in a different format? > > Is this possible? What if the user called in from an G.729 enabled > endpoint, would a license be needed to play an encoded stream? > > Regards > HASSAN From msc at freeswitch.org Thu Aug 12 14:17:42 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Aug 2010 14:17:42 -0700 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> Message-ID: On Thu, Aug 12, 2010 at 2:06 PM, Nyamul Hassan wrote: > What if we record the session in encoded format, and use licenses when > we need to play in a different format? > > Is this possible? What if the user called in from an G.729 enabled > endpoint, would a license be needed to play an encoded stream? > > Regards > HASSAN > > Can I just say that I hate g.729? :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/0c9ec916/attachment.html From david.ponzone at ipeva.fr Thu Aug 12 14:19:15 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 12 Aug 2010 23:19:15 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> Message-ID: AFAIK, record using native file format is not possible. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 12/08/2010 ? 23:06, Nyamul Hassan a ?crit : > What if we record the session in encoded format, and use licenses when > we need to play in a different format? > > Is this possible? What if the user called in from an G.729 enabled > endpoint, would a license be needed to play an encoded stream? > > Regards > HASSAN > > > On 2010-08-13, Brian West wrote: >> You do make an excellent point too... I was thinking 711 to 729 plus >> recording... it gets interesting when you toss stereo in it. >> >> /b >> >> On Aug 12, 2010, at 3:28 PM, David Ponzone wrote: >> >>> No, if you need to switch one call from G729 to G711, you need one >>> license. >>> But recording a call between 2 G729 endpoints require to record >>> both legs, >>> so you need to decode 2 streams, so you need 2 licenses. >>> I am pretty sure it's the same on all commercial platforms, but of >>> course, >>> they won't mention that in big letters on their marketing papers... >>> >>> David Ponzone Direction Technique >>> email: david.ponzone at ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr - www.ipeva-studio.com >>> >>> Ce message et toutes les pi?ces jointes sont confidentiels et >>> ?tablis ? >>> l'intention exclusive de ses destinataires. Toute utilisation ou >>> diffusion >>> non autoris?e est interdite. Tout message ?lectronique est >>> susceptible >>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce >>> message >>> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >>> destinataire de >>> ce message, merci de le d?truire imm?diatement et d'avertir >>> l'exp?diteur. >>> >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > Sent from my mobile device > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/18f4fb8e/attachment.html From brian at freeswitch.org Thu Aug 12 14:24:56 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Aug 2010 16:24:56 -0500 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> Message-ID: <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> No you can... just not in stereo :P /b On Aug 12, 2010, at 4:19 PM, David Ponzone wrote: > AFAIK, record using native file format is not possible. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > From david.ponzone at ipeva.fr Thu Aug 12 14:35:10 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 12 Aug 2010 23:35:10 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> Message-ID: Brian, you mean now I can use record_session to record a G729 call to native format ? I remember it was not possible some weeks ago. Perhaps I am confusing something, I know there are various record functions in FS. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 12/08/2010 ? 23:24, Brian West a ?crit : > No you can... just not in stereo :P > > /b > > On Aug 12, 2010, at 4:19 PM, David Ponzone wrote: > >> AFAIK, record using native file format is not possible. >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/c0758269/attachment-0001.html From jaybinks at gmail.com Thu Aug 12 14:32:31 2010 From: jaybinks at gmail.com (Jay Binks) Date: Fri, 13 Aug 2010 07:32:31 +1000 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: Message-ID: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> Sounds like the rtp stream has the sync AND your fs is sending 2833 to your upstream. So the upstream is accepting your 2833 AND doing their own tone detection. Have a look here , http://wiki.freeswitch.org/wiki/RTP_Issues Maybe you can't send 2833 to your upstream ?? Jay On 13/08/2010, at 3:29 AM, Giovanni Maruzzelli wrote: > On Thu, Aug 12, 2010 at 7:02 PM, Dennis wrote: >> hi there, >> >> at the moment we have a serious and urgent problem with transmitting > > While you probably will receive a good answer here on the mailing > list, if your problem is urgent (and maybe will require exchanges of > questions and answers), I would counsel you to join the #freeswitch > IRC channel in the irc.freenode.net IRC server, and pose your question > again there, so you'll have a quick interaction with the "gurus". > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pjintheusa at gmail.com Thu Aug 12 15:31:17 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 12 Aug 2010 18:31:17 -0400 Subject: [Freeswitch-users] Enterprise Originate and group_confirm In-Reply-To: References: Message-ID: Perhaps I am going about this the wrong way. This is what I am trying to acheive. Basically a simple call forward/follow me service as follows: 1) call comes in via the PSTN hits the dial plan 2) aim is to call two numbers simultaneously - the first to pick up and press 1, is bridged the PSTN call. 3) the two called numbers will try upto three gateways (if gw 1 fails with NO_DESTINATION_ROUTE, try gateway 2 etc 4) if no one accepts the call (I.e. does not pick up OR picks up and then hangs up without pressing 1) the call should go to voicemail This should be possible with a combination of group_confirm, continue_on_fail & enterprise originate, but I cannot seem to get it quite right. Has anyone else achieved this using the dial plan? Thanks Pj On Wednesday, August 11, 2010, Phillip Jones wrote: > Hi there, > > I have the following dialplan which uses enterprise originate and group_confirm: > > The plan is that two numbers are called simutanuously, each number has several gateway to use, incase a gateway fails. > > > > > > > > > > > > Every thing works great in terms of mutiple legs ringing,?bridging,?hanging up etc. > > I have one issue though: > > When either of the numbers, 6095553828 for example, answers the call but does not accept it (i.e. does not press 1)?and hangs up, the dialplan tries the next gateway for the same number and therefore 6095553828 ends up getting another call even though he has rejected it. I was expecting that once the call is answered and rejected?(not bridged) then that part of the dialplan would stop. > > > Any help appreciated. > > Thanks > > Pj > > > From shamun.toha at gmail.com Thu Aug 12 13:16:21 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Thu, 12 Aug 2010 22:16:21 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype4COM or SkypeKIT we are using? Message-ID: NOTE: Please have a look: http://forum.skype.com/index.php?showtopic=637293&st=0&p=2897693&#entry2897693 I am completely now confused: Does that mean Skype Module in FreeSwitch will be dead or is dead? Does that mean Skype Module in FreeSwitch, can now handle 100 to 300 or 999 simultaneous calls? Does that mean Skype Module in FreeSwitch, will be never needed? Does that mean Skype Module in FreeSwitch, will need to be completely renew after running it for months? Thank you Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/51bc4a1e/attachment.html From tony.tin at noahmedia.com.hk Thu Aug 12 19:16:15 2010 From: tony.tin at noahmedia.com.hk (Tony Tin) Date: Fri, 13 Aug 2010 10:16:15 +0800 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: Message-ID: try this http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate Tony On Fri, Aug 13, 2010 at 1:02 AM, Dennis wrote: > hi there, > > at the moment we have a serious and urgent problem with transmitting > dtmf-inputs in the right way from one endpoint through fs to another > endpoint. we are not very shure, if the problem is on our or on the > carriers side. > > digits send through fs from one side to another are not received > correctly. is there someone who could help us (quite quickly)? > > we are receiving the dtmf-signals on fs as RFC-2833. our fs-setting > are the following: > > > > > > > - we are receiving a call and we connect to another side through fs. > - the caller enters the following dtmf-digits: 987654321. > - fs receives the dtmf-inputs absolutely correctly !!! > - the target receives the following dtmf-digits: 98877665543321 > > it is important to say, that the call comes over the "normal > phone-line" (or what ever a phonecall is called, which does not come > over voip - landline?). per VOIP (cirpack) our fs-servers receive this > call/dtmf-inputs and then we send the call to the target over the > "normal phone-line". > > does anybody have a clue where our problems might be? > what could we do to find out more about this probleme? where could we > debug? > > thanks a lot > dennis > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/1624876f/attachment.html From petedao at gmail.com Thu Aug 12 19:23:25 2010 From: petedao at gmail.com (Pete Kay) Date: Fri, 13 Aug 2010 10:23:25 +0800 Subject: [Freeswitch-users] re-invite problem In-Reply-To: References: <5693B149-67BD-4675-8254-8DD963E33B42@freeswitch.org> Message-ID: Hi, Yes, here is the packet trace for the re-invite... recv 513 bytes from udp/[192.168.1.114]:5060 at 07:37:55.968109: ------------------------------------------------------------------------ INVITE sip:sipp at 192.168.1.114:5070 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 From: sipp ;tag=1 To: ;tag=4tj1e3m89ZSBj Contact: sip:1899 at 192.168.1.3:9871 Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 CSeq: 1 INVITE Session-Expires: 120;refresher=uac Min-SE: 90 Supported: 100rel, timer Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x7f7f50004670): msg 0x7f7f502518d0 (513 bytes) from udp/192.168.1.114:5070/sip next=(nil) nta: received INVITE sip:sipp at 192.168.1.114:5070 SIP/2.0 (CSeq 1) nta: canonizing sip:sipp at 192.168.1.114:5070 with contact nta: INVITE (1) going to existing leg nta: timer shortened to 200 ms nua: nua_stack_process_request: entering soa_init_offer_answer(static::0x7f7f501de8e0) called tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 tport_resolve addrinfo = 192.168.1.114:5060 tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 tport_vsend(0x7f7f50004670): 387 bytes of 387 to udp/192.168.1.114:5060 tport_vsend returned 387 send 387 bytes to udp/[192.168.1.114]:5060 at 07:37:55.968290: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 Record-Route: From: sipp ;tag=1 To: ;tag=4tj1e3m89ZSBj Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 CSeq: 1 INVITE User-Agent: FreeSWITCH Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (1) nua(0x7f7e3408b0a0): event i_invite 100 Trying nua: nua_application_event: entering nua: nua_handle_magic: entering nua(0x7f7e3408b0a0): ready call updated: received nua(0x7f7e3408b0a0): event i_state 100 Trying nua: nua_application_event: entering 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel sofia/external/1899 at 192.168.1.3:9871 entering state [received][100] nua: nua_respond: entering nua(0x7f7e3408b0a0): sent signal r_respond nua: nua_handle_magic: entering nua(0x7f7e3408b0a0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering soa_set_params(static::0x7f7f501de8e0, ...) called nua: nua_invite_server_respond: entering soa_generate_offer(static::0x7f7f501de8e0, 0) called soa_static_offer_answer_action(0x7f7f501de8e0, soa_generate_offer): called soa_sdp_mode_set(0x7f7f50226fe0, (nil), ""): called soa_get_local_sdp(static::0x7f7f501de8e0, [(nil)], [0x7f7f60520d30], [0x7f7f60520d3c]) called tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 tport_resolve addrinfo = 192.168.1.114:5060 tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 tport_vsend(0x7f7f50004670): 944 bytes of 944 to udp/192.168.1.114:5060 tport_vsend returned 944 send 944 bytes to udp/[192.168.1.114]:5060 at 07:37:55.968539: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 Record-Route: From: sipp ;tag=1 To: ;tag=4tj1e3m89ZSBj Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 CSeq: 1 INVITE Contact: User-Agent: FreeSWITCH Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Require: timer Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 213 v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.114 s=- c=IN IP4 192.168.1.114 t=0 0 m=audio 27938 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (1) nua(0x7f7e3408b0a0): ready call updated: completed sent offer soa_get_local_sdp(static::0x7f7f501de8e0, [0x7f7f60520e08], [0x7f7f60520e00], [(nil)]) called nua(0x7f7e3408b0a0): event i_state 200 OK nua: nua_application_event: entering 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel sofia/external/1899 at 192.168.1.3:9871 entering state [completed][200] nua: nua_handle_magic: entering tport_wakeup_pri(0x7f7f50004670): events IN tport_recv_event(0x7f7f50004670) tport_recv_iovec(0x7f7f50004670) msg 0x7f7f5022be60 from (udp/192.168.1.114:5070) has 525 bytes, veclen = 1 recv 525 bytes from udp/[192.168.1.114]:5060 at 07:37:55.968843: ------------------------------------------------------------------------ ACK sip:sipp at 192.168.1.114:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.2 Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-8 From: sipp ;tag=1 To: ;tag=4tj1e3m89ZSBj Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 CSeq: 1 ACK Max-Forwards: 69 Content-Type: application/sdp Content-Length: 133 v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ------------------------------------------------------------------------ tport_deliver(0x7f7f50004670): msg 0x7f7f5022be60 (525 bytes) from udp/192.168.1.114:5070/sip next=(nil) nta: received ACK sip:sipp at 192.168.1.114:5070 SIP/2.0 (CSeq 1) nta: ACK (1) is going to INVITE (1) nua: process_ack_or_cancel: entering soa_set_remote_sdp(static::0x7f7f501de8e0, (nil), 0x7f7f501eb298, 133) called soa_process_answer(static::0x7f7f501de8e0) called soa_static_offer_answer_action(0x7f7f501de8e0, soa_process_answer): called soa_sdp_mode_set(0x7f7f50226fe0, 0x7f7f50245750, ""): called soa_activate(static::0x7f7f501de8e0, (nil)) called soa_clear_remote_sdp(static::0x7f7f501de8e0) called nua(0x7f7e3408b0a0): event i_ack 200 OK nua: nua_application_event: entering nua: nua_handle_magic: entering nua(0x7f7e3408b0a0): ready call updated: ready received answer soa_get_remote_sdp(static::0x7f7f501de8e0, [0x7f7f60520968], [0x7f7f60520960], [(nil)]) called soa_get_params(static::0x7f7f501de8e0, ...) called nua(0x7f7e3408b0a0): event i_state 200 OK nua: nua_application_event: entering 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel sofia/external/1899 at 192.168.1.3:9871 entering state [ready][200] 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3296 Remote SDP: v=0 o=user1 53655765 2353687637 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 nua: nua_respond: entering nua(0x7f7e3408b0a0): sent signal r_respond 2010-08-10 00:37:55.968013 [NOTICE] sofia.c:3712 Hangup sofia/external/1899 at 192.168.1.3:9871 [CS_HIBERNATE] [INCOMPATIBLE_DESTINATION] 2010-08-10 00:37:55.969068 [DEBUG] switch_channel.c:1683 Send signal sofia/external/1899 at 192.168.1.3:9871 [KILL] 2010-08-10 00:37:55.969068 [DEBUG] switch_core_session.c:932 Send signal sofia/external/1899 at 192.168.1.3:9871 [BREAK] nua: nua_handle_magic: entering nua(0x7f7e3408b0a0): event i_active 200 Call active nua: nua_application_event: entering nua(): refresh session after 88 seconds (in [88..88]) nua(0x7f7e3408b0a0): recv signal r_respond 488 Not Acceptable Here nua(0x7f7e3408b0a0): event i_error 500 Responding to a Non-Existing Request nua: nua_application_event: entering 2010-08-10 00:37:55.969068 [DEBUG] switch_core_state_machine.c:398 (sofia/external/1899 at 192.168.1.3:9871) Running State Change CS_HANGUP 2010-08-10 00:37:55.969068 [DEBUG] switch_core_state_machine.c:434 (sofia/external/1899 at 192.168.1.3:9871) State HANGUP 2010-08-10 00:37:55.969068 [DEBUG] mod_sofia.c:338 Channel sofia/external/1899 at 192.168.1.3:9871 hanging up, cause: INCOMPATIBLE_DESTINATION 2010-08-10 00:37:55.969068 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/external/1899 at 192.168.1.3:9871 nua: nua_bye: entering nua(0x7f7e3408b0a0): sent signal r_bye nua(0x7f7e3408b0a0): recv signal r_bye nua: nua_stack_set_params: entering soa_set_params(static::0x7f7f501de8e0, ...) called soa_terminate(static::0x7f7f501de8e0) called 2010-08-10 00:37:55.969068 [NOTICE] switch_ivr_bridge.c:710 Hangup sofia/external/sipp at 192.168.1.3:8970 [CS_HIBERNATE] [INCOMPATIBLE_DESTINATION] soa_init_offer_answer(static::0x7f7f501de8e0) called nta: selecting scheme sip tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 tport_resolve addrinfo = 192.168.1.114:5060 tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 2010-08-10 00:37:55.970127 [DEBUG] switch_channel.c:1683 Send signal sofia/external/sipp at 192.168.1.3:8970 [KILL] 2010-08-10 00:37:55.970127 [DEBUG] mod_limit.c:195 originate_disposition=[SUCCESS] 2010-08-10 00:37:55.970127 [DEBUG] switch_core_session.c:932 Send signal sofia/external/sipp at 192.168.1.3:8970 [BREAK] tport_vsend(0x7f7f50004670): 619 bytes of 619 to udp/192.168.1.114:5060 tport_vsend returned 619 send 619 bytes to udp/[192.168.1.114]:5060 at 07:37:55.970197: ------------------------------------------------------------------------ 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:46 sofia/external/1899 at 192.168.1.3:9871 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION BYE sip:1899 at 192.168.1.3:9871 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKeFg2t9SeUmHSN Route: Max-Forwards: 70 From: "sipp" ;tag=4tj1e3m89ZSBj To: ;tag=1 Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 CSeq: 376874 BYE Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:398 (sofia/external/sipp at 192.168.1.3:8970) Running State Change CS_HANGUP Supported: timer, precondition, path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 02010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:434 (sofia/external/1899 at 192.168.1.3:9871) State HANGUP going to sleep ------------------------------------------------------------------------ nta: sent BYE (376874) to UDP/192.168.1.114:5060 tport_pend(0x7f7f50004670): pending 0x7f7f50258650 for udp/192.168.1.114:5070 (already 0) 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:461 Hangup Command decre_call_stat(44 60 43 59 14 429178 0): 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:434 (sofia/external/sipp at 192.168.1.3:8970) State HANGUP 2010-08-10 00:37:55.970127 [DEBUG] mod_sofia.c:338 Channel sofia/external/sipp at 192.168.1.3:8970 hanging up, cause: INCOMPATIBLE_DESTINATION 2010-08-10 00:37:55.970127 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/external/sipp at 192.168.1.3:8970 nua: nua_bye: entering nua(0x7f7f5022abd0): sent signal r_bye nua(0x7f7f5022abd0): recv signal r_bye nua: nua_stack_set_params: entering soa_set_params(static::0x7f7f50202440, ...) called soa_terminate(static::0x7f7f50202440) called soa_init_offer_answer(static::0x7f7f50202440) called nta: selecting scheme sip tport_tsend(0x7f7f50004670) tpn = */192.168.1.114:5060 tport_resolve addrinfo = 192.168.1.114:5060 tport_by_addrinfo(0x7f7f50004670): not found by name */192.168.1.114:5060 tport_vsend(0x7f7f50004670): 637 bytes of 637 to udp/192.168.1.114:5060 tport_vsend returned 637 send 637 bytes to udp/[192.168.1.114]:5060 at 07:37:55.970869: ------------------------------------------------------------------------ BYE sip:sipp at 192.168.1.3:8970 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKFr9tv4ajrX7BH Route: Max-Forwards: 70 From: sut ;tag=3HS8c834cQ3rp To: sipp ;tag=1 Call-ID: 1-9173 at 192.168.1.3 CSeq: 376873 BYE Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ nta: sent BYE (376873) to */192.168.1.114:5060 tport_pend(0x7f7f50004670): pending 0x7f7f50254a40 for udp/192.168.1.114:5070 (already 1) 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:46 sofia/external/sipp at 192.168.1.3:8970 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:434 (sofia/external/sipp at 192.168.1.3:8970) State HANGUP going to sleep 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:476 (sofia/external/sipp at 192.168.1.3:8970) State Change CS_HANGUP -> CS_REPORTING 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:932 Send signal sofia/external/sipp at 192.168.1.3:8970 [BREAK] 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:398 (sofia/external/sipp at 192.168.1.3:8970) Running State Change CS_REPORTING 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 (sofia/external/sipp at 192.168.1.3:8970) State REPORTING 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:53 sofia/external/sipp at 192.168.1.3:8970 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 (sofia/external/sipp at 192.168.1.3:8970) State REPORTING going to sleep 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:411 (sofia/external/sipp at 192.168.1.3:8970) State Change CS_REPORTING -> CS_DESTROY 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:1068 Session 17184 (sofia/external/sipp at 192.168.1.3:8970) Locked, Waiting on external entities 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:476 (sofia/external/1899 at 192.168.1.3:9871) State Change CS_HANGUP -> CS_REPORTING 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1086 Session 17184 (sofia/external/sipp at 192.168.1.3:8970) Ended 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:932 Send signal sofia/external/1899 at 192.168.1.3:9871 [BREAK] 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/sipp at 192.168.1.3:8970 [CS_DESTROY] 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:398 (sofia/external/1899 at 192.168.1.3:9871) Running State Change CS_REPORTING 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 (sofia/external/1899 at 192.168.1.3:9871) State REPORTING 2010-08-10 00:37:55.971183 [DEBUG] mod_cdr_bdb.c:139 Is number 1899 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 (sofia/external/sipp at 192.168.1.3:8970) State DESTROY 2010-08-10 00:37:55.971183 [DEBUG] mod_sofia.c:255 sofia/external/sipp at 192.168.1.3:8970 SOFIA DESTROY 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:60 sofia/external/sipp at 192.168.1.3:8970 Standard DESTROY 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 (sofia/external/sipp at 192.168.1.3:8970) State DESTROY going to sleep 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:53 sofia/external/1899 at 192.168.1.3:9871 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 (sofia/external/1899 at 192.168.1.3:9871) State REPORTING going to sleep 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:411 (sofia/external/1899 at 192.168.1.3:9871) State Change CS_REPORTING -> CS_DESTROY 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:1068 Session 17185 (sofia/external/1899 at 192.168.1.3:9871) Locked, Waiting on external entities 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1086 Session 17185 (sofia/external/1899 at 192.168.1.3:9871) Ended 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1899 at 192.168.1.3:9871 [CS_DESTROY] 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 (sofia/external/1899 at 192.168.1.3:9871) State DESTROY 2010-08-10 00:37:55.971183 [DEBUG] mod_sofia.c:255 sofia/external/1899 at 192.168.1.3:9871 SOFIA DESTROY 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:60 sofia/external/1899 at 192.168.1.3:9871 Standard DESTROY 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 (sofia/external/1899 at 192.168.1.3:9871) State DESTROY going to sleep nta: timer set next to 301 ms nta: timer E fired, retransmit BYE (376874) tport_release(0x7f7f50004670): 0x7f7f50258650 by 0x7f7f50275640 with (nil) tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 tport_resolve addrinfo = 192.168.1.114:5060 tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 tport_vsend(0x7f7f50004670): 619 bytes of 619 to udp/192.168.1.114:5060 tport_vsend returned 619 send 619 bytes to udp/[192.168.1.114]:5060 at 07:37:56.469550: ------------------------------------------------------------------------ BYE sip:1899 at 192.168.1.3:9871 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKeFg2t9SeUmHSN Route: Max-Forwards: 70 From: "sipp" ;tag=4tj1e3m89ZSBj To: ;tag=1 Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 CSeq: 376874 BYE Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ nta: resent BYE (376874) to UDP/192.168.1.114:5060 tport_pend(0x7f7f50004670): pending 0x7f7f50258650 for udp/192.168.1.114:5070 (already 1) nta_outgoing_timer: 1/2 resent, 0/3 tout, 0/1 term, 0/4 free nta: timer set next to 1 ms nta: timer E fired, retransmit BYE (376873) tport_release(0x7f7f50004670): 0x7f7f50254a40 by 0x7f7f50247670 with (nil) tport_tsend(0x7f7f50004670) tpn = */192.168.1.114:5060 tport_resolve addrinfo = 192.168.1.114:5060 tport_by_addrinfo(0x7f7f50004670): not found by name */192.168.1.114:5060 tport_vsend(0x7f7f50004670): 637 bytes of 637 to udp/192.168.1.114:5060 tport_vsend returned 637 send 637 bytes to udp/[192.168.1.114]:5060 at 07:37:56.470547: ------------------------------------------------------------------------ BYE sip:sipp at 192.168.1.3:8970 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKFr9tv4ajrX7BH Route: Max-Forwards: 70 From: sut ;tag=3HS8c834cQ3rp To: sipp ;tag=1 Call-ID: 1-9173 at 192.168.1.3 CSeq: 376873 BYE Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ nta: resent BYE (376873) to */192.168.1.114:5060 tport_pend(0x7f7f50004670): pending 0x7f7f50254a40 for udp/192.168.1.114:5070 (already 1) nta_outgoing_timer: 1/2 resent, 0/3 tout, 0/1 term, 0/4 free nta: timer set next to 638 ms On Fri, Aug 13, 2010 at 2:05 AM, Steven Ayre wrote: > Do you have a packet trace available? > > -Steve > > > > On 12 August 2010 14:25, Pete Kay wrote: >> >> Hi, >> >> I check tags and callid. ?It is the same dialog. ?Also, the invite is >> accepted and I can see UAC does respond 200 OK . ?The freeswitch sends >> out BYE after ACK. >> >> The problem I am seeing is that the re-invite triggers the dialplan >> execution which based on its logic is responding with a 488 within the >> dialplan using the respond app. ?When freeswitch receives the 488, it >> can't recognize the dialog so it sends out BYE. >> >> Therefore, I think the way to solve this is to configure sofia so that >> the invite won't trigger the execution of dialplan. >> >> Is there anyway to do that? >> >> Thanks, >> P >> >> On Thu, Aug 12, 2010 at 9:00 PM, Brian West wrote: >> > Chances are it is in fact a new dialog have you double checked the >> > to/from tags and callid? ?Also if its without an SDP you would have to >> > enable 3pcc on the profile to accept it possibly. >> > >> > /b >> > >> > On Aug 12, 2010, at 3:11 AM, Pete Kay wrote: >> > >> >> I am running a b2bua with freeswitch. ?It is fine until a Mitel UAS >> >> starts sending INVITE without sdp and ACK with sdp. Freeswitch seems >> >> to treat it as another dialog and sends it to dialplan handling. >> >> >> >> Within the dialplan, how can I recognize request as a re-invite and >> >> possibly ignore it? >> >> >> >> Does anyone know how to resolve this problem? >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sos at sokhapkin.dyndns.org Thu Aug 12 19:29:47 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Thu, 12 Aug 2010 22:29:47 -0400 Subject: [Freeswitch-users] re-invite problem In-Reply-To: References: Message-ID: <201008122229.47612.sos@sokhapkin.dyndns.org> Looks like you have no 3rd party call control enabled in FS settings... On Thursday 12 August 2010, Pete Kay wrote: > Hi, > > Yes, here is the packet trace for the re-invite... > > recv 513 bytes from udp/[192.168.1.114]:5060 at 07:37:55.968109: > ------------------------------------------------------------------------ > INVITE sip:sipp at 192.168.1.114:5070 SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 > Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 > From: sipp ;tag=1 > To: ;tag=4tj1e3m89ZSBj > Contact: sip:1899 at 192.168.1.3:9871 > Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 > CSeq: 1 INVITE > Session-Expires: 120;refresher=uac > Min-SE: 90 > Supported: 100rel, timer > Max-Forwards: 69 > Content-Length: 0 > > ------------------------------------------------------------------------ > tport_deliver(0x7f7f50004670): msg 0x7f7f502518d0 (513 bytes) from > udp/192.168.1.114:5070/sip next=(nil) > nta: received INVITE sip:sipp at 192.168.1.114:5070 SIP/2.0 (CSeq 1) > nta: canonizing sip:sipp at 192.168.1.114:5070 with contact > nta: INVITE (1) going to existing leg > nta: timer shortened to 200 ms > nua: nua_stack_process_request: entering > soa_init_offer_answer(static::0x7f7f501de8e0) called > tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 > tport_resolve addrinfo = 192.168.1.114:5060 > tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 > tport_vsend(0x7f7f50004670): 387 bytes of 387 to udp/192.168.1.114:5060 > tport_vsend returned 387 > send 387 bytes to udp/[192.168.1.114]:5060 at 07:37:55.968290: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 > Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 > Record-Route: > From: sipp ;tag=1 > To: ;tag=4tj1e3m89ZSBj > Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 > CSeq: 1 INVITE > User-Agent: FreeSWITCH > Content-Length: 0 > > ------------------------------------------------------------------------ > nta: sent 100 Trying for INVITE (1) > nua(0x7f7e3408b0a0): event i_invite 100 Trying > nua: nua_application_event: entering > nua: nua_handle_magic: entering > nua(0x7f7e3408b0a0): ready call updated: received > nua(0x7f7e3408b0a0): event i_state 100 Trying > nua: nua_application_event: entering > 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel > sofia/external/1899 at 192.168.1.3:9871 entering state [received][100] > nua: nua_respond: entering > nua(0x7f7e3408b0a0): sent signal r_respond > nua: nua_handle_magic: entering > nua(0x7f7e3408b0a0): recv signal r_respond 200 OK > nua: nua_stack_set_params: entering > soa_set_params(static::0x7f7f501de8e0, ...) called > nua: nua_invite_server_respond: entering > soa_generate_offer(static::0x7f7f501de8e0, 0) called > soa_static_offer_answer_action(0x7f7f501de8e0, soa_generate_offer): called > soa_sdp_mode_set(0x7f7f50226fe0, (nil), ""): called > soa_get_local_sdp(static::0x7f7f501de8e0, [(nil)], [0x7f7f60520d30], > [0x7f7f60520d3c]) called > tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 > tport_resolve addrinfo = 192.168.1.114:5060 > tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 > tport_vsend(0x7f7f50004670): 944 bytes of 944 to udp/192.168.1.114:5060 > tport_vsend returned 944 > send 944 bytes to udp/[192.168.1.114]:5060 at 07:37:55.968539: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 > Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 > Record-Route: > From: sipp ;tag=1 > To: ;tag=4tj1e3m89ZSBj > Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 > CSeq: 1 INVITE > Contact: > User-Agent: FreeSWITCH > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Require: timer > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uac > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 213 > > v=0 > o=user1 53655765 2353687637 IN IP4 192.168.1.114 > s=- > c=IN IP4 192.168.1.114 > t=0 0 > m=audio 27938 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > ------------------------------------------------------------------------ > nta: sent 200 OK for INVITE (1) > nua(0x7f7e3408b0a0): ready call updated: completed sent offer > soa_get_local_sdp(static::0x7f7f501de8e0, [0x7f7f60520e08], > [0x7f7f60520e00], [(nil)]) called > nua(0x7f7e3408b0a0): event i_state 200 OK > nua: nua_application_event: entering > 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel > sofia/external/1899 at 192.168.1.3:9871 entering state [completed][200] > nua: nua_handle_magic: entering > tport_wakeup_pri(0x7f7f50004670): events IN > tport_recv_event(0x7f7f50004670) > tport_recv_iovec(0x7f7f50004670) msg 0x7f7f5022be60 from > (udp/192.168.1.114:5070) has 525 bytes, veclen = 1 > recv 525 bytes from udp/[192.168.1.114]:5060 at 07:37:55.968843: > ------------------------------------------------------------------------ > ACK sip:sipp at 192.168.1.114:5070 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.2 > Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-8 > From: sipp ;tag=1 > To: ;tag=4tj1e3m89ZSBj > Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 > CSeq: 1 ACK > Max-Forwards: 69 > Content-Type: application/sdp > Content-Length: 133 > > v=0 > o=user1 53655765 2353687637 IN IP4 192.168.1.3 > s=- > c=IN IP4 192.168.1.3 > t=0 0 > m=audio 6000 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > ------------------------------------------------------------------------ > tport_deliver(0x7f7f50004670): msg 0x7f7f5022be60 (525 bytes) from > udp/192.168.1.114:5070/sip next=(nil) > nta: received ACK sip:sipp at 192.168.1.114:5070 SIP/2.0 (CSeq 1) > nta: ACK (1) is going to INVITE (1) > nua: process_ack_or_cancel: entering > soa_set_remote_sdp(static::0x7f7f501de8e0, (nil), 0x7f7f501eb298, 133) > called soa_process_answer(static::0x7f7f501de8e0) called > soa_static_offer_answer_action(0x7f7f501de8e0, soa_process_answer): called > soa_sdp_mode_set(0x7f7f50226fe0, 0x7f7f50245750, ""): called > soa_activate(static::0x7f7f501de8e0, (nil)) called > soa_clear_remote_sdp(static::0x7f7f501de8e0) called > nua(0x7f7e3408b0a0): event i_ack 200 OK > nua: nua_application_event: entering > nua: nua_handle_magic: entering > nua(0x7f7e3408b0a0): ready call updated: ready received answer > soa_get_remote_sdp(static::0x7f7f501de8e0, [0x7f7f60520968], > [0x7f7f60520960], [(nil)]) called > soa_get_params(static::0x7f7f501de8e0, ...) called > nua(0x7f7e3408b0a0): event i_state 200 OK > nua: nua_application_event: entering > 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel > sofia/external/1899 at 192.168.1.3:9871 entering state [ready][200] > 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3296 Remote SDP: > v=0 > o=user1 53655765 2353687637 IN IP4 192.168.1.3 > s=- > c=IN IP4 192.168.1.3 > t=0 0 > m=audio 6000 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > > nua: nua_respond: entering > nua(0x7f7e3408b0a0): sent signal r_respond > 2010-08-10 00:37:55.968013 [NOTICE] sofia.c:3712 Hangup > sofia/external/1899 at 192.168.1.3:9871 [CS_HIBERNATE] > [INCOMPATIBLE_DESTINATION] > 2010-08-10 00:37:55.969068 [DEBUG] switch_channel.c:1683 Send signal > sofia/external/1899 at 192.168.1.3:9871 [KILL] > 2010-08-10 00:37:55.969068 [DEBUG] switch_core_session.c:932 Send > signal sofia/external/1899 at 192.168.1.3:9871 [BREAK] > nua: nua_handle_magic: entering > nua(0x7f7e3408b0a0): event i_active 200 Call active > nua: nua_application_event: entering > nua(): refresh session after 88 seconds (in [88..88]) > nua(0x7f7e3408b0a0): recv signal r_respond 488 Not Acceptable Here > nua(0x7f7e3408b0a0): event i_error 500 Responding to a Non-Existing Request > nua: nua_application_event: entering > 2010-08-10 00:37:55.969068 [DEBUG] switch_core_state_machine.c:398 > (sofia/external/1899 at 192.168.1.3:9871) Running State Change CS_HANGUP > 2010-08-10 00:37:55.969068 [DEBUG] switch_core_state_machine.c:434 > (sofia/external/1899 at 192.168.1.3:9871) State HANGUP > 2010-08-10 00:37:55.969068 [DEBUG] mod_sofia.c:338 Channel > sofia/external/1899 at 192.168.1.3:9871 hanging up, cause: > INCOMPATIBLE_DESTINATION > 2010-08-10 00:37:55.969068 [DEBUG] mod_sofia.c:396 Sending BYE to > sofia/external/1899 at 192.168.1.3:9871 > nua: nua_bye: entering > nua(0x7f7e3408b0a0): sent signal r_bye > nua(0x7f7e3408b0a0): recv signal r_bye > nua: nua_stack_set_params: entering > soa_set_params(static::0x7f7f501de8e0, ...) called > soa_terminate(static::0x7f7f501de8e0) called > 2010-08-10 00:37:55.969068 [NOTICE] switch_ivr_bridge.c:710 Hangup > sofia/external/sipp at 192.168.1.3:8970 [CS_HIBERNATE] > [INCOMPATIBLE_DESTINATION] > soa_init_offer_answer(static::0x7f7f501de8e0) called > nta: selecting scheme sip > tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 > tport_resolve addrinfo = 192.168.1.114:5060 > tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 > 2010-08-10 00:37:55.970127 [DEBUG] switch_channel.c:1683 Send signal > sofia/external/sipp at 192.168.1.3:8970 [KILL] > 2010-08-10 00:37:55.970127 [DEBUG] mod_limit.c:195 > originate_disposition=[SUCCESS] > 2010-08-10 00:37:55.970127 [DEBUG] switch_core_session.c:932 Send > signal sofia/external/sipp at 192.168.1.3:8970 [BREAK] > tport_vsend(0x7f7f50004670): 619 bytes of 619 to udp/192.168.1.114:5060 > tport_vsend returned 619 > send 619 bytes to udp/[192.168.1.114]:5060 at 07:37:55.970197: > ------------------------------------------------------------------------ > 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:46 > sofia/external/1899 at 192.168.1.3:9871 Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > BYE sip:1899 at 192.168.1.3:9871 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKeFg2t9SeUmHSN > Route: > Max-Forwards: 70 > From: "sipp" ;tag=4tj1e3m89ZSBj > To: ;tag=1 > Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 > CSeq: 376874 BYE > Contact: > User-Agent: FreeSWITCH > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:398 > (sofia/external/sipp at 192.168.1.3:8970) Running State Change CS_HANGUP > Supported: timer, precondition, path, replaces > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 02010-08-10 00:37:55.970127 [DEBUG] > switch_core_state_machine.c:434 (sofia/external/1899 at 192.168.1.3:9871) > State HANGUP going to sleep > > > ------------------------------------------------------------------------ > nta: sent BYE (376874) to UDP/192.168.1.114:5060 > tport_pend(0x7f7f50004670): pending 0x7f7f50258650 for > udp/192.168.1.114:5070 (already 0) > 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:461 > Hangup Command decre_call_stat(44 60 43 59 14 429178 0): > > 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:434 > (sofia/external/sipp at 192.168.1.3:8970) State HANGUP > 2010-08-10 00:37:55.970127 [DEBUG] mod_sofia.c:338 Channel > sofia/external/sipp at 192.168.1.3:8970 hanging up, cause: > INCOMPATIBLE_DESTINATION > 2010-08-10 00:37:55.970127 [DEBUG] mod_sofia.c:396 Sending BYE to > sofia/external/sipp at 192.168.1.3:8970 > nua: nua_bye: entering > nua(0x7f7f5022abd0): sent signal r_bye > nua(0x7f7f5022abd0): recv signal r_bye > nua: nua_stack_set_params: entering > soa_set_params(static::0x7f7f50202440, ...) called > soa_terminate(static::0x7f7f50202440) called > soa_init_offer_answer(static::0x7f7f50202440) called > nta: selecting scheme sip > tport_tsend(0x7f7f50004670) tpn = */192.168.1.114:5060 > tport_resolve addrinfo = 192.168.1.114:5060 > tport_by_addrinfo(0x7f7f50004670): not found by name */192.168.1.114:5060 > tport_vsend(0x7f7f50004670): 637 bytes of 637 to udp/192.168.1.114:5060 > tport_vsend returned 637 > send 637 bytes to udp/[192.168.1.114]:5060 at 07:37:55.970869: > ------------------------------------------------------------------------ > BYE sip:sipp at 192.168.1.3:8970 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKFr9tv4ajrX7BH > Route: > Max-Forwards: 70 > From: sut ;tag=3HS8c834cQ3rp > To: sipp ;tag=1 > Call-ID: 1-9173 at 192.168.1.3 > CSeq: 376873 BYE > Contact: > User-Agent: FreeSWITCH > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > ------------------------------------------------------------------------ > nta: sent BYE (376873) to */192.168.1.114:5060 > tport_pend(0x7f7f50004670): pending 0x7f7f50254a40 for > udp/192.168.1.114:5070 (already 1) > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:46 > sofia/external/sipp at 192.168.1.3:8970 Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:434 > (sofia/external/sipp at 192.168.1.3:8970) State HANGUP going to sleep > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:476 > (sofia/external/sipp at 192.168.1.3:8970) State Change CS_HANGUP -> > CS_REPORTING > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:932 Send > signal sofia/external/sipp at 192.168.1.3:8970 [BREAK] > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:398 > (sofia/external/sipp at 192.168.1.3:8970) Running State Change > CS_REPORTING > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 > (sofia/external/sipp at 192.168.1.3:8970) State REPORTING > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:53 > sofia/external/sipp at 192.168.1.3:8970 Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 > (sofia/external/sipp at 192.168.1.3:8970) State REPORTING going to sleep > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:411 > (sofia/external/sipp at 192.168.1.3:8970) State Change CS_REPORTING -> > CS_DESTROY > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:1068 Session > 17184 (sofia/external/sipp at 192.168.1.3:8970) Locked, Waiting on > external entities > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:476 > (sofia/external/1899 at 192.168.1.3:9871) State Change CS_HANGUP -> > CS_REPORTING > 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1086 Session > 17184 (sofia/external/sipp at 192.168.1.3:8970) Ended > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:932 Send > signal sofia/external/1899 at 192.168.1.3:9871 [BREAK] > 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/external/sipp at 192.168.1.3:8970 [CS_DESTROY] > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:398 > (sofia/external/1899 at 192.168.1.3:9871) Running State Change > CS_REPORTING > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 > (sofia/external/1899 at 192.168.1.3:9871) State REPORTING > 2010-08-10 00:37:55.971183 [DEBUG] mod_cdr_bdb.c:139 Is number 1899 > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 > (sofia/external/sipp at 192.168.1.3:8970) State DESTROY > 2010-08-10 00:37:55.971183 [DEBUG] mod_sofia.c:255 > sofia/external/sipp at 192.168.1.3:8970 SOFIA DESTROY > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:60 > sofia/external/sipp at 192.168.1.3:8970 Standard DESTROY > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 > (sofia/external/sipp at 192.168.1.3:8970) State DESTROY going to sleep > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:53 > sofia/external/1899 at 192.168.1.3:9871 Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 > (sofia/external/1899 at 192.168.1.3:9871) State REPORTING going to sleep > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:411 > (sofia/external/1899 at 192.168.1.3:9871) State Change CS_REPORTING -> > CS_DESTROY > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:1068 Session > 17185 (sofia/external/1899 at 192.168.1.3:9871) Locked, Waiting on > external entities > 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1086 Session > 17185 (sofia/external/1899 at 192.168.1.3:9871) Ended > 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/external/1899 at 192.168.1.3:9871 [CS_DESTROY] > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 > (sofia/external/1899 at 192.168.1.3:9871) State DESTROY > 2010-08-10 00:37:55.971183 [DEBUG] mod_sofia.c:255 > sofia/external/1899 at 192.168.1.3:9871 SOFIA DESTROY > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:60 > sofia/external/1899 at 192.168.1.3:9871 Standard DESTROY > 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 > (sofia/external/1899 at 192.168.1.3:9871) State DESTROY going to sleep > nta: timer set next to 301 ms > nta: timer E fired, retransmit BYE (376874) > tport_release(0x7f7f50004670): 0x7f7f50258650 by 0x7f7f50275640 with (nil) > tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 > tport_resolve addrinfo = 192.168.1.114:5060 > tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 > tport_vsend(0x7f7f50004670): 619 bytes of 619 to udp/192.168.1.114:5060 > tport_vsend returned 619 > send 619 bytes to udp/[192.168.1.114]:5060 at 07:37:56.469550: > ------------------------------------------------------------------------ > BYE sip:1899 at 192.168.1.3:9871 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKeFg2t9SeUmHSN > Route: > Max-Forwards: 70 > From: "sipp" ;tag=4tj1e3m89ZSBj > To: ;tag=1 > Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 > CSeq: 376874 BYE > Contact: > User-Agent: FreeSWITCH > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > ------------------------------------------------------------------------ > nta: resent BYE (376874) to UDP/192.168.1.114:5060 > tport_pend(0x7f7f50004670): pending 0x7f7f50258650 for > udp/192.168.1.114:5070 (already 1) > nta_outgoing_timer: 1/2 resent, 0/3 tout, 0/1 term, 0/4 free > nta: timer set next to 1 ms > nta: timer E fired, retransmit BYE (376873) > tport_release(0x7f7f50004670): 0x7f7f50254a40 by 0x7f7f50247670 with (nil) > tport_tsend(0x7f7f50004670) tpn = */192.168.1.114:5060 > tport_resolve addrinfo = 192.168.1.114:5060 > tport_by_addrinfo(0x7f7f50004670): not found by name */192.168.1.114:5060 > tport_vsend(0x7f7f50004670): 637 bytes of 637 to udp/192.168.1.114:5060 > tport_vsend returned 637 > send 637 bytes to udp/[192.168.1.114]:5060 at 07:37:56.470547: > ------------------------------------------------------------------------ > BYE sip:sipp at 192.168.1.3:8970 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKFr9tv4ajrX7BH > Route: > Max-Forwards: 70 > From: sut ;tag=3HS8c834cQ3rp > To: sipp ;tag=1 > Call-ID: 1-9173 at 192.168.1.3 > CSeq: 376873 BYE > Contact: > User-Agent: FreeSWITCH > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > ------------------------------------------------------------------------ > nta: resent BYE (376873) to */192.168.1.114:5060 > tport_pend(0x7f7f50004670): pending 0x7f7f50254a40 for > udp/192.168.1.114:5070 (already 1) > nta_outgoing_timer: 1/2 resent, 0/3 tout, 0/1 term, 0/4 free > nta: timer set next to 638 ms > > On Fri, Aug 13, 2010 at 2:05 AM, Steven Ayre wrote: > > Do you have a packet trace available? > > > > -Steve > > > > On 12 August 2010 14:25, Pete Kay wrote: > >> Hi, > >> > >> I check tags and callid. It is the same dialog. Also, the invite is > >> accepted and I can see UAC does respond 200 OK . The freeswitch sends > >> out BYE after ACK. > >> > >> The problem I am seeing is that the re-invite triggers the dialplan > >> execution which based on its logic is responding with a 488 within the > >> dialplan using the respond app. When freeswitch receives the 488, it > >> can't recognize the dialog so it sends out BYE. > >> > >> Therefore, I think the way to solve this is to configure sofia so that > >> the invite won't trigger the execution of dialplan. > >> > >> Is there anyway to do that? > >> > >> Thanks, > >> P > >> > >> On Thu, Aug 12, 2010 at 9:00 PM, Brian West wrote: > >> > Chances are it is in fact a new dialog have you double checked the > >> > to/from tags and callid? Also if its without an SDP you would have to > >> > enable 3pcc on the profile to accept it possibly. > >> > > >> > /b > >> > > >> > On Aug 12, 2010, at 3:11 AM, Pete Kay wrote: > >> >> I am running a b2bua with freeswitch. It is fine until a Mitel UAS > >> >> starts sending INVITE without sdp and ACK with sdp. Freeswitch seems > >> >> to treat it as another dialog and sends it to dialplan handling. > >> >> > >> >> Within the dialplan, how can I recognize request as a re-invite and > >> >> possibly ignore it? > >> >> > >> >> Does anyone know how to resolve this problem? > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > >> >rs http://www.freeswitch.org > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From petedao at gmail.com Thu Aug 12 20:28:11 2010 From: petedao at gmail.com (Pete Kay) Date: Fri, 13 Aug 2010 11:28:11 +0800 Subject: [Freeswitch-users] re-invite problem In-Reply-To: <201008122229.47612.sos@sokhapkin.dyndns.org> References: <201008122229.47612.sos@sokhapkin.dyndns.org> Message-ID: Hi, I have tried setting 3pcc to true and proxy but the result is the same. P On Fri, Aug 13, 2010 at 10:29 AM, Sergey Okhapkin wrote: > Looks like you have no 3rd party call control enabled in FS settings... > > On Thursday 12 August 2010, Pete Kay wrote: >> Hi, >> >> Yes, here is the packet trace for the re-invite... >> >> recv 513 bytes from udp/[192.168.1.114]:5060 at 07:37:55.968109: >> ? ?------------------------------------------------------------------------ >> ? ?INVITE sip:sipp at 192.168.1.114:5070 SIP/2.0 >> ? ?Record-Route: >> ? ?Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 >> ? ?Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 >> ? ?From: sipp ;tag=1 >> ? ?To: ;tag=4tj1e3m89ZSBj >> ? ?Contact: sip:1899 at 192.168.1.3:9871 >> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >> ? ?CSeq: 1 INVITE >> ? ?Session-Expires: 120;refresher=uac >> ? ?Min-SE: 90 >> ? ?Supported: 100rel, timer >> ? ?Max-Forwards: 69 >> ? ?Content-Length: 0 >> >> ? ?------------------------------------------------------------------------ >> tport_deliver(0x7f7f50004670): msg 0x7f7f502518d0 (513 bytes) from >> udp/192.168.1.114:5070/sip next=(nil) >> nta: received INVITE sip:sipp at 192.168.1.114:5070 SIP/2.0 (CSeq 1) >> nta: canonizing sip:sipp at 192.168.1.114:5070 with contact >> nta: INVITE (1) going to existing leg >> nta: timer shortened to 200 ms >> nua: nua_stack_process_request: entering >> soa_init_offer_answer(static::0x7f7f501de8e0) called >> tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 >> tport_resolve addrinfo = 192.168.1.114:5060 >> tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 >> tport_vsend(0x7f7f50004670): 387 bytes of 387 to udp/192.168.1.114:5060 >> tport_vsend returned 387 >> send 387 bytes to udp/[192.168.1.114]:5060 at 07:37:55.968290: >> ? ?------------------------------------------------------------------------ >> ? ?SIP/2.0 100 Trying >> ? ?Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 >> ? ?Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 >> ? ?Record-Route: >> ? ?From: sipp ;tag=1 >> ? ?To: ;tag=4tj1e3m89ZSBj >> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >> ? ?CSeq: 1 INVITE >> ? ?User-Agent: FreeSWITCH >> ? ?Content-Length: 0 >> >> ? ?------------------------------------------------------------------------ >> nta: sent 100 Trying for INVITE (1) >> nua(0x7f7e3408b0a0): event i_invite 100 Trying >> nua: nua_application_event: entering >> nua: nua_handle_magic: entering >> nua(0x7f7e3408b0a0): ready call updated: received >> nua(0x7f7e3408b0a0): event i_state 100 Trying >> nua: nua_application_event: entering >> 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel >> sofia/external/1899 at 192.168.1.3:9871 entering state [received][100] >> nua: nua_respond: entering >> nua(0x7f7e3408b0a0): sent signal r_respond >> nua: nua_handle_magic: entering >> nua(0x7f7e3408b0a0): recv signal r_respond 200 OK >> nua: nua_stack_set_params: entering >> soa_set_params(static::0x7f7f501de8e0, ...) called >> nua: nua_invite_server_respond: entering >> soa_generate_offer(static::0x7f7f501de8e0, 0) called >> soa_static_offer_answer_action(0x7f7f501de8e0, soa_generate_offer): called >> soa_sdp_mode_set(0x7f7f50226fe0, (nil), ""): called >> soa_get_local_sdp(static::0x7f7f501de8e0, [(nil)], [0x7f7f60520d30], >> [0x7f7f60520d3c]) called >> tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 >> tport_resolve addrinfo = 192.168.1.114:5060 >> tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 >> tport_vsend(0x7f7f50004670): 944 bytes of 944 to udp/192.168.1.114:5060 >> tport_vsend returned 944 >> send 944 bytes to udp/[192.168.1.114]:5060 at 07:37:55.968539: >> ? ?------------------------------------------------------------------------ >> ? ?SIP/2.0 200 OK >> ? ?Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 >> ? ?Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 >> ? ?Record-Route: >> ? ?From: sipp ;tag=1 >> ? ?To: ;tag=4tj1e3m89ZSBj >> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >> ? ?CSeq: 1 INVITE >> ? ?Contact: >> ? ?User-Agent: FreeSWITCH >> ? ?Accept: application/sdp >> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >> ? ?Require: timer >> ? ?Supported: timer, precondition, path, replaces >> ? ?Session-Expires: 120;refresher=uac >> ? ?Min-SE: 120 >> ? ?Content-Type: application/sdp >> ? ?Content-Disposition: session >> ? ?Content-Length: 213 >> >> ? ?v=0 >> ? ?o=user1 53655765 2353687637 IN IP4 192.168.1.114 >> ? ?s=- >> ? ?c=IN IP4 192.168.1.114 >> ? ?t=0 0 >> ? ?m=audio 27938 RTP/AVP 0 18 101 >> ? ?a=rtpmap:0 PCMU/8000 >> ? ?a=rtpmap:18 G729/8000 >> ? ?a=rtpmap:101 telephone-event/8000 >> ? ?a=ptime:20 >> ? ?------------------------------------------------------------------------ >> nta: sent 200 OK for INVITE (1) >> nua(0x7f7e3408b0a0): ready call updated: completed sent offer >> soa_get_local_sdp(static::0x7f7f501de8e0, [0x7f7f60520e08], >> [0x7f7f60520e00], [(nil)]) called >> nua(0x7f7e3408b0a0): event i_state 200 OK >> nua: nua_application_event: entering >> 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel >> sofia/external/1899 at 192.168.1.3:9871 entering state [completed][200] >> nua: nua_handle_magic: entering >> tport_wakeup_pri(0x7f7f50004670): events IN >> tport_recv_event(0x7f7f50004670) >> tport_recv_iovec(0x7f7f50004670) msg 0x7f7f5022be60 from >> (udp/192.168.1.114:5070) has 525 bytes, veclen = 1 >> recv 525 bytes from udp/[192.168.1.114]:5060 at 07:37:55.968843: >> ? ?------------------------------------------------------------------------ >> ? ?ACK sip:sipp at 192.168.1.114:5070 SIP/2.0 >> ? ?Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.2 >> ? ?Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-8 >> ? ?From: sipp ;tag=1 >> ? ?To: ;tag=4tj1e3m89ZSBj >> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >> ? ?CSeq: 1 ACK >> ? ?Max-Forwards: 69 >> ? ?Content-Type: application/sdp >> ? ?Content-Length: ? 133 >> >> ? ?v=0 >> ? ?o=user1 53655765 2353687637 IN IP4 192.168.1.3 >> ? ?s=- >> ? ?c=IN IP4 192.168.1.3 >> ? ?t=0 0 >> ? ?m=audio 6000 RTP/AVP 0 >> ? ?a=rtpmap:0 PCMU/8000 >> ? ?------------------------------------------------------------------------ >> tport_deliver(0x7f7f50004670): msg 0x7f7f5022be60 (525 bytes) from >> udp/192.168.1.114:5070/sip next=(nil) >> nta: received ACK sip:sipp at 192.168.1.114:5070 SIP/2.0 (CSeq 1) >> nta: ACK (1) is going to INVITE (1) >> nua: process_ack_or_cancel: entering >> soa_set_remote_sdp(static::0x7f7f501de8e0, (nil), 0x7f7f501eb298, 133) >> ?called soa_process_answer(static::0x7f7f501de8e0) called >> soa_static_offer_answer_action(0x7f7f501de8e0, soa_process_answer): called >> soa_sdp_mode_set(0x7f7f50226fe0, 0x7f7f50245750, ""): called >> soa_activate(static::0x7f7f501de8e0, (nil)) called >> soa_clear_remote_sdp(static::0x7f7f501de8e0) called >> nua(0x7f7e3408b0a0): event i_ack 200 OK >> nua: nua_application_event: entering >> nua: nua_handle_magic: entering >> nua(0x7f7e3408b0a0): ready call updated: ready received answer >> soa_get_remote_sdp(static::0x7f7f501de8e0, [0x7f7f60520968], >> [0x7f7f60520960], [(nil)]) called >> soa_get_params(static::0x7f7f501de8e0, ...) called >> nua(0x7f7e3408b0a0): event i_state 200 OK >> nua: nua_application_event: entering >> 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel >> sofia/external/1899 at 192.168.1.3:9871 entering state [ready][200] >> 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3296 Remote SDP: >> v=0 >> o=user1 53655765 2353687637 IN IP4 192.168.1.3 >> s=- >> c=IN IP4 192.168.1.3 >> t=0 0 >> m=audio 6000 RTP/AVP 0 >> a=rtpmap:0 PCMU/8000 >> >> nua: nua_respond: entering >> nua(0x7f7e3408b0a0): sent signal r_respond >> 2010-08-10 00:37:55.968013 [NOTICE] sofia.c:3712 Hangup >> sofia/external/1899 at 192.168.1.3:9871 [CS_HIBERNATE] >> [INCOMPATIBLE_DESTINATION] >> 2010-08-10 00:37:55.969068 [DEBUG] switch_channel.c:1683 Send signal >> sofia/external/1899 at 192.168.1.3:9871 [KILL] >> 2010-08-10 00:37:55.969068 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/1899 at 192.168.1.3:9871 [BREAK] >> nua: nua_handle_magic: entering >> nua(0x7f7e3408b0a0): event i_active 200 Call active >> nua: nua_application_event: entering >> nua(): refresh session after 88 seconds (in [88..88]) >> nua(0x7f7e3408b0a0): recv signal r_respond 488 Not Acceptable Here >> nua(0x7f7e3408b0a0): event i_error 500 Responding to a Non-Existing Request >> nua: nua_application_event: entering >> 2010-08-10 00:37:55.969068 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/1899 at 192.168.1.3:9871) Running State Change CS_HANGUP >> 2010-08-10 00:37:55.969068 [DEBUG] switch_core_state_machine.c:434 >> (sofia/external/1899 at 192.168.1.3:9871) State HANGUP >> 2010-08-10 00:37:55.969068 [DEBUG] mod_sofia.c:338 Channel >> sofia/external/1899 at 192.168.1.3:9871 hanging up, cause: >> INCOMPATIBLE_DESTINATION >> 2010-08-10 00:37:55.969068 [DEBUG] mod_sofia.c:396 Sending BYE to >> sofia/external/1899 at 192.168.1.3:9871 >> nua: nua_bye: entering >> nua(0x7f7e3408b0a0): sent signal r_bye >> nua(0x7f7e3408b0a0): recv signal r_bye >> nua: nua_stack_set_params: entering >> soa_set_params(static::0x7f7f501de8e0, ...) called >> soa_terminate(static::0x7f7f501de8e0) called >> 2010-08-10 00:37:55.969068 [NOTICE] switch_ivr_bridge.c:710 Hangup >> sofia/external/sipp at 192.168.1.3:8970 [CS_HIBERNATE] >> [INCOMPATIBLE_DESTINATION] >> soa_init_offer_answer(static::0x7f7f501de8e0) called >> nta: selecting scheme sip >> tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 >> tport_resolve addrinfo = 192.168.1.114:5060 >> tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 >> 2010-08-10 00:37:55.970127 [DEBUG] switch_channel.c:1683 Send signal >> sofia/external/sipp at 192.168.1.3:8970 [KILL] >> 2010-08-10 00:37:55.970127 [DEBUG] mod_limit.c:195 >> originate_disposition=[SUCCESS] >> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/sipp at 192.168.1.3:8970 [BREAK] >> tport_vsend(0x7f7f50004670): 619 bytes of 619 to udp/192.168.1.114:5060 >> tport_vsend returned 619 >> send 619 bytes to udp/[192.168.1.114]:5060 at 07:37:55.970197: >> ? ?------------------------------------------------------------------------ >> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:46 >> sofia/external/1899 at 192.168.1.3:9871 Standard HANGUP, cause: >> INCOMPATIBLE_DESTINATION >> ? ?BYE sip:1899 at 192.168.1.3:9871 SIP/2.0 >> ? ?Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKeFg2t9SeUmHSN >> ? ?Route: >> ? ?Max-Forwards: 70 >> ? ?From: "sipp" ;tag=4tj1e3m89ZSBj >> ? ?To: ;tag=1 >> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >> ? ?CSeq: 376874 BYE >> ? ?Contact: >> ? ?User-Agent: FreeSWITCH >> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/sipp at 192.168.1.3:8970) Running State Change CS_HANGUP >> ? ?Supported: timer, precondition, path, replaces >> ? ?Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >> ? ?Content-Length: 02010-08-10 00:37:55.970127 [DEBUG] >> switch_core_state_machine.c:434 (sofia/external/1899 at 192.168.1.3:9871) >> State HANGUP going to sleep >> >> >> ? ?------------------------------------------------------------------------ >> nta: sent BYE (376874) to UDP/192.168.1.114:5060 >> tport_pend(0x7f7f50004670): pending 0x7f7f50258650 for >> udp/192.168.1.114:5070 (already 0) >> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:461 >> Hangup Command decre_call_stat(44 60 43 59 14 429178 0): >> >> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:434 >> (sofia/external/sipp at 192.168.1.3:8970) State HANGUP >> 2010-08-10 00:37:55.970127 [DEBUG] mod_sofia.c:338 Channel >> sofia/external/sipp at 192.168.1.3:8970 hanging up, cause: >> INCOMPATIBLE_DESTINATION >> 2010-08-10 00:37:55.970127 [DEBUG] mod_sofia.c:396 Sending BYE to >> sofia/external/sipp at 192.168.1.3:8970 >> nua: nua_bye: entering >> nua(0x7f7f5022abd0): sent signal r_bye >> nua(0x7f7f5022abd0): recv signal r_bye >> nua: nua_stack_set_params: entering >> soa_set_params(static::0x7f7f50202440, ...) called >> soa_terminate(static::0x7f7f50202440) called >> soa_init_offer_answer(static::0x7f7f50202440) called >> nta: selecting scheme sip >> tport_tsend(0x7f7f50004670) tpn = */192.168.1.114:5060 >> tport_resolve addrinfo = 192.168.1.114:5060 >> tport_by_addrinfo(0x7f7f50004670): not found by name */192.168.1.114:5060 >> tport_vsend(0x7f7f50004670): 637 bytes of 637 to udp/192.168.1.114:5060 >> tport_vsend returned 637 >> send 637 bytes to udp/[192.168.1.114]:5060 at 07:37:55.970869: >> ? ?------------------------------------------------------------------------ >> ? ?BYE sip:sipp at 192.168.1.3:8970 SIP/2.0 >> ? ?Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKFr9tv4ajrX7BH >> ? ?Route: >> ? ?Max-Forwards: 70 >> ? ?From: sut ;tag=3HS8c834cQ3rp >> ? ?To: sipp ;tag=1 >> ? ?Call-ID: 1-9173 at 192.168.1.3 >> ? ?CSeq: 376873 BYE >> ? ?Contact: >> ? ?User-Agent: FreeSWITCH >> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >> ? ?Supported: timer, precondition, path, replaces >> ? ?Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >> ? ?Content-Length: 0 >> >> ? ?------------------------------------------------------------------------ >> nta: sent BYE (376873) to */192.168.1.114:5060 >> tport_pend(0x7f7f50004670): pending 0x7f7f50254a40 for >> udp/192.168.1.114:5070 (already 1) >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:46 >> sofia/external/sipp at 192.168.1.3:8970 Standard HANGUP, cause: >> INCOMPATIBLE_DESTINATION >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:434 >> (sofia/external/sipp at 192.168.1.3:8970) State HANGUP going to sleep >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:476 >> (sofia/external/sipp at 192.168.1.3:8970) State Change CS_HANGUP -> >> CS_REPORTING >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/sipp at 192.168.1.3:8970 [BREAK] >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/sipp at 192.168.1.3:8970) Running State Change >> CS_REPORTING >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 >> (sofia/external/sipp at 192.168.1.3:8970) State REPORTING >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:53 >> sofia/external/sipp at 192.168.1.3:8970 Standard REPORTING, cause: >> INCOMPATIBLE_DESTINATION >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 >> (sofia/external/sipp at 192.168.1.3:8970) State REPORTING going to sleep >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:411 >> (sofia/external/sipp at 192.168.1.3:8970) State Change CS_REPORTING -> >> CS_DESTROY >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:1068 Session >> 17184 (sofia/external/sipp at 192.168.1.3:8970) Locked, Waiting on >> external entities >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:476 >> (sofia/external/1899 at 192.168.1.3:9871) State Change CS_HANGUP -> >> CS_REPORTING >> 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1086 Session >> 17184 (sofia/external/sipp at 192.168.1.3:8970) Ended >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/1899 at 192.168.1.3:9871 [BREAK] >> 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1088 Close >> Channel sofia/external/sipp at 192.168.1.3:8970 [CS_DESTROY] >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:398 >> (sofia/external/1899 at 192.168.1.3:9871) Running State Change >> CS_REPORTING >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 >> (sofia/external/1899 at 192.168.1.3:9871) State REPORTING >> 2010-08-10 00:37:55.971183 [DEBUG] mod_cdr_bdb.c:139 Is number 1899 >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 >> (sofia/external/sipp at 192.168.1.3:8970) State DESTROY >> 2010-08-10 00:37:55.971183 [DEBUG] mod_sofia.c:255 >> sofia/external/sipp at 192.168.1.3:8970 SOFIA DESTROY >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:60 >> sofia/external/sipp at 192.168.1.3:8970 Standard DESTROY >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 >> (sofia/external/sipp at 192.168.1.3:8970) State DESTROY going to sleep >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:53 >> sofia/external/1899 at 192.168.1.3:9871 Standard REPORTING, cause: >> INCOMPATIBLE_DESTINATION >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 >> (sofia/external/1899 at 192.168.1.3:9871) State REPORTING going to sleep >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:411 >> (sofia/external/1899 at 192.168.1.3:9871) State Change CS_REPORTING -> >> CS_DESTROY >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:1068 Session >> 17185 (sofia/external/1899 at 192.168.1.3:9871) Locked, Waiting on >> external entities >> 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1086 Session >> 17185 (sofia/external/1899 at 192.168.1.3:9871) Ended >> 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1088 Close >> Channel sofia/external/1899 at 192.168.1.3:9871 [CS_DESTROY] >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 >> (sofia/external/1899 at 192.168.1.3:9871) State DESTROY >> 2010-08-10 00:37:55.971183 [DEBUG] mod_sofia.c:255 >> sofia/external/1899 at 192.168.1.3:9871 SOFIA DESTROY >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:60 >> sofia/external/1899 at 192.168.1.3:9871 Standard DESTROY >> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 >> (sofia/external/1899 at 192.168.1.3:9871) State DESTROY going to sleep >> nta: timer set next to 301 ms >> nta: timer E fired, retransmit BYE (376874) >> tport_release(0x7f7f50004670): 0x7f7f50258650 by 0x7f7f50275640 with (nil) >> tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 >> tport_resolve addrinfo = 192.168.1.114:5060 >> tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 >> tport_vsend(0x7f7f50004670): 619 bytes of 619 to udp/192.168.1.114:5060 >> tport_vsend returned 619 >> send 619 bytes to udp/[192.168.1.114]:5060 at 07:37:56.469550: >> ? ?------------------------------------------------------------------------ >> ? ?BYE sip:1899 at 192.168.1.3:9871 SIP/2.0 >> ? ?Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKeFg2t9SeUmHSN >> ? ?Route: >> ? ?Max-Forwards: 70 >> ? ?From: "sipp" ;tag=4tj1e3m89ZSBj >> ? ?To: ;tag=1 >> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >> ? ?CSeq: 376874 BYE >> ? ?Contact: >> ? ?User-Agent: FreeSWITCH >> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >> ? ?Supported: timer, precondition, path, replaces >> ? ?Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >> ? ?Content-Length: 0 >> >> ? ?------------------------------------------------------------------------ >> nta: resent BYE (376874) to UDP/192.168.1.114:5060 >> tport_pend(0x7f7f50004670): pending 0x7f7f50258650 for >> udp/192.168.1.114:5070 (already 1) >> nta_outgoing_timer: 1/2 resent, 0/3 tout, 0/1 term, 0/4 free >> nta: timer set next to 1 ms >> nta: timer E fired, retransmit BYE (376873) >> tport_release(0x7f7f50004670): 0x7f7f50254a40 by 0x7f7f50247670 with (nil) >> tport_tsend(0x7f7f50004670) tpn = */192.168.1.114:5060 >> tport_resolve addrinfo = 192.168.1.114:5060 >> tport_by_addrinfo(0x7f7f50004670): not found by name */192.168.1.114:5060 >> tport_vsend(0x7f7f50004670): 637 bytes of 637 to udp/192.168.1.114:5060 >> tport_vsend returned 637 >> send 637 bytes to udp/[192.168.1.114]:5060 at 07:37:56.470547: >> ? ?------------------------------------------------------------------------ >> ? ?BYE sip:sipp at 192.168.1.3:8970 SIP/2.0 >> ? ?Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKFr9tv4ajrX7BH >> ? ?Route: >> ? ?Max-Forwards: 70 >> ? ?From: sut ;tag=3HS8c834cQ3rp >> ? ?To: sipp ;tag=1 >> ? ?Call-ID: 1-9173 at 192.168.1.3 >> ? ?CSeq: 376873 BYE >> ? ?Contact: >> ? ?User-Agent: FreeSWITCH >> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >> ? ?Supported: timer, precondition, path, replaces >> ? ?Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >> ? ?Content-Length: 0 >> >> ? ?------------------------------------------------------------------------ >> nta: resent BYE (376873) to */192.168.1.114:5060 >> tport_pend(0x7f7f50004670): pending 0x7f7f50254a40 for >> udp/192.168.1.114:5070 (already 1) >> nta_outgoing_timer: 1/2 resent, 0/3 tout, 0/1 term, 0/4 free >> nta: timer set next to 638 ms >> >> On Fri, Aug 13, 2010 at 2:05 AM, Steven Ayre wrote: >> > Do you have a packet trace available? >> > >> > -Steve >> > >> > On 12 August 2010 14:25, Pete Kay wrote: >> >> Hi, >> >> >> >> I check tags and callid. ?It is the same dialog. ?Also, the invite is >> >> accepted and I can see UAC does respond 200 OK . ?The freeswitch sends >> >> out BYE after ACK. >> >> >> >> The problem I am seeing is that the re-invite triggers the dialplan >> >> execution which based on its logic is responding with a 488 within the >> >> dialplan using the respond app. ?When freeswitch receives the 488, it >> >> can't recognize the dialog so it sends out BYE. >> >> >> >> Therefore, I think the way to solve this is to configure sofia so that >> >> the invite won't trigger the execution of dialplan. >> >> >> >> Is there anyway to do that? >> >> >> >> Thanks, >> >> P >> >> >> >> On Thu, Aug 12, 2010 at 9:00 PM, Brian West wrote: >> >> > Chances are it is in fact a new dialog have you double checked the >> >> > to/from tags and callid? ?Also if its without an SDP you would have to >> >> > enable 3pcc on the profile to accept it possibly. >> >> > >> >> > /b >> >> > >> >> > On Aug 12, 2010, at 3:11 AM, Pete Kay wrote: >> >> >> I am running a b2bua with freeswitch. ?It is fine until a Mitel UAS >> >> >> starts sending INVITE without sdp and ACK with sdp. Freeswitch seems >> >> >> to treat it as another dialog and sends it to dialplan handling. >> >> >> >> >> >> Within the dialplan, how can I recognize request as a re-invite and >> >> >> possibly ignore it? >> >> >> >> >> >> Does anyone know how to resolve this problem? >> >> > >> >> > _______________________________________________ >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> >> >rs http://www.freeswitch.org >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rupa at rupa.com Thu Aug 12 20:46:33 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 12 Aug 2010 22:46:33 -0500 Subject: [Freeswitch-users] A further question about originate In-Reply-To: <4C645E2F.3040608@ewetel.de> References: <4C645E2F.3040608@ewetel.de> Message-ID: The var doesn't survive over sip. The first example initiates a sip call that is then handled by FS. No var. The second one is handled internally. On Thu, Aug 12, 2010 at 3:48 PM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > I try to set a custom channel variable within originate command as > described here: > > http://wiki.freeswitch.org/wiki/Mod_commands > > [...] > Note: you can set any channel variable, even custom ones. Use single > quotes to enclose values with spaces, commas, etc. > > originate {my_own_var=my_value}sofia/mydomain.com/that.ext at 1.2.3.4 > 15555551212 > originate {my_own_var='my value'}sofia/mydomain.com/that.ext at 1.2.3.4 > 15555551212 > > [...] > > > > My command is like this: > > originate {my_var=my_data}sofia/internal/2850 at 85.16.246.6 &park > > When I call info in dialplan, the variable is missing > > > When I do > originate {my_var=my_data}user/2850 at 85.16.246.6 &park > > the variable is there! > > > Any reasons for this? > > regards > helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFMZF4v4tZeNddg3dwRAiwOAJoCGHZEZCrGfDDUPDB10p4p9GCzAQCfbtKS > o+9fmLXxswvtTFebbnoWbuM= > =A/i/ > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100812/6b9696fd/attachment.html From mnhassan at usa.net Thu Aug 12 21:06:11 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Fri, 13 Aug 2010 10:06:11 +0600 Subject: [Freeswitch-users] mod_nibblebill line #487 Message-ID: Hi, I'm using the latest git, and the mod_nibblebill.c was last edited on Jul/30. On line #487, there is this: switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Last successful billing time was %s\n", date); But, to me, it seems like, the "date" gets assigned only on #490-491. Am I reading this wrong? Regards HASSAN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/e03d69e5/attachment.html From odermann at googlemail.com Fri Aug 13 01:24:51 2010 From: odermann at googlemail.com (Dennis) Date: Fri, 13 Aug 2010 10:24:51 +0200 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> Message-ID: thanks for your help! just as a short info inbetween, while we are still trying to fix the problem. the cirpack of our carrier always receives dtmf-inputs as inband tones (because, at least in germany, all dtmf-inputs are sent over the landline as inband tones). because we always had problems with the carriers cirpack and fs in conjunction with inband tones (sometimes they where sent two or three times), our carrier added rfc-2833 to their cirpack. now the carrier sends both: rfc-2833 AND inband tones. the problem is, that the carrier tells us, that the cirpack can not filter the inband tones, so that they just can send rfc-2833. but they tell us, that they add some stops (or something like this) to the inband tones, so that the inband tones are not functional anymore. but this does not seem to work correctly. if we listen to dtmf-inputs only on fs, we do not have any problems, because fs ignores the inband tones and only reacts to the rfc-2833 signal. therefor we never had any problems. now, we have a project, where not fs listens to the dtmf-inputs, but a callcenter on the other side (outbound). therefor the cirpack has to send clean dtmf-inputs to the callcenter. but we believe, that the cirpack on the outgoing side is receiving his own inband tones AND rfc-2833 signals from the incoming side of the cirpack. so the cirpack on the outgoing side send his own inband tones PLUS the converted rfc-2833 signals to the other side (callcenter). we have to go on testing. we spoke with someone from nokia about their iwsd and they told us, that they filter all inband tones, so that we only receive rfc-2833. we will do some testing with them. but i wonder, why the nokia iwsd can filter inband tones and the cirpack can't. any ideas? will keep you informed, what we found out. From steveayre at gmail.com Fri Aug 13 02:01:17 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 13 Aug 2010 10:01:17 +0100 Subject: [Freeswitch-users] FreeSwitch - Skype4COM or SkypeKIT we are using? In-Reply-To: References: Message-ID: On 12 August 2010 21:16, Shamun toha md wrote: > NOTE: Please have a look: > http://forum.skype.com/index.php?showtopic=637293&st=0&p=2897693&#entry2897693 > > I am completely now confused: > > Does that mean Skype Module in FreeSwitch will be dead or is dead? > No. A Skype module will always be required for FreeSWITCH to talk to Skype, and the existing one will continue to work just fine. > Does that mean Skype Module in FreeSwitch, can now handle 100 to 300 or 999 > simultaneous calls? > The Skype module is currently unchanged. Your concurrent calls will be the same as before. > Does that mean Skype Module in FreeSwitch, will be never needed? > No, a Skype endpoint module will always be required for FreeSWITCH to talk to Skype. It *may* mean that mod_skyopen will end up being rewritten to use the SDK which would have advantages - removing the reliance on running a separate client, allowing it to run on non-Linux systems and increasing the number of channels possible. But the cost, terms and licensing of the SDK isn't clear yet and until they are it won't be possible to know whether this will be possible. I'm not the developer of mod_skyopen - whoever is would be able to give the best answer on how likely it is that mod_skypopen will be updated to use the SDK. > Does that mean Skype Module in FreeSwitch, will need to be completely renew > after running it for months? > It means there might be a new version to upgrade to at some point. But you'll want to be upgrading FreeSWITCH to get bugfixes periodically anyway so that's not an issue. If you don't upgrade the previous version will continue working just fine. I doubt the interface would change with an upgrade (the developers will strive to avoid any big changes so as to be backwards compatible), so it'd just be a case of recompiling and updating the mod_skypopen configuration. > > > > Thank you > Best regards > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/42bbd134/attachment.html From a.afzali2003 at gmail.com Fri Aug 13 02:26:32 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Fri, 13 Aug 2010 13:56:32 +0430 Subject: [Freeswitch-users] FIFO queue callers ID: how to set to a real-life callers ID, not a static entry? In-Reply-To: <4C62FB3D.4030707@isptelecom.net> References: <4C62FB3D.4030707@isptelecom.net> Message-ID: Victor, If your agent's SIP client supports SIP UPDATE message, fifo will update the caller's ID after agent answer it's call, although I see in latest code that with some modified behavior, FIFO sends the actual caller's ID when it calls agent. -- afshin On Thu, Aug 12, 2010 at 12:04 AM, Victor Chukalovskiy wrote: > Dear FreeSWITCH community, > > I'm currently putting FreeSWITCH into production environment (great > experience so far). > We want agent of a fifo queue to be able to see caller's ID number of the > customer calling. > > The default FS behaviour is to pass queue name as a callers ID name and > callers ID number. > > I understand that putting {origination_caller_id_name=foo,origination_caller_id_number=123} > into Members definition string in fifo.conf.xml will override default static > callers name/ID with another static callers name/ID. > > But what is really needed is that fifo member (agent) is presented with > real-life callers ID number of the calling party. > > Any help on how to do this is much appreciated. > -Victor > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/780ec57b/attachment.html From gmaruzz at celliax.org Fri Aug 13 04:49:04 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 13 Aug 2010 13:49:04 +0200 Subject: [Freeswitch-users] FreeSwitch - Skype4COM or SkypeKIT we are using? In-Reply-To: References: Message-ID: I'm one of the developers of mod_skypopen, and I concur completely with Steven. mod_skypopen will continue to live and strive, and will continue to be exactly as in its wiki page. As Public Enemy says: "Don't believe the hype!" -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From tculjaga at gmail.com Fri Aug 13 04:52:53 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 13 Aug 2010 13:52:53 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: <3D922F11-C874-4D4B-893F-3B449A55EB02@freeswitch.org> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <3D922F11-C874-4D4B-893F-3B449A55EB02@freeswitch.org> Message-ID: On Thu, Aug 12, 2010 at 10:24 PM, Brian West wrote: > Nothing at all is fishy. He said he wanted to transcode and he wanted to > record. That would require more licenses. I specifically answered the > question he asked. > > aaaaa :) well, yes for transcoding of course you need that :)) i misunderstood the thing :) Apologize, T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/3f9062b6/attachment-0001.html From lists at infosecurity.ch Fri Aug 13 05:29:59 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Fri, 13 Aug 2010 14:29:59 +0200 Subject: [Freeswitch-users] Limits on number of concurrent registered users? Message-ID: <4C653AC7.8010009@infosecurity.ch> Hi all, i am preparing a setup where i would have a huge number of connected users via SIP/TLS . The amount of concurrent calls will not be such high. However he amount of concurrently registered users could be even of 15.000 . Are there some specific internal design/architectural limits of FreeSWITCH on the number of concurrent users? Fabio p.s. we'll try to do some stress testing next week From ken at ukgb.net Fri Aug 13 07:41:01 2010 From: ken at ukgb.net (Ken Gillett) Date: Fri, 13 Aug 2010 15:41:01 +0100 Subject: [Freeswitch-users] Mobile extensions Message-ID: I was wondering about the use of mobile phones running a SIP client. If they are configured as a FreeSwitch extension, they stop working once away from the local LAN. But if they ALSO register to the VOIP provider's (external) SIP server, when they ARE on the local LAN, they will get the calls twice. If they ONLY register to the external SIP server, you lose the power and flexibility of managing them as extensions. I cannot be unique in wanting to use Mobiles as extensions. How do others get around this? Ken G i l l e t t _/_/_/_/_/_/_/_/ From t.mahe at telemaque.fr Fri Aug 13 07:48:34 2010 From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=) Date: Fri, 13 Aug 2010 16:48:34 +0200 Subject: [Freeswitch-users] Mobile extensions In-Reply-To: References: Message-ID: <4C655B42.9040504@telemaque.fr> Hi Ken, vpn on my phone, always registered to internal profile... :) On 13/08/2010 16:41, Ken Gillett wrote: > I was wondering about the use of mobile phones running a SIP client. If they are configured as a FreeSwitch extension, they stop working once away from the local LAN. But if they ALSO register to the VOIP provider's (external) SIP server, when they ARE on the local LAN, they will get the calls twice. If they ONLY register to the external SIP server, you lose the power and flexibility of managing them as extensions. > > I cannot be unique in wanting to use Mobiles as extensions. How do others get around this? > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Cordialement, ********************************** * Tristan Mah? * * T?l?maque - Service Voix * * Tel: +33 4.92.90.99.85 * * Mob: +33 6.24.16.43.01 * * Fax: +33 4.92.90.91.46 * ********************************** From brian at freeswitch.org Fri Aug 13 07:53:18 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Aug 2010 09:53:18 -0500 Subject: [Freeswitch-users] Mobile extensions In-Reply-To: <4C655B42.9040504@telemaque.fr> References: <4C655B42.9040504@telemaque.fr> Message-ID: Happen to be running on vmware? Also is the registration in the db twice? /b On Aug 13, 2010, at 9:48 AM, Tristan Mah? wrote: > Hi Ken, > > vpn on my phone, always registered to internal profile... :) From tgraziano at myitdepartment.net Fri Aug 13 07:59:21 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Fri, 13 Aug 2010 10:59:21 -0400 Subject: [Freeswitch-users] Mobile extensions In-Reply-To: <4C655B42.9040504@telemaque.fr> References: <4C655B42.9040504@telemaque.fr> Message-ID: or make sure you firewall in front of FS allows remote users to come through... assuming your dns records are setup properly it will work by name inside or out... On Fri, Aug 13, 2010 at 10:48 AM, Tristan Mah? wrote: > Hi Ken, > > vpn on my phone, always registered to internal profile... :) > > On 13/08/2010 16:41, Ken Gillett wrote: > > I was wondering about the use of mobile phones running a SIP client. If > they are configured as a FreeSwitch extension, they stop working once away > from the local LAN. But if they ALSO register to the VOIP provider's > (external) SIP server, when they ARE on the local LAN, they will get the > calls twice. If they ONLY register to the external SIP server, you lose the > power and flexibility of managing them as extensions. > > > > I cannot be unique in wanting to use Mobiles as extensions. How do others > get around this? > > > > > > Ken G i l l e t t > > > > _/_/_/_/_/_/_/_/ > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Cordialement, > > ********************************** > * Tristan Mah? * > * T?l?maque - Service Voix * > * Tel: +33 4.92.90.99.85 * > * Mob: +33 6.24.16.43.01 * > * Fax: +33 4.92.90.91.46 * > ********************************** > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/9d4018ee/attachment.html From freeswitch-list at puzzled.xs4all.nl Fri Aug 13 08:00:56 2010 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 13 Aug 2010 17:00:56 +0200 Subject: [Freeswitch-users] Mobile extensions In-Reply-To: <4C655B42.9040504@telemaque.fr> References: <4C655B42.9040504@telemaque.fr> Message-ID: <4C655E28.40408@puzzled.xs4all.nl> On 08/13/2010 04:48 PM, Tristan Mah? wrote: > Hi Ken, > > vpn on my phone, always registered to internal profile... :) All 2 hours before the phone's battery dies :) Why not just do SIP/TLS? regards, Patrick > On 13/08/2010 16:41, Ken Gillett wrote: >> I was wondering about the use of mobile phones running a SIP client. If they are configured as a FreeSwitch extension, they stop working once away from the local LAN. But if they ALSO register to the VOIP provider's (external) SIP server, when they ARE on the local LAN, they will get the calls twice. If they ONLY register to the external SIP server, you lose the power and flexibility of managing them as extensions. >> >> I cannot be unique in wanting to use Mobiles as extensions. How do others get around this? From t.mahe at telemaque.fr Fri Aug 13 08:04:39 2010 From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=) Date: Fri, 13 Aug 2010 17:04:39 +0200 Subject: [Freeswitch-users] Mobile extensions In-Reply-To: References: <4C655B42.9040504@telemaque.fr> Message-ID: <4C655F07.6070101@telemaque.fr> Nope, vpn connects me inside LAN, and I only have one register ( using sipdroid on android X10mini ). Did not tested on production, 'cause registrar is kamailio, but lab register on fs is successful and no double reg issue in db. btw, hello brian, it's been a while ! sorry I'm too busy at the moment to hang out on fs :/ On 13/08/2010 16:53, Brian West wrote: > Happen to be running on vmware? Also is the registration in the db twice? > > /b > > On Aug 13, 2010, at 9:48 AM, Tristan Mah? wrote: > > >> Hi Ken, >> >> vpn on my phone, always registered to internal profile... :) >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Cordialement, ********************************** * Tristan Mah? * * T?l?maque - Service Voix * * Tel: +33 4.92.90.99.85 * * Mob: +33 6.24.16.43.01 * * Fax: +33 4.92.90.91.46 * ********************************** From t.mahe at telemaque.fr Fri Aug 13 08:08:20 2010 From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=) Date: Fri, 13 Aug 2010 17:08:20 +0200 Subject: [Freeswitch-users] Mobile extensions In-Reply-To: <4C655E28.40408@puzzled.xs4all.nl> References: <4C655B42.9040504@telemaque.fr> <4C655E28.40408@puzzled.xs4all.nl> Message-ID: <4C655FE4.7020607@telemaque.fr> On 13/08/2010 17:00, Patrick Lists wrote: > On 08/13/2010 04:48 PM, Tristan Mah? wrote: > >> Hi Ken, >> >> vpn on my phone, always registered to internal profile... :) >> > All 2 hours before the phone's battery dies :) > Why not just do SIP/TLS? > Because I don't need to let it run, I have followme ringing the gsm for incoming. Just need this for outgoing calls. And vpnc + sipdroid = less than 2 hours if you do some ssh/web browsing troubleshooting with it :D > regards, > Patrick > > >> On 13/08/2010 16:41, Ken Gillett wrote: >> >>> I was wondering about the use of mobile phones running a SIP client. If they are configured as a FreeSwitch extension, they stop working once away from the local LAN. But if they ALSO register to the VOIP provider's (external) SIP server, when they ARE on the local LAN, they will get the calls twice. If they ONLY register to the external SIP server, you lose the power and flexibility of managing them as extensions. >>> >>> I cannot be unique in wanting to use Mobiles as extensions. How do others get around this? >>> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Cordialement, ********************************** * Tristan Mah? * * T?l?maque - Service Voix * * Tel: +33 4.92.90.99.85 * * Mob: +33 6.24.16.43.01 * * Fax: +33 4.92.90.91.46 * ********************************** From 12ukwn at gmail.com Fri Aug 13 08:24:37 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 13 Aug 2010 17:24:37 +0200 Subject: [Freeswitch-users] Mobile extensions In-Reply-To: <4C655E28.40408@puzzled.xs4all.nl> References: <4C655B42.9040504@telemaque.fr> <4C655E28.40408@puzzled.xs4all.nl> Message-ID: <20100813172437.3cfc221e@anubis.defcon1> Le Fri, 13 Aug 2010 17:00:56 +0200, Patrick Lists a ?crit : > On 08/13/2010 04:48 PM, Tristan Mah? wrote: > > Hi Ken, > > > > vpn on my phone, always registered to internal profile... :) > > All 2 hours before the phone's battery dies :) > Why not just do SIP/TLS? Because Tristan is in france, and here providers do all they can to block you from using any other kind of comms than their's. And because the VPN provide the wanted security (acknowledging the fact that the LAN is snoop safe, oeuf corse.) JY -- Anyone who considers protocol unimportant has never dealt with a cat. -- R. Heinlein From jeff at jefflenk.com Fri Aug 13 08:31:38 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 13 Aug 2010 08:31:38 -0700 (PDT) Subject: [Freeswitch-users] mod_nibblebill line #487 In-Reply-To: References: Message-ID: <1281713498997-5420461.post@n2.nabble.com> Your correct - corrected in git head -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-nibblebill-line-487-tp5418802p5420461.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jmesquita at freeswitch.org Fri Aug 13 09:53:15 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 13 Aug 2010 13:53:15 -0300 Subject: [Freeswitch-users] CentOS 5.x and Python 2.6: segmentation fault importing ESL In-Reply-To: References: Message-ID: The flags must inserted on the makefile are the ones when you execute the python-config script using the python2.6 binary. If not, give me access and I will make the mods for you so you can learn. Regards, Jo?o Mesquita On Thu, Aug 12, 2010 at 5:03 AM, Neil Patel wrote: > Hi Jo?o, > > I am still not able to get it to compile for 2.6. > > By 'path' I assume you mean whatever version /usr/bin/python points to? I > tried aliasing that with 2.6, it still didn't work. Note that it's not > recommended to replace 2.6 with 2.4 on CentOS, so I've installed it in > parallel. > > How exactly do I modify python-config? I tried setting the first line to > '#! /usr/bin/python2.6', but that didn't work. Also tried changed the > 'pyver' variable to '2.6', but then I get a compilation error saying that > lib is not found. > > Thanks, > Neil > > 2010/8/11 Jo?o Mesquita > >> Neil, that's because ESL is being compiled against 2.4 and not 2.6. It >> will use whatever is set on your path. >> >> You don't have to change anything on SWIG. You only have to change what is >> being generated by the python-config script located in >> ${SRC}/libs/esl/python >> >> Hope that helps. >> >> Jo?o Mesquita >> >> >> On Tue, Aug 10, 2010 at 4:12 AM, Neil Patel wrote: >> >>> Hi All, >>> >>> I am using Python 2.6 on my CentOS installation, installed in parallel to >>> the standard Python 2.4. I've compiled FS's python ESL module using 'make >>> pymod', but the ESL python module causes a segfault when I try to import it >>> in Python 2.6 (imports fine in 2.4). I'm guessing this is because the ESL >>> module was compiled for Python 2.4. How do I change the ESL config/makefiles >>> to compile for 2.6? I'm not familiar with SWIG, is there some configuration >>> I need to change with it to use python2.6 executable? >>> >>> Thanks, >>> Neil >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/64ae0464/attachment.html From pjintheusa at gmail.com Fri Aug 13 09:57:29 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 13 Aug 2010 12:57:29 -0400 Subject: [Freeswitch-users] Enterprise Originate and group_confirm In-Reply-To: References: Message-ID: Ok I know am talking to myself here but perhaps this useful to someone. As I see it this dialplan should work: The idea here is that it should only call 2125556599 once. This works if you answer the call and let it timeout (don't press 1 or hangup). The next gateway is not tried. But if you answer and then hangup - without pressing 1, then the next gateway is tried and you get the call again. fail_on_single_reject would seem to apply here, but does not have the desired effect. Also setting a leg_timeout as follows: > > > > data="{ignore_early_media=true,origination_caller_id_number=2155556240,group_confirm_file=prompts\press-1-to-accept-call-from.wav,group_confirm_key=1} > > [leg_timeout=25]sofia/gateway/quest/6095553828|[leg_timeout=25]sofia/gateway/broadvox1/6095553828 > > :_:{ignore_early_media=true,origination_caller_id_number=2155556240,group_confirm_file=prompts\press-1-to-accept-call-from.wav,group_confirm_key=1} > > [leg_timeout=25]sofia/gateway/quest/2155554374|[leg_timeout=25]sofia/gateway/broadvox1/2155554374"/> > data="PPNSystem.Telephony.LeaveVoicemailHandler"/> > > > > Every thing works great in terms of mutiple legs ringing, bridging, hanging > up etc. > > I have one issue though: > > When either of the numbers, 6095553828 for example, answers the call but > does not accept it (i.e. does not press 1) and hangs up, the dialplan tries > the next gateway for the same number and therefore 6095553828 ends up > getting another call even though he has rejected it. I was expecting that > once the call is answered and rejected (not bridged) then that part of the > dialplan would stop. > > Any help appreciated. > > Thanks > > Pj > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/8995659e/attachment.html From brian at freeswitch.org Fri Aug 13 10:09:58 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Aug 2010 12:09:58 -0500 Subject: [Freeswitch-users] Enterprise Originate and group_confirm In-Reply-To: References: Message-ID: <23CC36CD-05AF-4612-A21A-CA3013DC2668@freeswitch.org> try export instead of set... but you should set all those inside the {} or export nolocal /b On Aug 13, 2010, at 11:57 AM, Phillip Jones wrote: > Ok I know am talking to myself here but perhaps this useful to someone. > From yehavi.bourvine at gmail.com Fri Aug 13 11:26:54 2010 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 13 Aug 2010 21:26:54 +0300 Subject: [Freeswitch-users] Limits on number of concurrent registered users? In-Reply-To: <4C653AC7.8010009@infosecurity.ch> References: <4C653AC7.8010009@infosecurity.ch> Message-ID: Do you have phones already connected with TLS? I tried doing so for SNOM and Polycom, and both disconnect after a while (and re-connect later). I couldn't decide so far whether it is a FreeSwitch or openSSL issue. Regards, __Yehavi: 2010/8/13 Fabio Pietrosanti (naif) > Hi all, > > i am preparing a setup where i would have a huge number of connected > users via SIP/TLS . > > The amount of concurrent calls will not be such high. > > However he amount of concurrently registered users could be even of 15.000 > . > > Are there some specific internal design/architectural limits of > FreeSWITCH on the number of concurrent users? > > Fabio > > p.s. we'll try to do some stress testing next week > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/194d32ff/attachment-0001.html From fdelawarde at wirelessmundi.com Fri Aug 13 01:27:54 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Fri, 13 Aug 2010 10:27:54 +0200 Subject: [Freeswitch-users] use of multiple templates and output files with mod_cdr_csv In-Reply-To: References: <1281605124.28815.239.camel@luna.tc.commsmundi.com> Message-ID: <1281688074.16764.25.camel@luna.tc.commsmundi.com> Thanks for the answers, > > Look at the accountcode channel variable: > > http://wiki.freeswitch.org/wiki/Mod_cdr_csv#accountcode I initially wanted to use the accountcode variable as an account code for billing and not for template selection, but I guess I can use it for template and create another variable (ex: realaccountcode) for the actual account code. Only thing would be that I'd be limited to two templates. Say I want to log to CSV and a couple of database tables with different fields, well I'll have to do it "manually". > what I would do is to log all the fields you need to Master.csv, and > then do selective imports of the fields you require depending on the > usage. > mysqlimport (LOAD DATA INFILE) allows that if you put a dummy variable > (@dummy) instead of a column name. Nice idea thanks, I could also have my own CDR script listening for hangup events, or to log everything to say an sqlite db and use a cron script to create other logs out of it, but the LOAD DATA INFILE method seems quite cool. Last question: Is there any way to load this module more than one time with different base directories to make it simpler? Thanks again, Fran?ois. From fdelawarde at wirelessmundi.com Fri Aug 13 01:33:54 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Fri, 13 Aug 2010 10:33:54 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> Message-ID: <1281688434.16764.31.camel@luna.tc.commsmundi.com> That's cool, any API function to play a file with output to another file (to convert a native format to wav for example)? It could be a way to record in native format without using licenses and later convert it to another format using free licenses when a call hangs up. Fran?ois. On Thu, 2010-08-12 at 16:24 -0500, Brian West wrote: > No you can... just not in stereo :P > > /b > > On Aug 12, 2010, at 4:19 PM, David Ponzone wrote: > > > AFAIK, record using native file format is not possible. From ktngl at yahoo.co.uk Fri Aug 13 01:46:04 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Fri, 13 Aug 2010 08:46:04 +0000 (GMT) Subject: [Freeswitch-users] PlayAndGetDigits reset input Message-ID: <923456.88070.qm@web29203.mail.ird.yahoo.com> Using PlayAndGetDigits I want to give option to the user to reset their input by entering a certain key sequence (##) How can I make PlayAndGetDigits reprocess immediatly if this key squenqence is pressed. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/a34f4e3c/attachment.html From marshall at gotspeech.net Fri Aug 13 11:33:22 2010 From: marshall at gotspeech.net (Marshall Harrison) Date: Fri, 13 Aug 2010 14:33:22 -0400 Subject: [Freeswitch-users] Audio quesion Message-ID: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> I have FreeSWITCH setup to answer calls on port 5080. If I have 2 simultaneous calls are both going over port 5080 or is that just listening port and the calls get routed to a different port? What port is the audio going over? 5080? A different port for each call? Thanks. Marshall Harrison, MVP Microsoft Communications Server GotSpeech Consulting LLC www.GotSpeech.Net | www.GotUC.Net * marshall at gotspeech.net |* marshall at gotuc.net |( 904.342.6205 Description: Microsoft MVP Program -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/93feea7b/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 8910 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/93feea7b/attachment.gif From sos at sokhapkin.dyndns.org Fri Aug 13 11:45:00 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 13 Aug 2010 14:45:00 -0400 Subject: [Freeswitch-users] Audio quesion In-Reply-To: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> Message-ID: <201008131445.00913.sos@sokhapkin.dyndns.org> You need to learn the basics of SIP. SIP does signaling only, audio parameters are passed in SDP. On Friday 13 August 2010, Marshall Harrison wrote: > I have FreeSWITCH setup to answer calls on port 5080. > > > > If I have 2 simultaneous calls are both going over port 5080 or is that > just listening port and the calls get routed to a different port? > > > > What port is the audio going over? 5080? A different port for each call? > > > > Thanks. > > > > Marshall Harrison, > > MVP Microsoft Communications Server > GotSpeech Consulting LLC > > www.GotSpeech.Net | > www.GotUC.Net > > * marshall at gotspeech.net |* > marshall at gotuc.net |( 904.342.6205 > > > > Description: Microsoft MVP Program > From pjintheusa at gmail.com Fri Aug 13 11:47:34 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 13 Aug 2010 14:47:34 -0400 Subject: [Freeswitch-users] Audio quesion In-Reply-To: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> Message-ID: 5080 is the SIP signaling port (call setup / teardown). Media will be on whatever port is negociated with the endpoint. On Fri, Aug 13, 2010 at 2:33 PM, Marshall Harrison wrote: > I have FreeSWITCH setup to answer calls on port 5080. > > > > If I have 2 simultaneous calls are both going over port 5080 or is that > just listening port and the calls get routed to a different port? > > > > What port is the audio going over? 5080? A different port for each call? > > > > Thanks. > > > > *Marshall Harrison, **MVP Microsoft Communications Server* > *GotSpeech Consulting LLC* > > www.GotSpeech.Net | www.GotUC.Net > > * marshall at gotspeech.net |* marshall at gotuc.net |( 904.342.6205 > > > > [image: Description: Microsoft MVP Program] > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/5e6f8662/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 8910 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/5e6f8662/attachment.gif From william.suffill at gmail.com Fri Aug 13 11:48:01 2010 From: william.suffill at gmail.com (William Suffill) Date: Fri, 13 Aug 2010 14:48:01 -0400 Subject: [Freeswitch-users] Audio quesion In-Reply-To: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> Message-ID: 5080 is just sip signaling. RTP will be negociated between endpoints out of a port range. http://wiki.freeswitch.org/wiki/Firewall covers it in a bit more detail to give you a general overview of what ports are used by FreeSWITCH and for what. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/88f1ee9e/attachment.html From pjintheusa at gmail.com Fri Aug 13 11:58:22 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 13 Aug 2010 14:58:22 -0400 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> Message-ID: Bear in mind that when a carrier says they 'filtered' inband DTMF, they probably mean they have clamped it. There is still DTMF on the line - a few ms, but it is there. Some devices see that clamped DTMF and will use it. For instance, HMP in rfc-2833 mode, will take the rfc-2833 and recreate the DTMF tones - so it always has to be listening for inband even thought it has negociated rfc-2833. In our situation - this resulted in echoed digits - 77227799 - etc just as you are seeing. There only way of diagnosing this IMO, and that is with wireshark (tshark) and listening to the line - that way you know exactly what you are dealing with. Our issue was solved btw, by using inband DTMF exclusively. Not ideal bt any means. On Fri, Aug 13, 2010 at 4:24 AM, Dennis wrote: > thanks for your help! > > just as a short info inbetween, while we are still trying to fix the > problem. > > the cirpack of our carrier always receives dtmf-inputs as inband tones > (because, at least in germany, all dtmf-inputs are sent over the > landline as inband tones). > because we always had problems with the carriers cirpack and fs in > conjunction with inband tones (sometimes they where sent two or three > times), our carrier added rfc-2833 to their cirpack. now the carrier > sends both: rfc-2833 AND inband tones. the problem is, that the > carrier tells us, that the cirpack can not filter the inband tones, so > that they just can send rfc-2833. but they tell us, that they add some > stops (or something like this) to the inband tones, so that the inband > tones are not functional anymore. but this does not seem to work > correctly. > > if we listen to dtmf-inputs only on fs, we do not have any problems, > because fs ignores the inband tones and only reacts to the rfc-2833 > signal. therefor we never had any problems. > now, we have a project, where not fs listens to the dtmf-inputs, but a > callcenter on the other side (outbound). therefor the cirpack has to > send clean dtmf-inputs to the callcenter. but we believe, that the > cirpack on the outgoing side is receiving his own inband tones AND > rfc-2833 signals from the incoming side of the cirpack. so the cirpack > on the outgoing side send his own inband tones PLUS the converted > rfc-2833 signals to the other side (callcenter). > we have to go on testing. > > we spoke with someone from nokia about their iwsd and they told us, > that they filter all inband tones, so that we only receive rfc-2833. > we will do some testing with them. > but i wonder, why the nokia iwsd can filter inband tones and the cirpack > can't. > > any ideas? > > will keep you informed, what we found out. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/b7f55aa2/attachment-0001.html From mnhassan at usa.net Fri Aug 13 12:01:44 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sat, 14 Aug 2010 01:01:44 +0600 Subject: [Freeswitch-users] g729 In-Reply-To: <1281688434.16764.31.camel@luna.tc.commsmundi.com> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> Message-ID: I don't think you can convert the encoded format into WAV without a license even when offline. But, what that would help is, it would avoid double license requirements for each recorded call. Regards HASSAN On 2010-08-13, Fran?ois Delawarde wrote: > That's cool, any API function to play a file with output to another file > (to convert a native format to wav for example)? > > It could be a way to record in native format without using licenses and > later convert it to another format using free licenses when a call hangs > up. > > Fran?ois. > > On Thu, 2010-08-12 at 16:24 -0500, Brian West wrote: >> No you can... just not in stereo :P >> >> /b >> >> On Aug 12, 2010, at 4:19 PM, David Ponzone wrote: >> >> > AFAIK, record using native file format is not possible. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From marshall at gotspeech.net Fri Aug 13 12:12:46 2010 From: marshall at gotspeech.net (Marshall Harrison) Date: Fri, 13 Aug 2010 15:12:46 -0400 Subject: [Freeswitch-users] Audio quesion In-Reply-To: <201008131445.00913.sos@sokhapkin.dyndns.org> References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> <201008131445.00913.sos@sokhapkin.dyndns.org> Message-ID: <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> I know that. I'm trying to determine how a client's serve which he claims only has port 5080 open is working. If what he is saying is true then I don't see how the calls are working. I can't get into the system so I am having to debug based on what he is telling me. FreeSWITCH is front ending a Microsoft Speech Server and is a trusted SIP Peer. When a call comes in SesWorker opens a port for RTP with the SIP Peer that can be anything from 1024 - 65535. That RTP channel is between FreeSWITCH and Speech Server. So does FS then open that same port with the originator of the call or does it negotiate a different port? Marshall Harrison marshall at gotspeech.net | ? 904.342.6205 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Okhapkin Sent: Friday, August 13, 2010 2:45 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quesion You need to learn the basics of SIP. SIP does signaling only, audio parameters are passed in SDP. On Friday 13 August 2010, Marshall Harrison wrote: > I have FreeSWITCH setup to answer calls on port 5080. > > > > If I have 2 simultaneous calls are both going over port 5080 or is > that just listening port and the calls get routed to a different port? > > > > What port is the audio going over? 5080? A different port for each call? > > > > Thanks. > > > > Marshall Harrison, > son> > MVP Microsoft Communications Server > GotSpeech Consulting LLC > > www.GotSpeech.Net | > www.GotUC.Net > > * marshall at gotspeech.net |* > marshall at gotuc.net |( 904.342.6205 > > > > Description: Microsoft MVP Program > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pjintheusa at gmail.com Fri Aug 13 12:24:54 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 13 Aug 2010 15:24:54 -0400 Subject: [Freeswitch-users] Enterprise Originate and group_confirm In-Reply-To: <23CC36CD-05AF-4612-A21A-CA3013DC2668@freeswitch.org> References: <23CC36CD-05AF-4612-A21A-CA3013DC2668@freeswitch.org> Message-ID: Ok - thanks Brian I have some further testing and it does not look like "fail_on_single_reject" applys to the | (OR) , just to the , (AND) So with , - you will not get the call because bad_gateway fails But with | You do get the call - fail_on_single_reject is ignored. I tried all combination with export, {} etc Make sense? On Fri, Aug 13, 2010 at 1:09 PM, Brian West wrote: > try export instead of set... but you should set all those inside the {} or > export nolocal > > /b > > On Aug 13, 2010, at 11:57 AM, Phillip Jones wrote: > > > Ok I know am talking to myself here but perhaps this useful to someone. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/bd383b1c/attachment.html From sos at sokhapkin.dyndns.org Fri Aug 13 12:26:03 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 13 Aug 2010 15:26:03 -0400 Subject: [Freeswitch-users] Audio quesion In-Reply-To: <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> <201008131445.00913.sos@sokhapkin.dyndns.org> <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> Message-ID: <201008131526.03405.sos@sokhapkin.dyndns.org> It depends on FS configuration. If bypass_media is enabled, FS will set direct audio path between MSS and SIP client, if not (default case), FS will open 2 RTP ports to proxy/transcode audio packets. On Friday 13 August 2010, Marshall Harrison wrote: > I know that. > > I'm trying to determine how a client's serve which he claims only has port > 5080 open is working. If what he is saying is true then I don't see how the > calls are working. I can't get into the system so I am having to debug > based on what he is telling me. > > FreeSWITCH is front ending a Microsoft Speech Server and is a trusted SIP > Peer. When a call comes in SesWorker opens a port for RTP with the SIP > Peer that can be anything from 1024 - 65535. That RTP channel is between > FreeSWITCH and Speech Server. > > So does FS then open that same port with the originator of the call or does > it negotiate a different port? > > Marshall Harrison > marshall at gotspeech.net | 904.342.6205 > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey > Okhapkin > Sent: Friday, August 13, 2010 2:45 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Audio quesion > > You need to learn the basics of SIP. SIP does signaling only, audio > parameters are passed in SDP. > > On Friday 13 August 2010, Marshall Harrison wrote: > > I have FreeSWITCH setup to answer calls on port 5080. > > > > > > > > If I have 2 simultaneous calls are both going over port 5080 or is > > that just listening port and the calls get routed to a different port? > > > > > > > > What port is the audio going over? 5080? A different port for each call? > > > > > > > > Thanks. > > > > > > > > Marshall Harrison, > > > son> > > MVP Microsoft Communications Server > > GotSpeech Consulting LLC > > > > www.GotSpeech.Net | > > www.GotUC.Net > > > > * marshall at gotspeech.net |* > > marshall at gotuc.net |( 904.342.6205 > > > > > > > > Description: Microsoft MVP Program > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveayre at gmail.com Fri Aug 13 12:48:59 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 13 Aug 2010 20:48:59 +0100 Subject: [Freeswitch-users] Audio quesion In-Reply-To: <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> <201008131445.00913.sos@sokhapkin.dyndns.org> <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> Message-ID: What is FreeSWITCH running on? Linux? Iptables has a module which recognises RTP packets attached to a SIP session and will mark them as RELATED. If his firewall is state based then it may be automatically opening the RTP ports for the call even if they are normally blocked. There will be two separate connections for a call: 1. Signalling 2. Media Signalling will be between port 5080 and a random port on the client. It can be UDP or TCP (normally UDP). Media is a separate protocol (RDP), it will always be a separate connection never to port 5080. There's just no way for it to piggyback on the SIP. SIP contains SDP, this contains the information on where to send media to. On a call to the server, the INVITE will contain SDP from the client which will contain the IP and port on the client which will receive media (one end of the connection). When the server starts sending media it will send a SIP 180/183/200 with SDP. This will contain the IP and port at the server end for the client to send media to. If you collect a packet trace of the call, you will be able to see the IPs and ports that media will be using. Or by capturing the RTP packets to see where they're going from/to. Media can either go via the server (which would require the port to be open, or a state based firewall) or bypass it (bypass media mode, where the client IPs are in the SDP not the server so the connection is opened directly between the clients, therefore no open port is required on the server). -Steve On 13 August 2010 20:12, Marshall Harrison wrote: > I know that. > > I'm trying to determine how a client's serve which he claims only has port > 5080 open is working. If what he is saying is true then I don't see how the > calls are working. I can't get into the system so I am having to debug > based > on what he is telling me. > > FreeSWITCH is front ending a Microsoft Speech Server and is a trusted SIP > Peer. When a call comes in SesWorker opens a port for RTP with the SIP > Peer > that can be anything from 1024 - 65535. That RTP channel is between > FreeSWITCH and Speech Server. > > So does FS then open that same port with the originator of the call or does > it negotiate a different port? > > Marshall Harrison > marshall at gotspeech.net | 904.342.6205 > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey > Okhapkin > Sent: Friday, August 13, 2010 2:45 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Audio quesion > > You need to learn the basics of SIP. SIP does signaling only, audio > parameters are passed in SDP. > > On Friday 13 August 2010, Marshall Harrison wrote: > > I have FreeSWITCH setup to answer calls on port 5080. > > > > > > > > If I have 2 simultaneous calls are both going over port 5080 or is > > that just listening port and the calls get routed to a different port? > > > > > > > > What port is the audio going over? 5080? A different port for each call? > > > > > > > > Thanks. > > > > > > > > Marshall Harrison, > > > son> > > MVP Microsoft Communications Server > > GotSpeech Consulting LLC > > > > www.GotSpeech.Net | > > www.GotUC.Net > > > > * marshall at gotspeech.net |* > > marshall at gotuc.net |( 904.342.6205 > > > > > > > > Description: Microsoft MVP Program > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/d4a4a747/attachment-0001.html From steveayre at gmail.com Fri Aug 13 12:50:16 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 13 Aug 2010 20:50:16 +0100 Subject: [Freeswitch-users] Audio quesion In-Reply-To: <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> <201008131445.00913.sos@sokhapkin.dyndns.org> <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> Message-ID: Just to check, how does he determine only 5080 is open? Does he mean it is the only port open in the firewall or does he just mean it's the only listening port? -Steve On 13 August 2010 20:12, Marshall Harrison wrote: > I know that. > > I'm trying to determine how a client's serve which he claims only has port > 5080 open is working. If what he is saying is true then I don't see how the > calls are working. I can't get into the system so I am having to debug > based > on what he is telling me. > > FreeSWITCH is front ending a Microsoft Speech Server and is a trusted SIP > Peer. When a call comes in SesWorker opens a port for RTP with the SIP > Peer > that can be anything from 1024 - 65535. That RTP channel is between > FreeSWITCH and Speech Server. > > So does FS then open that same port with the originator of the call or does > it negotiate a different port? > > Marshall Harrison > marshall at gotspeech.net | 904.342.6205 > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey > Okhapkin > Sent: Friday, August 13, 2010 2:45 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Audio quesion > > You need to learn the basics of SIP. SIP does signaling only, audio > parameters are passed in SDP. > > On Friday 13 August 2010, Marshall Harrison wrote: > > I have FreeSWITCH setup to answer calls on port 5080. > > > > > > > > If I have 2 simultaneous calls are both going over port 5080 or is > > that just listening port and the calls get routed to a different port? > > > > > > > > What port is the audio going over? 5080? A different port for each call? > > > > > > > > Thanks. > > > > > > > > Marshall Harrison, > > > son> > > MVP Microsoft Communications Server > > GotSpeech Consulting LLC > > > > www.GotSpeech.Net | > > www.GotUC.Net > > > > * marshall at gotspeech.net |* > > marshall at gotuc.net |( 904.342.6205 > > > > > > > > Description: Microsoft MVP Program > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/f7f870bb/attachment.html From msc at freeswitch.org Fri Aug 13 13:25:15 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Aug 2010 13:25:15 -0700 Subject: [Freeswitch-users] PlayAndGetDigits reset input In-Reply-To: <923456.88070.qm@web29203.mail.ird.yahoo.com> References: <923456.88070.qm@web29203.mail.ird.yahoo.com> Message-ID: On Fri, Aug 13, 2010 at 1:46 AM, Nigel Kent wrote: > Using PlayAndGetDigits I want to give option to the user to reset their > input by entering a certain key sequence (##) > > How can I make PlayAndGetDigits reprocess immediatly if this key squenqence > is pressed. > I don't believe playandgetdigits does this. You will have to listen for ## (or whatever) and then loop back and re-execute playandgetdigits. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/a51047da/attachment.html From marshall at gotspeech.net Fri Aug 13 12:14:25 2010 From: marshall at gotspeech.net (Marshall Harrison) Date: Fri, 13 Aug 2010 15:14:25 -0400 Subject: [Freeswitch-users] Audio quesion In-Reply-To: References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> Message-ID: <030401cb3b1b$be805bb0$3b811310$@gotspeech.net> OK. FreeSWITCH is front ending a Microsoft Speech Server and is a trusted SIP Peer. When a call comes in SesWorker (MSS) opens a port for RTP with the SIP Peer 9FS) that can be anything from 1024 - 65535. That RTP channel is between FreeSWITCH and Speech Server. So does FS then open that same port with the originator of the call or does it negotiate a different port? Marshall Harrison * marshall at gotspeech.net | ( 904.342.6205 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, August 13, 2010 2:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quesion 5080 is the SIP signaling port (call setup / teardown). Media will be on whatever port is negociated with the endpoint. On Fri, Aug 13, 2010 at 2:33 PM, Marshall Harrison wrote: I have FreeSWITCH setup to answer calls on port 5080. If I have 2 simultaneous calls are both going over port 5080 or is that just listening port and the calls get routed to a different port? What port is the audio going over? 5080? A different port for each call? Thanks. Marshall Harrison, MVP Microsoft Communications Server GotSpeech Consulting LLC www.GotSpeech.Net | www.GotUC.Net * marshall at gotspeech.net |* marshall at gotuc.net |( 904.342.6205 Description: Microsoft MVP Program _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/34d8981d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 8910 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/34d8981d/attachment-0001.gif From marshall at gotspeech.net Fri Aug 13 13:05:58 2010 From: marshall at gotspeech.net (Marshall Harrison) Date: Fri, 13 Aug 2010 16:05:58 -0400 Subject: [Freeswitch-users] Audio quesion In-Reply-To: References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> <201008131445.00913.sos@sokhapkin.dyndns.org> <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> Message-ID: <032201cb3b22$f27d83f0$d7788bd0$@gotspeech.net> According to him 5080 is the only port open on the firewall for the IP of the server.. Things have progressed some. FS and MSS are running on the same box. Inbound calls, outbound calls, blind transfers and supervised (whisper) transfers were working outside the firewall. Now he has moved it inside the firewall and calls work but no audio. I've had him open ports 1024 - 65535 which are what MSS requires but still no audio. I'll have him check bypass_media setting. Could be that their NAT is screwing things up. Marshall Harrison * marshall at gotspeech.net | ( 904.342.6205 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Friday, August 13, 2010 3:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quesion Just to check, how does he determine only 5080 is open? Does he mean it is the only port open in the firewall or does he just mean it's the only listening port? -Steve On 13 August 2010 20:12, Marshall Harrison wrote: I know that. I'm trying to determine how a client's serve which he claims only has port 5080 open is working. If what he is saying is true then I don't see how the calls are working. I can't get into the system so I am having to debug based on what he is telling me. FreeSWITCH is front ending a Microsoft Speech Server and is a trusted SIP Peer. When a call comes in SesWorker opens a port for RTP with the SIP Peer that can be anything from 1024 - 65535. That RTP channel is between FreeSWITCH and Speech Server. So does FS then open that same port with the originator of the call or does it negotiate a different port? Marshall Harrison marshall at gotspeech.net | 904.342.6205 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Okhapkin Sent: Friday, August 13, 2010 2:45 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quesion You need to learn the basics of SIP. SIP does signaling only, audio parameters are passed in SDP. On Friday 13 August 2010, Marshall Harrison wrote: > I have FreeSWITCH setup to answer calls on port 5080. > > > > If I have 2 simultaneous calls are both going over port 5080 or is > that just listening port and the calls get routed to a different port? > > > > What port is the audio going over? 5080? A different port for each call? > > > > Thanks. > > > > Marshall Harrison, > son> > MVP Microsoft Communications Server > GotSpeech Consulting LLC > > www.GotSpeech.Net | > www.GotUC.Net > > * marshall at gotspeech.net |* > marshall at gotuc.net |( 904.342.6205 > > > > Description: Microsoft MVP Program > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/8e40df77/attachment-0001.html From marshall at gotspeech.net Fri Aug 13 13:17:07 2010 From: marshall at gotspeech.net (Marshall Harrison) Date: Fri, 13 Aug 2010 16:17:07 -0400 Subject: [Freeswitch-users] Audio quesion In-Reply-To: References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> <201008131445.00913.sos@sokhapkin.dyndns.org> <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> Message-ID: <032701cb3b24$80e31be0$82a953a0$@gotspeech.net> Running on Windows. Do you know what the port range is that FS uses for RTP? The only trace I have has the SDP showing audio on port 16700 which falls within the 1024 - 65535 that I had them open Marshall Harrison * marshall at gotspeech.net | ( 904.342.6205 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Friday, August 13, 2010 3:49 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quesion What is FreeSWITCH running on? Linux? Iptables has a module which recognises RTP packets attached to a SIP session and will mark them as RELATED. If his firewall is state based then it may be automatically opening the RTP ports for the call even if they are normally blocked. There will be two separate connections for a call: 1. Signalling 2. Media Signalling will be between port 5080 and a random port on the client. It can be UDP or TCP (normally UDP). Media is a separate protocol (RDP), it will always be a separate connection never to port 5080. There's just no way for it to piggyback on the SIP. SIP contains SDP, this contains the information on where to send media to. On a call to the server, the INVITE will contain SDP from the client which will contain the IP and port on the client which will receive media (one end of the connection). When the server starts sending media it will send a SIP 180/183/200 with SDP. This will contain the IP and port at the server end for the client to send media to. If you collect a packet trace of the call, you will be able to see the IPs and ports that media will be using. Or by capturing the RTP packets to see where they're going from/to. Media can either go via the server (which would require the port to be open, or a state based firewall) or bypass it (bypass media mode, where the client IPs are in the SDP not the server so the connection is opened directly between the clients, therefore no open port is required on the server). -Steve On 13 August 2010 20:12, Marshall Harrison wrote: I know that. I'm trying to determine how a client's serve which he claims only has port 5080 open is working. If what he is saying is true then I don't see how the calls are working. I can't get into the system so I am having to debug based on what he is telling me. FreeSWITCH is front ending a Microsoft Speech Server and is a trusted SIP Peer. When a call comes in SesWorker opens a port for RTP with the SIP Peer that can be anything from 1024 - 65535. That RTP channel is between FreeSWITCH and Speech Server. So does FS then open that same port with the originator of the call or does it negotiate a different port? Marshall Harrison marshall at gotspeech.net | 904.342.6205 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Okhapkin Sent: Friday, August 13, 2010 2:45 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quesion You need to learn the basics of SIP. SIP does signaling only, audio parameters are passed in SDP. On Friday 13 August 2010, Marshall Harrison wrote: > I have FreeSWITCH setup to answer calls on port 5080. > > > > If I have 2 simultaneous calls are both going over port 5080 or is > that just listening port and the calls get routed to a different port? > > > > What port is the audio going over? 5080? A different port for each call? > > > > Thanks. > > > > Marshall Harrison, > son> > MVP Microsoft Communications Server > GotSpeech Consulting LLC > > www.GotSpeech.Net | > www.GotUC.Net > > * marshall at gotspeech.net |* > marshall at gotuc.net |( 904.342.6205 > > > > Description: Microsoft MVP Program > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/9c7e7d93/attachment-0001.html From brian at freeswitch.org Fri Aug 13 14:09:23 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Aug 2010 16:09:23 -0500 Subject: [Freeswitch-users] Audio quesion In-Reply-To: <032201cb3b22$f27d83f0$d7788bd0$@gotspeech.net> References: <02c701cb3b16$02793bd0$076bb370$@gotspeech.net> <201008131445.00913.sos@sokhapkin.dyndns.org> <030301cb3b1b$83c7b450$8b571cf0$@gotspeech.net> <032201cb3b22$f27d83f0$d7788bd0$@gotspeech.net> Message-ID: <917D5B4C-65A7-4802-9B14-689CE93BE62B@freeswitch.org> If you're on windows the on board nat stuff will poke holes for you with uPNP ... /b On Aug 13, 2010, at 3:05 PM, Marshall Harrison wrote: > According to him 5080 is the only port open on the firewall for the IP of the server.. > > Things have progressed some. > > FS and MSS are running on the same box. Inbound calls, outbound calls, blind transfers and supervised (whisper) transfers were working outside the firewall. > > Now he has moved it inside the firewall and calls work but no audio. I?ve had him open ports 1024 ? 65535 which are what MSS requires but still no audio. > > I?ll have him check bypass_media setting. > > Could be that their NAT is screwing things up. > From gmaruzz at celliax.org Fri Aug 13 14:31:50 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 13 Aug 2010 23:31:50 +0200 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Jason, please test with the latest git version. Many messy changes to an already messy code, so maybe some bug or side effect (ok, bug) has been introduced. commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 Author: Giovanni Maruzzelli Date: Fri Aug 13 16:19:20 2010 -0500 skypopen: now answer a call only when directed to do it (before was trying to answer any incoming call). Lot of changes to a messy part, so maybe some problem will come out... Signed-off-by: Giovanni Maruzzelli -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Fri Aug 13 14:35:37 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 13 Aug 2010 23:35:37 +0200 Subject: [Freeswitch-users] mod_skypopen (skype endpoint) changes, please test Message-ID: Hi FreeSWITCHers, I've made some long due modifications to mod_skypopen, that maybe introduced bugs. Please test with the latest git and report any *new* problem (ok, old problems too), here in the mailing list, or - much better - in the Jira. ========= commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 Author: Giovanni Maruzzelli Date: Fri Aug 13 16:19:20 2010 -0500 skypopen: now answer a call only when directed to do it (before was trying to answer any incoming call). Lot of changes to a messy part, so maybe some problem will come out... Signed-off-by: Giovanni Maruzzelli =========== Thank you to all, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From egable+freeswitch at gmail.com Fri Aug 13 17:25:48 2010 From: egable+freeswitch at gmail.com (Eliot Gable) Date: Fri, 13 Aug 2010 20:25:48 -0400 Subject: [Freeswitch-users] Limits on number of concurrent registered users? In-Reply-To: References: <4C653AC7.8010009@infosecurity.ch> Message-ID: There are no built-in limits on the number of registered users. The only thing that matters in terms of limits is how fast of a server you deploy FreeSWITCH on. Obviously, handling a register request takes some processing power. Handling requests from 15,000 users will take a decent amount of processing power. The amount it takes depends on how often a user re-registers. If they are doing it once per hour then you will need to be able to process just over 4 per second, which is not substantial at all. If they register every 15 seconds, then you are looking at 1,000 per second, which is something else entirely. As for TLS/SSL, I have used it for some time and never had an issue with the phone disconnecting. I tested using a Polycom phone. On Fri, Aug 13, 2010 at 2:26 PM, Yehavi Bourvine wrote: > Do you have phones already connected with TLS? I tried doing so for SNOM and > Polycom, and both disconnect after a while (and re-connect later). I > couldn't decide so far whether it is a FreeSwitch or openSSL issue. > > ????????????????????? Regards, __Yehavi: > > 2010/8/13 Fabio Pietrosanti (naif) >> >> Hi all, >> >> i am preparing a setup where i would have a huge number of connected >> users via SIP/TLS . >> >> The amount of concurrent calls will not be such high. >> >> However he amount of concurrently registered users could be even of 15.000 >> . >> >> Are there some specific internal design/architectural limits of >> FreeSWITCH on the number of concurrent users? >> >> Fabio >> >> p.s. we'll try to do some stress testing next week >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero From Victor at isptelecom.net Fri Aug 13 19:46:50 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Fri, 13 Aug 2010 22:46:50 -0400 Subject: [Freeswitch-users] FIFO queue callers ID: how to set to a real-life callers ID, not a static entry? In-Reply-To: References: <4C62FB3D.4030707@isptelecom.net> Message-ID: <4C66039A.7070106@isptelecom.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/aaa2ef61/attachment.html From babak.freeswitch at gmail.com Fri Aug 13 22:22:16 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Sat, 14 Aug 2010 09:52:16 +0430 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> Message-ID: thanx for ur answers may be I'm wrong but it seems if all my endpoints support g729 and I record to native format I would just need 10 licenses -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/fca34bfe/attachment.html From steveu at coppice.org Fri Aug 13 22:48:16 2010 From: steveu at coppice.org (Steve Underwood) Date: Sat, 14 Aug 2010 13:48:16 +0800 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> Message-ID: <4C662E20.8090209@coppice.org> On 08/14/2010 01:22 PM, babak yakhchali wrote: > thanx for ur answers > may be I'm wrong but it seems if all my endpoints support g729 and I > record to native format I would just need 10 licenses > FS will be playing piggy in the middle, monitoring 2 audio streams per call. If you want to decode both, that is 2 licences. This may seem unfair, as you are using 2 decoders and no encoders - the same amount of licenced resources as one normal call. Why can't that be arranged to be a single licence? The answer lies in the terms of the patent licencing conditions. Steve From steveayre at gmail.com Sat Aug 14 02:06:53 2010 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 14 Aug 2010 10:06:53 +0100 Subject: [Freeswitch-users] g729 In-Reply-To: <4C662E20.8090209@coppice.org> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> <4C662E20.8090209@coppice.org> Message-ID: This even goes for recording to native format, as FS must decode the G729 audio in each direction, mix them together into a single stream and then encode them back to G729. 2 decoders, 1 encoder = 2 licences. Recording to native format only doesn't require a license if it's a single stream so that mixing isn't required. -Steve On 14 August 2010 06:48, Steve Underwood wrote: > On 08/14/2010 01:22 PM, babak yakhchali wrote: > > thanx for ur answers > > may be I'm wrong but it seems if all my endpoints support g729 and I > > record to native format I would just need 10 licenses > > > FS will be playing piggy in the middle, monitoring 2 audio streams per > call. If you want to decode both, that is 2 licences. This may seem > unfair, as you are using 2 decoders and no encoders - the same amount of > licenced resources as one normal call. Why can't that be arranged to be > a single licence? The answer lies in the terms of the patent licencing > conditions. > > Steve > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/2c7ffabc/attachment-0001.html From mnhassan at usa.net Sat Aug 14 03:22:40 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sat, 14 Aug 2010 16:22:40 +0600 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> <4C662E20.8090209@coppice.org> Message-ID: That was my point exactly. If FS can write to file in native G729 during the call, even if it is different streams, then you can have another program at the end of the call that can do the decode into SLIN, mux them together. That way, only one license can be put aside for that "after call program". Regards HASSAN On Sat, Aug 14, 2010 at 15:06, Steven Ayre wrote: > This even goes for recording to native format, as FS must decode the G729 > audio in each direction, mix them together into a single stream and then > encode them back to G729. 2 decoders, 1 encoder = 2 licences. > > Recording to native format only doesn't require a license if it's a single > stream so that mixing isn't required. > > -Steve > > > > > On 14 August 2010 06:48, Steve Underwood wrote: > >> On 08/14/2010 01:22 PM, babak yakhchali wrote: >> > thanx for ur answers >> > may be I'm wrong but it seems if all my endpoints support g729 and I >> > record to native format I would just need 10 licenses >> > >> FS will be playing piggy in the middle, monitoring 2 audio streams per >> call. If you want to decode both, that is 2 licences. This may seem >> unfair, as you are using 2 decoders and no encoders - the same amount of >> licenced resources as one normal call. Why can't that be arranged to be >> a single licence? The answer lies in the terms of the patent licencing >> conditions. >> >> Steve >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/3ccfff52/attachment.html From steveu at coppice.org Sat Aug 14 03:34:02 2010 From: steveu at coppice.org (Steve Underwood) Date: Sat, 14 Aug 2010 18:34:02 +0800 Subject: [Freeswitch-users] g729 In-Reply-To: References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> <4C662E20.8090209@coppice.org> Message-ID: <4C66711A.3040006@coppice.org> On 08/14/2010 05:06 PM, Steven Ayre wrote: > This even goes for recording to native format, as FS must decode the > G729 audio in each direction, mix them together into a single stream > and then encode them back to G729. 2 decoders, 1 encoder = 2 licences. > > Recording to native format only doesn't require a license if it's a > single stream so that mixing isn't required. Decoding and recoding G.729 leads to an awful quality loss. It best avoided. If you really want to mix the streams, leave the output at a bit higher rate in the subsequent recording if you want the conversation to be heard in detail later on. Steve From dujinfang at gmail.com Sat Aug 14 05:07:09 2010 From: dujinfang at gmail.com (Seven Du) Date: Sat, 14 Aug 2010 20:07:09 +0800 Subject: [Freeswitch-users] speak in play_and_get_digits Message-ID: Hi, in uuid_broadcast, all the following forms work: xxxx.wav a wav file phrase:demo_ivr_main_menu phrase macro speak:hello in play_and_get_digits, speak doesn't work. I know that I could put the actual text into a macro, however, a speak version would be much more clear, especially for demo. EXECUTE portaudio/0000000000 play_and_get_digits(1 9 3 5000 # speak:'flite|kal|please input some digits, end with #' none x_cc_dtmf .*) 2010-08-14 20:00:05.680230 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/speak:flite|kal|please input some digits, end with #.L16 Is there any other way to get it work? Or should I report a bug or request a new feature? Thanks. -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From ktngl at yahoo.co.uk Sat Aug 14 04:14:38 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Sat, 14 Aug 2010 11:14:38 +0000 (GMT) Subject: [Freeswitch-users] PlayAndGetDigits reset input In-Reply-To: Message-ID: <546981.582.qm@web29216.mail.ird.yahoo.com> I think what you suggest would be better - to listen for the digits. Do you have any resource or tips that I can check to see how to implement listening and catching a desired sequence --- On Fri, 13/8/10, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] PlayAndGetDigits reset input To: "FreeSWITCH Users Help" Date: Friday, 13 August, 2010, 20:25 On Fri, Aug 13, 2010 at 1:46 AM, Nigel Kent wrote: Using PlayAndGetDigits I want to give option to the user to reset their input by entering a certain key sequence (##) How can I make PlayAndGetDigits reprocess immediatly if this key squenqence is pressed. I don't believe playandgetdigits does this. You will have to listen for ## (or whatever) and then loop back and re-execute playandgetdigits. -MC -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/c2c4eac8/attachment.html From chhabra.pankaj at gmail.com Fri Aug 13 23:28:13 2010 From: chhabra.pankaj at gmail.com (Pankaj Chhabra) Date: Sat, 14 Aug 2010 11:58:13 +0530 Subject: [Freeswitch-users] Has anyone used Sangama USB FXO card with FreeSWITCH? Message-ID: Hi, Has anyone used Sangoma USB FXO with FreeSWITCH?. If yes, could you please let me know if it works properly? I plan to use this to run FreeSWITCH on my laptop and connect to PSTN using this card. Here is the link to Sangoma USB FXO - http://www.sangoma.com/products/hardware_products/analog_telephony/usb_fxo.html Thanks, Pankaj From jason at cloudtree.net Fri Aug 13 15:44:30 2010 From: jason at cloudtree.net (Jason Jeffords) Date: Fri, 13 Aug 2010 18:44:30 -0400 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Thank you Giovanni, We will test ASAP. Jason On Fri, Aug 13, 2010 at 5:31 PM, Giovanni Maruzzelli wrote: > Jason, > > please test with the latest git version. > > Many messy changes to an already messy code, so maybe some bug or side > effect (ok, bug) has been introduced. > > commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 > Author: Giovanni Maruzzelli > Date: Fri Aug 13 16:19:20 2010 -0500 > > skypopen: now answer a call only when directed to do it (before > was trying to answer any incoming call). Lot of changes to a messy > part, so maybe some problem will come out... > > Signed-off-by: Giovanni Maruzzelli > > -giovanni > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/d0520f0a/attachment.html From m.reinacher at gmail.com Fri Aug 13 14:25:47 2010 From: m.reinacher at gmail.com (Matthias Reinacher) Date: Fri, 13 Aug 2010 23:25:47 +0200 Subject: [Freeswitch-users] TLS problem using SNOM phones Message-ID: Hello all, i have a problem using different Snom phones (300/320/820/821) with Freeswitch and TLS in a internet-wide setup (phones being registered via internet at a FS w/ public IP). Sometimes (25-50% of cases) the first try to call someone (internal) won't go through. Logs from phone, Freeswitch and ssldump show that the phone sends an INVITE, the FS asks for more credentials (digest auth b/c phone IP not in ACL), the phone answers with an INVITE w/ more auth credentials -- and this second INVITE package is not received by the FS. It does arrive at the server though, as verified by ssldump. Interestingly, if one presses "Cancel" on the phone, a new TCP/SSL connection is created, apparently the old one died -- see logs from phone, FS, and ssldump below. Also, Snom 820/821 show the registrar as not registered after such an action (presumably b/c original SSL connection is dead). Has anybody encountered this behaviour and can verify it? Are there any ideas what causes the problem here (FS, OpenSSL, phone?) and if there is any remedy for it? I was planning on using FS as a production phone system w/ encrypted signalling and audio using Snom phones. I can't do that right now b/c of the abovementioned problem, and i have not yet found a solution. Any help is greatly appreciated. TIA Matthias Logs: phone, FS, ssldump, please also see annotation enclosed in "--- ... ---" - phone: Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:225 (1280 bytes): INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:363 (350 bytes): SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:524 (844 bytes): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 From: "Matthias" ;tag=rllbt2d9pz To: ;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="XXXX.dyndns.org", nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" Content-Length: 0 Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:528 (418 bytes): ACK sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" ;tag=rllbt2d9pz To: ;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 --- the following packet is not received by FS --- Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:544 (1541 bytes): INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="1001",realm="XXXX.dyndns.org",nonce="a3346430-8ce9-11df-9347-29b67737619e",uri="sip:9999 at XXXX.dyndns.org;user=phone",qop=auth,nc=00000001,cnonce="4fd48758",response="bb47aafad2fc3980d7540b0d2cbf1b03",algorithm=MD5 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- before this packet is sent, a new TCP/SSL connection is negotiated --- Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:29:936 (386 bytes): CANCEL sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Content-Length: 0 Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:30:562 (345 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=78.53.13.206 From: "Matthias" ;tag=rllbt2d9pz To: ;tag=XUQm659jUQQvB Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Content-Length: 0 - Freeswitch: recv 1280 bytes from tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.774063: ------------------------------------------------------------------------ INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 350 bytes to tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.777861: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=XXX.XXX.XXX.XXX From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-07-11 14:41:33.778081 [DEBUG] sofia.c:5999 IP XXX.XXX.XXX.XXX Rejected by acl "domains". Falling back to Digest auth. send 844 bytes to tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.864787: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=XXX.XXX.XXX.XXX From: "Matthias" ;tag=rllbt2d9pz To: ;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="XXXX.dyndns.org", nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 418 bytes from tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:34.082329: ------------------------------------------------------------------------ ACK sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" ;tag=rllbt2d9pz To: ;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 --- here, the 2nd INVITE packet w/ credentials is missing, note time difference to next packet --- ------------------------------------------------------------------------ recv 386 bytes from tls/[XXX.XXX.XXX.XXX]:4655 at 12:41:41.905857: ------------------------------------------------------------------------ CANCEL sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Content-Length: 0 ------------------------------------------------------------------------ send 345 bytes to tls/[XXX.XXX.XXX.XXX]:4655 at 12:41:41.906402: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=XXX.XXX.XXX.XXX From: "Matthias" ;tag=rllbt2d9pz To: ;tag=XUQm659jUQQvB Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Content-Length: 0 - ssldump: 1 254 2335.8000 (417.3058) C>SV3.1(1300) application_data --------------------------------------------------------------- INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --------------------------------------------------------------- 1 255 2335.8043 (0.0043) S>CV3.1(138) application_data --------------------------------------------------------------- SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 --------------------------------------------------------------- 1 256 2335.8044 (0.0000) S>CV3.1(183) application_data --------------------------------------------------------------- From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE --------------------------------------------------------------- 1 257 2335.8045 (0.0000) S>CV3.1(89) application_data --------------------------------------------------------------- User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 --------------------------------------------------------------- 1 258 2335.8935 (0.0890) S>CV3.1(161) application_data --------------------------------------------------------------- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 --------------------------------------------------------------- 1 259 2335.8935 (0.0000) S>CV3.1(84) application_data --------------------------------------------------------------- From: "Matthias" ;tag=rllbt2d9pz --------------------------------------------------------------- 1 260 2335.8935 (0.0000) S>CV3.1(85) application_data --------------------------------------------------------------- To: ;tag=vjyU4aSFye19F --------------------------------------------------------------- 1 261 2335.8935 (0.0000) S>CV3.1(72) application_data --------------------------------------------------------------- Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE --------------------------------------------------------------- 1 262 2335.8935 (0.0000) S>CV3.1(542) application_data --------------------------------------------------------------- User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="XXXX.dyndns.org", nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" Content-Length: 0 --------------------------------------------------------------- 1 263 2336.1083 (0.2147) C>SV3.1(438) application_data --------------------------------------------------------------- ACK sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" ;tag=rllbt2d9pz To: ;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 ACK Max-Forwards: 70 Contact: ;reg-id=1 Content-Length: 0 --- here, the 2nd INVITE packet is received, but it doesn't arrive at FS --- --------------------------------------------------------------- 1 2336.1229 (0.0145) S>C TCP FIN 1 264 2336.2127 (0.0898) C>SV3.1(1561) application_data --------------------------------------------------------------- INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="1001",realm="XXXX.dyndns.org",nonce="a3346430-8ce9-11df-9347-29b67737619e",uri="sip:9999 at XXXX.dyndns.org;user=phone",qop=auth,nc=00000001,cnonce="4fd48758",response="bb47aafad2fc3980d7540b0d2cbf1b03",algorithm=MD5 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --------------------------------------------------------------- --- a new TCP/SSL connection is initiated for the next packet --- 1 2336.2128 (0.0001) C>S TCP FIN New TCP connection #7: XXX.XXX.XXX (4655) <-> XXX.XXX.XXX (5061) 7 1 0.1214 (0.1214) C>SV3.1(63) Handshake ClientHello Version 3.1 random[32]= 3c 26 79 5c 23 37 ce ec 6b 40 cc 95 14 16 f8 e9 fd 5e 36 da dd 56 ab 2a 8e 0e 15 aa 4c d7 93 98 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_NULL_MD5 TLS_RSA_WITH_NULL_SHA TLS_DH_anon_WITH_3DES_EDE_CBC_SHA TLS_DH_anon_WITH_RC4_128_MD5 TLS_RSA_WITH_DES_CBC_SHA TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_DH_anon_WITH_DES_CBC_SHA compression methods NULL 7 2 0.1219 (0.0005) S>CV3.1(74) Handshake ServerHello Version 3.1 random[32]= 4c 39 bc 05 60 47 db 31 51 42 29 ae 9f 45 c5 2f 56 5b 02 dd 4b 3b c1 f7 f3 b7 80 9b 8b 16 7a a0 session_id[32]= f5 49 8b 05 df 1d 7a 30 6d b5 58 79 b6 a3 e4 1d c2 c1 98 fc de f5 64 8f 85 cc 7c 7f a9 08 9f d7 cipherSuite TLS_RSA_WITH_RC4_128_SHA compressionMethod NULL 7 3 0.1219 (0.0000) S>CV3.1(644) Handshake Certificate 7 4 0.1219 (0.0000) S>CV3.1(4) Handshake ServerHelloDone 7 5 0.4413 (0.3193) C>SV3.1(134) Handshake ClientKeyExchange EncryptedPreMasterSecret[128]= 38 71 d5 4f 2f 4f 66 08 37 92 57 ef 4a bd 13 60 6f 7c 0f 23 47 83 a5 59 95 25 91 58 10 c2 b9 ea 8d 12 88 d9 33 06 a8 8c aa e6 17 b8 91 35 0f 56 49 a7 9f 77 04 9c fe b1 17 70 c0 a3 5f bc 77 ff 25 41 a5 2f cd f3 24 d7 3c da 9f a7 86 30 a8 64 7d ca cd 06 6c d4 a0 fc 18 b7 5a 04 69 45 ae 28 4e 0f bf 3c 49 87 fc f6 91 3b 1d 00 6e 81 46 5f e3 7e cd a2 40 b2 37 3b c3 fc 0a 17 8b aa 76 88 7 6 0.4413 (0.0000) C>SV3.1(1) ChangeCipherSpec 7 7 0.4413 (0.0000) C>SV3.1(36) Handshake Finished verify_data[12]= 64 c2 1f cb 54 7c ec 60 0a 45 db e4 7 8 0.4504 (0.0091) S>CV3.1(1) ChangeCipherSpec 7 9 0.4504 (0.0000) S>CV3.1(36) Handshake Finished verify_data[12]= 17 f3 73 52 66 96 4a 87 ba 6d 42 f5 7 10 0.5550 (0.1046) C>SV3.1(406) application_data --------------------------------------------------------------- CANCEL sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" ;tag=rllbt2d9pz To: Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Content-Length: 0 --------------------------------------------------------------- 7 11 0.5559 (0.0008) S>CV3.1(163) application_data --------------------------------------------------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=78.53.13.206 --------------------------------------------------------------- 7 12 0.5560 (0.0000) S>CV3.1(84) application_data --------------------------------------------------------------- From: "Matthias" ;tag=rllbt2d9pz --------------------------------------------------------------- 7 13 0.5560 (0.0000) S>CV3.1(85) application_data --------------------------------------------------------------- To: ;tag=XUQm659jUQQvB --------------------------------------------------------------- 7 14 0.5560 (0.0000) S>CV3.1(72) application_data --------------------------------------------------------------- Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100813/28bc92d6/attachment-0001.html From a.afzali2003 at gmail.com Sat Aug 14 06:40:41 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sat, 14 Aug 2010 18:10:41 +0430 Subject: [Freeswitch-users] Limits on number of concurrent registered users? In-Reply-To: References: <4C653AC7.8010009@infosecurity.ch> Message-ID: Why you don't use SER family to handle such huge registrations and integrate it with FreeSWITCH for service implementations ? -- afshin On Sat, Aug 14, 2010 at 4:55 AM, Eliot Gable > wrote: > There are no built-in limits on the number of registered users. The > only thing that matters in terms of limits is how fast of a server you > deploy FreeSWITCH on. Obviously, handling a register request takes > some processing power. Handling requests from 15,000 users will take a > decent amount of processing power. The amount it takes depends on how > often a user re-registers. If they are doing it once per hour then you > will need to be able to process just over 4 per second, which is not > substantial at all. If they register every 15 seconds, then you are > looking at 1,000 per second, which is something else entirely. > > As for TLS/SSL, I have used it for some time and never had an issue > with the phone disconnecting. I tested using a Polycom phone. > > On Fri, Aug 13, 2010 at 2:26 PM, Yehavi Bourvine > wrote: > > Do you have phones already connected with TLS? I tried doing so for SNOM > and > > Polycom, and both disconnect after a while (and re-connect later). I > > couldn't decide so far whether it is a FreeSwitch or openSSL issue. > > > > Regards, __Yehavi: > > > > 2010/8/13 Fabio Pietrosanti (naif) > >> > >> Hi all, > >> > >> i am preparing a setup where i would have a huge number of connected > >> users via SIP/TLS . > >> > >> The amount of concurrent calls will not be such high. > >> > >> However he amount of concurrently registered users could be even of > 15.000 > >> . > >> > >> Are there some specific internal design/architectural limits of > >> FreeSWITCH on the number of concurrent users? > >> > >> Fabio > >> > >> p.s. we'll try to do some stress testing next week > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Eliot Gable > > "We do not inherit the Earth from our ancestors: we borrow it from our > children." ~David Brower > > "I decided the words were too conservative for me. We're not borrowing > from our children, we're stealing from them--and it's not even > considered to be a crime." ~David Brower > > "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to > live; not live to eat.) ~Marcus Tullius Cicero > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/85aca9e4/attachment.html From mnhassan at usa.net Sat Aug 14 06:41:14 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sat, 14 Aug 2010 19:41:14 +0600 Subject: [Freeswitch-users] PlayAndGetDigits reset input In-Reply-To: <546981.582.qm@web29216.mail.ird.yahoo.com> References: <546981.582.qm@web29216.mail.ird.yahoo.com> Message-ID: How about using ** as the character sequence to reset, and # as the terminating character. Then, once you have the digit sequence returned by play_and_get_digits, you can strip off everything left to the last ** sequence via a script? Regards HASSAN On Sat, Aug 14, 2010 at 17:14, Nigel Kent wrote: > I think what you suggest would be better - to listen for the digits. > > Do you have any resource or tips that I can check to see how to implement > listening and catching a desired sequence > > --- On *Fri, 13/8/10, Michael Collins * wrote: > > > From: Michael Collins > Subject: Re: [Freeswitch-users] PlayAndGetDigits reset input > To: "FreeSWITCH Users Help" > Date: Friday, 13 August, 2010, 20:25 > > > > > On Fri, Aug 13, 2010 at 1:46 AM, Nigel Kent > > wrote: > > Using PlayAndGetDigits I want to give option to the user to reset their > input by entering a certain key sequence (##) > > How can I make PlayAndGetDigits reprocess immediatly if this key squenqence > is pressed. > > I don't believe playandgetdigits does this. You will have to listen for ## > (or whatever) and then loop back and re-execute playandgetdigits. > -MC > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/f0dc17ae/attachment.html From dujinfang at gmail.com Sat Aug 14 06:49:29 2010 From: dujinfang at gmail.com (Seven Du) Date: Sat, 14 Aug 2010 21:49:29 +0800 Subject: [Freeswitch-users] PlayAndGetDigits reset input In-Reply-To: <546981.582.qm@web29216.mail.ird.yahoo.com> References: <546981.582.qm@web29216.mail.ird.yahoo.com> Message-ID: You can get the input digits immediately as long as you desired sequence ending with a "terminator" key(say #), then you will get all the digits in a channel var you can test against if it matches you desired sequence or not and decided if you want to start over the play_and_get_digits again. On Sat, Aug 14, 2010 at 7:14 PM, Nigel Kent wrote: > I think what you suggest would be better - to listen for the digits. > > Do you have any resource or tips that I can check to see how to implement > listening and catching a desired sequence > > --- On *Fri, 13/8/10, Michael Collins * wrote: > > > From: Michael Collins > Subject: Re: [Freeswitch-users] PlayAndGetDigits reset input > To: "FreeSWITCH Users Help" > Date: Friday, 13 August, 2010, 20:25 > > > > > On Fri, Aug 13, 2010 at 1:46 AM, Nigel Kent > > wrote: > > Using PlayAndGetDigits I want to give option to the user to reset their > input by entering a certain key sequence (##) > > How can I make PlayAndGetDigits reprocess immediatly if this key squenqence > is pressed. > > I don't believe playandgetdigits does this. You will have to listen for ## > (or whatever) and then loop back and re-execute playandgetdigits. > -MC > > > -----Inline Attachment Follows----- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/de53e23d/attachment.html From sos at sokhapkin.dyndns.org Sat Aug 14 08:19:07 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sat, 14 Aug 2010 11:19:07 -0400 Subject: [Freeswitch-users] Strange early media SDP behavior Message-ID: <201008141119.07759.sos@sokhapkin.dyndns.org> Late negotiation is on, bypass_media=true. Leg B receives the following SDP in SIP 183 message: v=0 o=root 4913 4913 IN IP4 64.21.13.41 s=session c=IN IP4 64.21.13.41 t=0 0 m=audio 37650 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 3 codecs in SDP offer. Freeswitch send to leg A SIP 183 with the following SDP: v=0 o=root 4913 4913 IN IP4 64.21.13.41 s=session c=IN IP4 64.21.13.41 t=0 0 m=audio 37650 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 1 codec only. Why 2 other codecs are removed from SDP? Tested on today's git, older versions have the same behavior. Is it expected behavior or a bug? From ktngl at yahoo.co.uk Sat Aug 14 07:26:33 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Sat, 14 Aug 2010 14:26:33 +0000 (GMT) Subject: [Freeswitch-users] PlayAndGetDigits reset input In-Reply-To: Message-ID: <263379.53424.qm@web29208.mail.ird.yahoo.com> That a good idea. Have (#) as a terminator and test for (#) as the last entered digit but I have further looked into my requirement and I would need to catch the key presses (##) on a active call also. Any ideas on that --- On Sat, 14/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] PlayAndGetDigits reset input To: "FreeSWITCH Users Help" Date: Saturday, 14 August, 2010, 13:49 You can get the input digits immediately as long as you?desired?sequence ending with a "terminator" key(say #), then you will get all the digits in a channel var you can test against if it matches you desired sequence or not and decided if you want to start over the play_and_get_digits again.?? On Sat, Aug 14, 2010 at 7:14 PM, Nigel Kent wrote: I think what you suggest would be better - to listen for the digits. Do you have any resource or tips that I can check to see how to implement listening and catching a desired sequence --- On Fri, 13/8/10, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] PlayAndGetDigits reset input To: "FreeSWITCH Users Help" Date: Friday, 13 August, 2010, 20:25 On Fri, Aug 13, 2010 at 1:46 AM, Nigel Kent wrote: Using PlayAndGetDigits I want to give option to the user to reset their input by entering a certain key sequence (##) How can I make PlayAndGetDigits reprocess immediatly if this key squenqence is pressed. I don't believe playandgetdigits does this. You will have to listen for ## (or whatever) and then loop back and re-execute playandgetdigits. -MC -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/099beb2f/attachment.html From kris at kriskinc.com Sat Aug 14 08:29:26 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 14 Aug 2010 11:29:26 -0400 Subject: [Freeswitch-users] Strange early media SDP behavior Message-ID: <0b17f5b9a6694ff22d394fdbbe7e2102@mail.gmail.com> It's standard SDP offer/answer. Fresswitch will not send three audio streams with three codecs. You're advertising three codecs with PCMU first. Freeswitch supports PCMU in your configuration and agrees on that codec. There are several configuration options in freeswitch and the phone that can influence this behavior. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Sat Aug 14 11:19:07 2010 Subject: [Freeswitch-users] Strange early media SDP behavior Late negotiation is on, bypass_media=true. Leg B receives the following SDP in SIP 183 message: v=0 o=root 4913 4913 IN IP4 64.21.13.41 s=session c=IN IP4 64.21.13.41 t=0 0 m=audio 37650 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 3 codecs in SDP offer. Freeswitch send to leg A SIP 183 with the following SDP: v=0 o=root 4913 4913 IN IP4 64.21.13.41 s=session c=IN IP4 64.21.13.41 t=0 0 m=audio 37650 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 1 codec only. Why 2 other codecs are removed from SDP? Tested on today's git, older versions have the same behavior. Is it expected behavior or a bug? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sos at sokhapkin.dyndns.org Sat Aug 14 08:36:20 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sat, 14 Aug 2010 11:36:20 -0400 Subject: [Freeswitch-users] Strange early media SDP behavior In-Reply-To: <0b17f5b9a6694ff22d394fdbbe7e2102@mail.gmail.com> References: <0b17f5b9a6694ff22d394fdbbe7e2102@mail.gmail.com> Message-ID: <201008141136.20600.sos@sokhapkin.dyndns.org> I would disagree, in bypass media mode FS is supposed to forward SDP as is, let endpoints to negotiate the codec. On Saturday 14 August 2010, Kristian Kielhofner wrote: > It's standard SDP offer/answer. Fresswitch will not send three audio > streams with three codecs. You're advertising three codecs with PCMU > first. Freeswitch supports PCMU in your configuration and agrees on that > codec. > > There are several configuration options in freeswitch and the phone that > can influence this behavior. > > > -- > Kristian Kielhofner > http://blog.krisk.org > From sos at sokhapkin.dyndns.org Sat Aug 14 08:47:24 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sat, 14 Aug 2010 11:47:24 -0400 Subject: [Freeswitch-users] Strange early media SDP behavior In-Reply-To: <201008141136.20600.sos@sokhapkin.dyndns.org> References: <0b17f5b9a6694ff22d394fdbbe7e2102@mail.gmail.com> <201008141136.20600.sos@sokhapkin.dyndns.org> Message-ID: <201008141147.24427.sos@sokhapkin.dyndns.org> Also when leg B answers the call (with the same 3 codecs), FS sends 200 OK to leg A with all 3 codecs, so codecs strip happens for early media only. The problem is that leg A early_sdp differs from local_sdp and mod_sofia disables SOA support which leads to problems later, when leg A uac sends SIP session timer reinvite. On Saturday 14 August 2010, Sergey Okhapkin wrote: > I would disagree, in bypass media mode FS is supposed to forward SDP as is, > let endpoints to negotiate the codec. > > On Saturday 14 August 2010, Kristian Kielhofner wrote: > > It's standard SDP offer/answer. Fresswitch will not send three audio > > streams with three codecs. You're advertising three codecs with PCMU > > first. Freeswitch supports PCMU in your configuration and agrees on that > > codec. > > > > There are several configuration options in freeswitch and the phone that > > can influence this behavior. > > > > > > -- > > Kristian Kielhofner > > http://blog.krisk.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kris at kriskinc.com Sat Aug 14 08:48:58 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 14 Aug 2010 11:48:58 -0400 Subject: [Freeswitch-users] Strange early media SDP behavior Message-ID: <8958bbff3a5162a77627db006db48e27@mail.gmail.com> I missed the part about bypass_media but offer/answer stays the same. In this case the remote end is selecting PCMU from the advertised codec list. What is it configured to do? Do you have a full trace showing both legs? It's not clear from the few SDPs you've posted which endpoint is which, where fs is, etc. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Sat Aug 14 11:36:20 2010 Subject: Re: [Freeswitch-users] Strange early media SDP behavior I would disagree, in bypass media mode FS is supposed to forward SDP as is, let endpoints to negotiate the codec. On Saturday 14 August 2010, Kristian Kielhofner wrote: > It's standard SDP offer/answer. Fresswitch will not send three audio > streams with three codecs. You're advertising three codecs with PCMU > first. Freeswitch supports PCMU in your configuration and agrees on that > codec. > > There are several configuration options in freeswitch and the phone that > can influence this behavior. > > > -- > Kristian Kielhofner > http://blog.krisk.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From david.ponzone at ipeva.fr Sat Aug 14 09:15:07 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Sat, 14 Aug 2010 18:15:07 +0200 Subject: [Freeswitch-users] PlayAndGetDigits reset input In-Reply-To: <546981.582.qm@web29216.mail.ird.yahoo.com> References: <546981.582.qm@web29216.mail.ird.yahoo.com> Message-ID: <8C74E9E1-7188-4631-AD63-2424C3EF1BC4@ipeva.fr> Nigel, Many solutions were given (listen to the whole seq and strip everyhing at the left of the **/##, or listen one dtmf at a time, concat that to previous dtmf, check if last 2 are **/##, if so empty the variable, go on listening to dtmf until required length/string is matched). Basically, the right solution(s) probably relies on how you may detect the the sequence entered is finished: terminator ? max length ? timeout ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 14/08/2010 ? 13:14, Nigel Kent a ?crit : > I think what you suggest would be better - to listen for the digits. > > Do you have any resource or tips that I can check to see how to > implement listening and catching a desired sequence > > --- On Fri, 13/8/10, Michael Collins wrote: > > From: Michael Collins > Subject: Re: [Freeswitch-users] PlayAndGetDigits reset input > To: "FreeSWITCH Users Help" > Date: Friday, 13 August, 2010, 20:25 > > > > On Fri, Aug 13, 2010 at 1:46 AM, Nigel Kent wrote: > Using PlayAndGetDigits I want to give option to the user to reset > their input by entering a certain key sequence (##) > > How can I make PlayAndGetDigits reprocess immediatly if this key > squenqence is pressed. > > I don't believe playandgetdigits does this. You will have to listen > for ## (or whatever) and then loop back and re-execute > playandgetdigits. > -MC > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/869deb84/attachment-0001.html From mike at jerris.com Sat Aug 14 09:20:44 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:20:44 -0400 Subject: [Freeswitch-users] FreeTDM and Digium analog cards In-Reply-To: References: Message-ID: <8F067F27-529B-4638-BD04-DC3A62B0EBA4@jerris.com> yes, the analog cards are supported as well. Mike On Jul 5, 2010, at 1:04 PM, Joseph L. Casale wrote: >> I don't understand your question. TDM refers to http://en.wikipedia.org/wiki/Time-division_multiplexing >> >> T1/E1 use TDM. So what do you mean by digital variants? You make it sound like TDM cards and the digital variants you talk about are different >things. >> >> FreeTDM support analog and digital cards for both Sangoma and Digium. There is some PIKA support but it may be broken by now since few or no >people seems to use it. > > Moises, > Thanks for taking the time to help. Sorry I wasn?t clear, what I meant > by referring to digital cards was E1/T1 based cards versus the analog > variants that function on the PSTN like a Digium TDM410. > > Tonight I am upgrading some hardware and will recompile fs, so I wondered > if I should use this opportunity to move from OpenZAP to FreeTDM but I > really couldn?t find a definitive answer wrt to its support for these > cards, all the docs and examples explicitly refer to the digital cards. > > I guess looking at http://wiki.sangoma.com/wanpipe-api-freetdm it does > show trunk_type definitions for FXO/S so I guess it does! > From mike at jerris.com Sat Aug 14 09:21:50 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:21:50 -0400 Subject: [Freeswitch-users] using FIFO to queue calls towards a gateway In-Reply-To: References: Message-ID: That should work fine. If you need more than 1 call per dial string, you can use the "simo" param. Mike On Jul 5, 2010, at 2:18 PM, afshin afzali wrote: > Hi FreeSWITCH, > > Suppose there is a small voice gateway that just has been equipped with a few FXO ports which connected to PSTN's analog lines. I'm going to use FIFO to manage call requests which going through this voice gateway. As I've learned about mod_fifo service, I think that the queue should have statically defined members (one for each port) which in their dial strings target that gateway (parametrized with the PSTN phone number by a session variable) and nothing else! Am I wrong about this? From mike at jerris.com Sat Aug 14 09:29:14 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:29:14 -0400 Subject: [Freeswitch-users] Receiving incoming SIP calls In-Reply-To: <20100704161433.GF29534@apple.rat.burntout.org> References: <20100704161433.GF29534@apple.rat.burntout.org> Message-ID: <65564BA9-6D46-4CCB-A8CB-6442EFED9D2B@jerris.com> This is normal, it will go on to try to authenticate the call on this profile with auth enabled. It is just saying it has not passed the ACL used to bypass authentication. Mike On Jul 4, 2010, at 12:14 PM, Alan Dawson wrote: > Hi, > > I'm new to VOIP, and freeswitch. > > I'm trying to enable certain internal extensions to receive incoming SIP calls, > eg people can dial sip:1001 at my.domain and be able to receive those calls. > but when they do this I get > > Rejected by acl "domains". > > log message. From mike at jerris.com Sat Aug 14 09:31:01 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:31:01 -0400 Subject: [Freeswitch-users] sofia_contact response via socket In-Reply-To: <000c01cb1ce7$03a81000$0af83000$@de> References: <000c01cb1ce7$03a81000$0af83000$@de> Message-ID: Yes, it is intended. sofia_contact is used inline to fill in the dialplan. It should not cause any problems with parsing the message correctly, you can confirm this is the case by running sofia_contact from fs_cli and see that it is handled properly. On Jul 6, 2010, at 4:41 AM, Oliver Sch?nbeck wrote: > I recently noticed that the response send after a call to sofia_contact via socket connection lags the two \n at the end. Which leads to some Problems parsing the messages correctly. > Is this behaviour intended? > > I?m using sofia_contact to check whether a user is currently registered or not so I am in the need to call this function via socket. > Is there an alternative? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/1ec677d5/attachment.html From mike at jerris.com Sat Aug 14 09:35:57 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:35:57 -0400 Subject: [Freeswitch-users] mod_conference dynamic config options In-Reply-To: <4C33E1C3.4040508@communicatefreely.net> References: <4C33E1C3.4040508@communicatefreely.net> Message-ID: you would either need to provide a patch for mod_conference to pull that value (maybe from a channel var, the args string is already quite cluttered on that app), or serve dynamic config as you suggest below. Mike On Jul 6, 2010, at 10:09 PM, Tim St. Pierre wrote: > I am trying to dynamically set the moh_sound variable in mod_conference, so that each customer can > have a specific music on hold for their company conference room. > > Most of the options are common, so I have a few template profiles in the conference.conf.xml file. > I can change the moh_sound variable to suit, but this would require me to create a profile for each > room, then reload it. I would prefer to have this pull from a database. > > I'm using LUA in the dialplan to setup the conference based on the database values, but I can only > see options for setting things like PIN, profile, etc. Is there any way I can set this or other > settings with a dialplan variable? > > Failing that, can mod_conference be bound to a lua script that will give it an XML profile each time > it is called? From mike at jerris.com Sat Aug 14 09:38:52 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:38:52 -0400 Subject: [Freeswitch-users] VM Force Password Change In-Reply-To: <4C33E095.8080400@communicatefreely.net> References: <016301cb1d0a$f38c2e80$daa48b80$@com> <01bc01cb1d23$93213900$b963ab00$@com> <4C33E095.8080400@communicatefreely.net> Message-ID: <454DD96A-F519-45AD-BC80-AD5E53D92711@jerris.com> Yes, you get a callback via your directory hook when they change their password. Mike On Jul 6, 2010, at 10:04 PM, Tim St. Pierre wrote: > If you use this method, what happens when a user tries to change their password from within the > voicemail app? Is there some sort of hook in mod_voicemail that we can use to push the update to > the database via the web server? > > -Tim From mike at jerris.com Sat Aug 14 09:41:07 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:41:07 -0400 Subject: [Freeswitch-users] mod_xml_curl question. In-Reply-To: <201007070932.42832.sos@sokhapkin.dyndns.org> References: <201007070932.42832.sos@sokhapkin.dyndns.org> Message-ID: set your dialplan param in your endpoint to be "XML:/path/to/static/file,XML" Mike On Jul 7, 2010, at 9:32 AM, Sergey Okhapkin wrote: > I want to use mod_xml_curl to retrieve only dynamic dialplan sections, but to > keep a large chunk of dialplan code in a static XML file to minimize the number > of http requests. But when mod_xml_curl is loaded, dialplan search/execution > starts with xml_curl execution. How to avoid that? I need static xml to be > looked up/executed first, and retrieve dialplan context from web server only if > the required dialplan context is not defined in static XML file. From mike at jerris.com Sat Aug 14 09:45:17 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:45:17 -0400 Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1278543145818-5267704.post@n2.nabble.com> References: <1278436848678-5261486.post@n2.nabble.com> <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> <1278517888279-5265912.post@n2.nabble.com> <1278543145818-5267704.post@n2.nabble.com> Message-ID: I had the same issue with other uclibc builds, we had a patch for uclibc in astlinux that fixes this, but I believe it was fixed in newer uclibc as well. This was some header bug in uclibc iirc. Mike On Jul 7, 2010, at 6:52 PM, mazilo wrote: > > Not really. At least, we gave it some tries. I believe it is not the problem > on FS, but rather a problem to the gcc-4.3.3 compiler with CodeSourcery > enhancements for OpenWRT. That said, if anyone wants to try, why not. > From 12ukwn at gmail.com Sat Aug 14 09:47:21 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 14 Aug 2010 18:47:21 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: <4C66711A.3040006@coppice.org> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> <4C662E20.8090209@coppice.org> <4C66711A.3040006@coppice.org> Message-ID: <20100814184721.53777e2c@anubis.defcon1> Le Sat, 14 Aug 2010 18:34:02 +0800, Steve Underwood a ?crit : > Decoding and recoding G.729 leads to an awful quality loss. It best What about decoding and recoding, lets says in iLBC? > avoided. If you really want to mix the streams, leave the output at a > bit higher rate in the subsequent recording if you want the conversation > to be heard in detail later on. If I understand this thread well, I'd need only one licence if I receive a G.729 call but want my internal network to stay in iLBC or G.722? JY -- Hand me a pair of leather pants and a CASIO keyboard -- I'm living for today! From mike at jerris.com Sat Aug 14 09:51:22 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 12:51:22 -0400 Subject: [Freeswitch-users] Problem of FreeSwitch and T.38 Test In-Reply-To: References: <5ECC3A8A-CDF9-4EF4-AD87-A3FAEDFF26AB@freeswitch.org> <12B00ED2-34A6-414C-A0C6-590A963E3CD2@gmail.com> Message-ID: <04663062-1656-4943-9DE5-50514A6851C0@jerris.com> http://wiki.freeswitch.org/wiki/T.38 On Jul 8, 2010, at 11:15 AM, chi zhang wrote: > Hi, David > Thanks for your patient...:) > In fact, i want to do two things. > 1st is: one sip user of FS receives a fax one PSTN user. > 2st is: one sip user of FS sends a fax to PSTN user. > In my test environment, i can only do this test in Voip system without T38 gateway nor real fax machine. So i can only simulate the fax receive/send process with two softphone. > Previously, i do one sip call from zoiper(1000) to 9178( receive fax's dialplan of default.xml ). It can be regarded as the receive fax process of FS. > Now, i want to debug the send fax source of FS, such as: user 2000 SEND fax to user 1000, and 1000 receives the fax and saves it. > From david.ponzone at ipeva.fr Sat Aug 14 09:55:27 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Sat, 14 Aug 2010 18:55:27 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: <20100814184721.53777e2c@anubis.defcon1> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> <4C662E20.8090209@coppice.org> <4C66711A.3040006@coppice.org> <20100814184721.53777e2c@anubis.defcon1> Message-ID: <82BE6096-BE27-4370-89A4-8E309D7685B2@ipeva.fr> If you can accept a delay before the recordings are available (the time to decode from G729 to something else and then mix the 2 streams), yes. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 14/08/2010 ? 18:47, Jean-Yves F. Barbier a ?crit : > Le Sat, 14 Aug 2010 18:34:02 +0800, > Steve Underwood a ?crit : > >> Decoding and recoding G.729 leads to an awful quality loss. It best > > What about decoding and recoding, lets says in iLBC? > >> avoided. If you really want to mix the streams, leave the output at a >> bit higher rate in the subsequent recording if you want the >> conversation >> to be heard in detail later on. > > If I understand this thread well, I'd need only one licence if I > receive > a G.729 call but want my internal network to stay in iLBC or G.722? > > JY > -- > Hand me a pair of leather pants and a CASIO keyboard -- I'm living for > today! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/dba56ce0/attachment.html From mike at jerris.com Sat Aug 14 10:10:16 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 13:10:16 -0400 Subject: [Freeswitch-users] Where to install sample scripts. In-Reply-To: <20100713181517.GB17806@hijacked.us> References: <333819028.136.1279039130967.JavaMail.root@srvr12.remotelinkml.com> <1503719566.140.1279039569183.JavaMail.root@srvr12.remotelinkml.com> <20100713181517.GB17806@hijacked.us> Message-ID: This is the same issue we have with all the examples in "scripts" . Likely I will start installing those to /usr/share somewhere. Can people comment on where these typically go on different distros? Mike On Jul 13, 2010, at 2:15 PM, Andrew Thompson wrote: > > To be honest, I hadn't considered this question at all. I included > freeswitch.erl mainly as an example of how to use the API, and as a way > to abastract some of the low level annoyances of async message passing > when you want synchronous behaviour. I never really thought about > treating it as a system lib. The ESL libs don't seem to auto-install > themselves (the wiki is unclear on that though). > > I could easily compile freeswitch.erl into a beam and install the src > and beam into /usr/local/lib/erlang/lib/freeswitch-1.0 or whatever, I > suppose, but is that even what people want? I'm hesitant to pollute the > erlang lib directory with my own stuff. Also, how do we do versioning? > When building from git all we get is 1.0.head or whatever. Should I just > override any installed version with a new one? > > Anyone have any thoughts? (Assuming anyone actually *uses* > freeswitch.erl besides myself). > > Andrew From 12ukwn at gmail.com Sat Aug 14 10:15:40 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 14 Aug 2010 19:15:40 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: <82BE6096-BE27-4370-89A4-8E309D7685B2@ipeva.fr> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> <4C662E20.8090209@coppice.org> <4C66711A.3040006@coppice.org> <20100814184721.53777e2c@anubis.defcon1> <82BE6096-BE27-4370-89A4-8E309D7685B2@ipeva.fr> Message-ID: <20100814191540.57a8caa3@anubis.defcon1> Le Sat, 14 Aug 2010 18:55:27 +0200, David Ponzone a ?crit : > If you can accept a delay before the recordings are available (the > time to decode from G729 to something else and then mix the 2 > streams), yes. No, that wasn't what I meant: I'd like to know if only one licence is enough if my provider's SDSL send me G.729 encoded comms; in fact: do I only need one license per comm to transcode G.729 into iLBC which is the prefered & main CODEC of my internal network? JY -- It's not Camelot, but it's not Cleveland, either. -- Kevin White, Mayor of Boston From mike at jerris.com Sat Aug 14 10:36:58 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Aug 2010 13:36:58 -0400 Subject: [Freeswitch-users] Howto install ESL lua, perl, python, ruby mods? In-Reply-To: <1279116449.1886.53.camel@anthony-desktop> References: <4C3CA8AB.5050104@puzzled.xs4all.nl> <6F4FF7BA-F5CC-4740-8089-CD51E7345674@freeswitch.org> <4C3CB9CE.1010404@puzzled.xs4all.nl> <4C3CC84F.8020207@puzzled.xs4all.nl> <1279052598.1886.13.camel@anthony-desktop> <4C3D76B5.9080103@puzzled.xs4all.nl> <1279116449.1886.53.camel@anthony-desktop> Message-ID: <62639CC0-5109-4D44-A749-E453358F29FD@jerris.com> could someone open a bug on jira about this, perferabally with details where each needs to be installed so we can fix the build system and packages. Mike On Jul 14, 2010, at 10:07 AM, Anthony Cosgrove wrote: > On Wed, 2010-07-14 at 10:35 +0200, Patrick Lists wrote: >> I can *make* the modules fine. The *installation* instructions are >> missing and that is what I'm looking for. >> >> Thanks, >> Patrick > > Sorry. Yeh... that part is missing. Ok so you'll want to take the > wrapper .so and the module for the particular language you are > interested in an place it in your site include locations. I wish I could > be more specific but every distro does it differently > > Perl: Copy ESL.so, ESL.pm, IVR.pm, Dispatch.pm to a location defined in > your @INC array. To see what's in @INC run perl -e "print join(\"\n\", > @INC);" > > Lua: I don't do much lua so not all of this may be accurate. Make sure > lua is installed on your system (luamod wouldn't compile for me > otherwise). ESL.so appears to have to be in the same folder as your > script. > > Python: Same thing. Either ESL.py and _ESL.so need to be in the same > folder or in your PYTHONPATH. You can get your path by invoking python > at the command line. > >>>> import sys >>>> print sys.path > ['', '/usr/lib64/python26.zip', '/usr/lib64/python2.6', > '/usr/lib64/python2.6/plat-linux2', '/usr/lib64/python2.6/lib-tk', > '/usr/lib64/python2.6/lib-old', '/usr/lib64/python2.6/lib-dynload', > '/usr/lib64/python2.6/site-packages', '/usr/lib64/portage/pym'] > From Nabble at slickdeals.endjunk.com Sat Aug 14 11:07:41 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Sat, 14 Aug 2010 11:07:41 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: References: <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> <1278517888279-5265912.post@n2.nabble.com> <1278543145818-5267704.post@n2.nabble.com> Message-ID: <1281809261831-5423774.post@n2.nabble.com> OpenWRT has created/included the patch for sofia.c file as pointed out above by http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5256137.html Anthony Minessale . Michael Jerris wrote: > > I had the same issue with other uclibc builds, we had a patch for uclibc > in astlinux that fixes this, but I believe it was fixed in newer uclibc as > well. This was some header bug in uclibc iirc. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5423774.html Sent from the freeswitch-users mailing list archive at Nabble.com. From david.ponzone at ipeva.fr Sat Aug 14 11:12:03 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Sat, 14 Aug 2010 20:12:03 +0200 Subject: [Freeswitch-users] g729 In-Reply-To: <20100814191540.57a8caa3@anubis.defcon1> References: <9DBDFB2B-9DA8-405D-8103-CB00CA1819AD@freeswitch.org> <579DFD39-65E5-4BDD-BFE7-95C711F899DD@freeswitch.org> <8B843454-C462-4821-A94C-0AB0252B92DD@freeswitch.org> <1281688434.16764.31.camel@luna.tc.commsmundi.com> <4C662E20.8090209@coppice.org> <4C66711A.3040006@coppice.org> <20100814184721.53777e2c@anubis.defcon1> <82BE6096-BE27-4370-89A4-8E309D7685B2@ipeva.fr> <20100814191540.57a8caa3@anubis.defcon1> Message-ID: <5CAC195A-519D-41B0-BE3C-40930B06D6BD@ipeva.fr> the original thread was about recording. for a regular call, you need one license per call (one licence call decode and encore). David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 14/08/2010 ? 19:15, Jean-Yves F. Barbier a ?crit : > Le Sat, 14 Aug 2010 18:55:27 +0200, > David Ponzone a ?crit : > >> If you can accept a delay before the recordings are available (the >> time to decode from G729 to something else and then mix the 2 >> streams), yes. > > No, that wasn't what I meant: I'd like to know if only one licence > is enough > if my provider's SDSL send me G.729 encoded comms; in fact: do I > only need > one license per comm to transcode G.729 into iLBC which is the > prefered & > main CODEC of my internal network? > > JY > -- > It's not Camelot, but it's not Cleveland, either. > -- Kevin White, Mayor of Boston > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/253ea5fb/attachment-0001.html From fraserredmond at gmail.com Sat Aug 14 12:01:09 2010 From: fraserredmond at gmail.com (Fraser Redmond) Date: Sat, 14 Aug 2010 20:01:09 +0100 Subject: [Freeswitch-users] Corrupted name for profile internal Message-ID: I've had something strange happening on one of my installations of FreeSwitch. After it's been running for a while (couple of days, on fairly low usage) my softphone no longer connects. In the console/log the failure has some unicode chars in the profile name, where I assume it should have 'internal' Doing a "sofia profile internal restart reloadxml" doesn't resolve it, but doing a full stop & start of freeswitch does fix it. Any ideas? 2010-08-14 19:04:32.235578 [WARNING] sofia_reg.c:2086 SIP auth failure (REGISTER) due to user-agent-filter. Filter "nat.auto" User-Agent "(SIPphone here)" 2010-08-14 19:04:32.235578 [WARNING] sofia_reg.c:1030 SIP auth failure (REGISTER) on sofia profile '??' for [username at 192.168.1.12] from ip 192.168.1.80 Cheers, Fraser -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/dbd9428f/attachment.html From mnhassan at usa.net Sat Aug 14 13:42:58 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sun, 15 Aug 2010 02:42:58 +0600 Subject: [Freeswitch-users] use of multiple templates and output files with mod_cdr_csv In-Reply-To: <1281688074.16764.25.camel@luna.tc.commsmundi.com> References: <1281605124.28815.239.camel@luna.tc.commsmundi.com> <1281688074.16764.25.camel@luna.tc.commsmundi.com> Message-ID: I don't think you can load the same module twice. I would suggest you take a look at mod_xml_cdr. It dumps a whole lot of information, which can be very useful for debugging as well. You're already using an external program to parse the CSV files. Just need it to parse the XML output, and you've got access to a whole lot more! Regards HASSAN On Fri, Aug 13, 2010 at 14:27, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > Thanks for the answers, > > > > Look at the accountcode channel variable: > > > http://wiki.freeswitch.org/wiki/Mod_cdr_csv#accountcode > > I initially wanted to use the accountcode variable as an account code > for billing and not for template selection, but I guess I can use it for > template and create another variable (ex: realaccountcode) for the > actual account code. > > Only thing would be that I'd be limited to two templates. Say I want to > log to CSV and a couple of database tables with different fields, well > I'll have to do it "manually". > > > > what I would do is to log all the fields you need to Master.csv, and > > then do selective imports of the fields you require depending on the > > usage. > > mysqlimport (LOAD DATA INFILE) allows that if you put a dummy variable > > (@dummy) instead of a column name. > > Nice idea thanks, I could also have my own CDR script listening for > hangup events, or to log everything to say an sqlite db and use a cron > script to create other logs out of it, but the LOAD DATA INFILE method > seems quite cool. > > Last question: Is there any way to load this module more than one time > with different base directories to make it simpler? > > Thanks again, > Fran?ois. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100815/29e8bd12/attachment.html From Prometheus001 at gmx.net Sat Aug 14 14:27:06 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 14 Aug 2010 23:27:06 +0200 Subject: [Freeswitch-users] TLS problem using SNOM phones In-Reply-To: References: Message-ID: <4C670A2A.6050806@gmx.net> Hello Matthias, I an working here with a number of 320 and 360 Phones with TLS and SRTP. This works well as longs as you have added ";transport=tls" to the outbound proxy and RTP/SAVP to "optional" in your SIP profile. Best regards Peter Matthias Reinacher schrieb: > Hello all, > i have a problem using different Snom phones (300/320/820/821) with > Freeswitch and TLS in a internet-wide setup (phones being registered > via internet at a FS w/ public IP). Sometimes (25-50% of cases) the > first try to call someone (internal) won't go through. Logs from > phone, Freeswitch and ssldump show that the phone sends an INVITE, the > FS asks for more credentials (digest auth b/c phone IP not in ACL), > the phone answers with an INVITE w/ more auth credentials -- and this > second INVITE package is not received by the FS. It does arrive at the > server though, as verified by ssldump. Interestingly, if one presses > "Cancel" on the phone, a new TCP/SSL connection is created, apparently > the old one died -- see logs from phone, FS, and ssldump below. Also, > Snom 820/821 show the registrar as not registered after such an action > (presumably b/c original SSL connection is dead). > > Has anybody encountered this behaviour and can verify it? Are there > any ideas what causes the problem here (FS, OpenSSL, phone?) and if > there is any remedy for it? I was planning on using FS as a production > phone system w/ encrypted signalling and audio using Snom phones. I > can't do that right now b/c of the abovementioned problem, and i have > not yet found a solution. > > Any help is greatly appreciated. > > TIA > Matthias > > > Logs: phone, FS, ssldump, please also see annotation enclosed in "--- > ... ---" > > - phone: > > Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:225 (1280 bytes): > > INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > Max-Forwards: 70 > Contact: > ;reg-id=1 > X-Serialnumber: 0004134517B7 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom821/8.4.4 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, > PRACK, MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 476 > > v=0 > o=root 739742427 739742427 IN IP4 192.168.0.114 > s=call > c=IN IP4 192.168.0.114 > t=0 0 > m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32 > inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:9 g722/8000 > a=rtpmap:99 g726-32/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=fmtp:18 annexb=no > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > > Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:363 (350 > bytes): > > SIP/2.0 100 Trying > Via: SIP/2.0/TLS > 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > > Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:524 (844 > bytes): > > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/TLS > 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 > From: "Matthias" ;tag=rllbt2d9pz > To: ;tag=vjyU4aSFye19F > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, refer > Proxy-Authenticate: Digest realm="XXXX.dyndns.org > ", > nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" > Content-Length: 0 > > > Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:528 (418 bytes): > > ACK sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport > From: "Matthias" ;tag=rllbt2d9pz > To: ;tag=vjyU4aSFye19F > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 ACK > Max-Forwards: 70 > Contact: > ;reg-id=1 > Content-Length: 0 > > --- the following packet is not received by FS --- > > > Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:544 (1541 bytes): > > INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-7g007jateb7f;rport > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 2 INVITE > Max-Forwards: 70 > Contact: > ;reg-id=1 > X-Serialnumber: 0004134517B7 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom821/8.4.4 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, > PRACK, MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Proxy-Authorization: Digest username="1001",realm="XXXX.dyndns.org > ",nonce="a3346430-8ce9-11df-9347-29b67737619e",uri="sip:9999 at XXXX.dyndns.org;user=phone",qop=auth,nc=00000001,cnonce="4fd48758",response="bb47aafad2fc3980d7540b0d2cbf1b03",algorithm=MD5 > Content-Type: application/sdp > Content-Length: 476 > > v=0 > o=root 739742427 739742427 IN IP4 192.168.0.114 > s=call > c=IN IP4 192.168.0.114 > t=0 0 > m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32 > inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:9 g722/8000 > a=rtpmap:99 g726-32/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=fmtp:18 annexb=no > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > --- before this packet is sent, a new TCP/SSL connection is negotiated --- > > > Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:29:936 (386 bytes): > > CANCEL sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 2 CANCEL > Max-Forwards: 70 > Reason: SIP;cause=487;text="Request terminated by user" > Content-Length: 0 > > > Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:30:562 (345 > bytes): > > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/TLS > 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=78.53.13.206 > From: "Matthias" ;tag=rllbt2d9pz > To: ;tag=XUQm659jUQQvB > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 2 CANCEL > Content-Length: 0 > > > > > > - Freeswitch: > > recv 1280 bytes from tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.774063: > > ------------------------------------------------------------------------ > INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > Max-Forwards: 70 > Contact: > ;reg-id=1 > X-Serialnumber: 0004134517B7 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom821/8.4.4 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, > PRACK, MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 476 > > > > v=0 > o=root 739742427 739742427 IN IP4 192.168.0.114 > s=call > c=IN IP4 192.168.0.114 > t=0 0 > m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32 > inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:9 g722/8000 > a=rtpmap:99 g726-32/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=fmtp:18 annexb=no > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > ------------------------------------------------------------------------ > send 350 bytes to tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.777861: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/TLS > 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=XXX.XXX.XXX.XXX > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > 2010-07-11 14:41:33.778081 [DEBUG] sofia.c:5999 IP XXX.XXX.XXX.XXX > Rejected by acl "domains". Falling back to Digest auth. > > send 844 bytes to tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.864787: > > ------------------------------------------------------------------------ > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/TLS > 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=XXX.XXX.XXX.XXX > From: "Matthias" ;tag=rllbt2d9pz > To: ;tag=vjyU4aSFye19F > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, refer > Proxy-Authenticate: Digest realm="XXXX.dyndns.org > ", > nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > recv 418 bytes from tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:34.082329: > > ------------------------------------------------------------------------ > ACK sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport > From: "Matthias" ;tag=rllbt2d9pz > To: ;tag=vjyU4aSFye19F > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 ACK > Max-Forwards: 70 > Contact: > ;reg-id=1 > Content-Length: 0 > > --- here, the 2nd INVITE packet w/ credentials is missing, note time > difference to next packet --- > > > > > ------------------------------------------------------------------------ > recv 386 bytes from tls/[XXX.XXX.XXX.XXX]:4655 at 12:41:41.905857: > > ------------------------------------------------------------------------ > CANCEL sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 2 CANCEL > Max-Forwards: 70 > Reason: SIP;cause=487;text="Request terminated by user" > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > send 345 bytes to tls/[XXX.XXX.XXX.XXX]:4655 at 12:41:41.906402: > > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/TLS > 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=XXX.XXX.XXX.XXX > From: "Matthias" ;tag=rllbt2d9pz > To: ;tag=XUQm659jUQQvB > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 2 CANCEL > Content-Length: 0 > > > > > - ssldump: > > 1 254 2335.8000 (417.3058) C>SV3.1(1300) application_data > --------------------------------------------------------------- > INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > Max-Forwards: 70 > Contact: > ;reg-id=1 > X-Serialnumber: 0004134517B7 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom821/8.4.4 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 476 > > > > v=0 > o=root 739742427 739742427 IN IP4 192.168.0.114 > s=call > c=IN IP4 192.168.0.114 > t=0 0 > m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32 > inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:9 g722/8000 > a=rtpmap:99 g726-32/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=fmtp:18 annexb=no > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > --------------------------------------------------------------- > 1 255 2335.8043 (0.0043) S>CV3.1(138) application_data > --------------------------------------------------------------- > SIP/2.0 100 Trying > Via: SIP/2.0/TLS > 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 > --------------------------------------------------------------- > 1 256 2335.8044 (0.0000) S>CV3.1(183) application_data > --------------------------------------------------------------- > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > --------------------------------------------------------------- > 1 257 2335.8045 (0.0000) S>CV3.1(89) application_data > --------------------------------------------------------------- > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Content-Length: 0 > > > > --------------------------------------------------------------- > 1 258 2335.8935 (0.0890) S>CV3.1(161) application_data > --------------------------------------------------------------- > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/TLS > 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 > --------------------------------------------------------------- > 1 259 2335.8935 (0.0000) S>CV3.1(84) application_data > --------------------------------------------------------------- > From: "Matthias" ;tag=rllbt2d9pz > --------------------------------------------------------------- > 1 260 2335.8935 (0.0000) S>CV3.1(85) application_data > --------------------------------------------------------------- > To: ;tag=vjyU4aSFye19F > --------------------------------------------------------------- > 1 261 2335.8935 (0.0000) S>CV3.1(72) application_data > --------------------------------------------------------------- > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 INVITE > --------------------------------------------------------------- > 1 262 2335.8935 (0.0000) S>CV3.1(542) application_data > --------------------------------------------------------------- > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, refer > Proxy-Authenticate: Digest realm="XXXX.dyndns.org > ", > nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" > Content-Length: 0 > > > > --------------------------------------------------------------- > 1 263 2336.1083 (0.2147) C>SV3.1(438) application_data > --------------------------------------------------------------- > ACK sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport > From: "Matthias" ;tag=rllbt2d9pz > To: ;tag=vjyU4aSFye19F > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 1 ACK > Max-Forwards: 70 > Contact: > ;reg-id=1 > Content-Length: 0 > > > > > --- here, the 2nd INVITE packet is received, but it doesn't arrive at > FS --- > > > > --------------------------------------------------------------- > 1 2336.1229 (0.0145) S>C TCP FIN > 1 264 2336.2127 (0.0898) C>SV3.1(1561) application_data > --------------------------------------------------------------- > INVITE sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-7g007jateb7f;rport > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 2 INVITE > Max-Forwards: 70 > Contact: > ;reg-id=1 > X-Serialnumber: 0004134517B7 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom821/8.4.4 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Proxy-Authorization: Digest username="1001",realm="XXXX.dyndns.org > ",nonce="a3346430-8ce9-11df-9347-29b67737619e",uri="sip:9999 at XXXX.dyndns.org;user=phone",qop=auth,nc=00000001,cnonce="4fd48758",response="bb47aafad2fc3980d7540b0d2cbf1b03",algorithm=MD5 > Content-Type: application/sdp > Content-Length: 476 > > > > v=0 > o=root 739742427 739742427 IN IP4 192.168.0.114 > s=call > c=IN IP4 192.168.0.114 > t=0 0 > m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32 > inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:9 g722/8000 > a=rtpmap:99 g726-32/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=fmtp:18 annexb=no > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > --------------------------------------------------------------- > > --- a new TCP/SSL connection is initiated for the next packet --- > > 1 2336.2128 (0.0001) C>S TCP FIN > New TCP connection #7: XXX.XXX.XXX (4655) <-> XXX.XXX.XXX (5061) > 7 1 0.1214 (0.1214) C>SV3.1(63) Handshake > ClientHello > Version 3.1 > random[32]= > 3c 26 79 5c 23 37 ce ec 6b 40 cc 95 14 16 f8 e9 > fd 5e 36 da dd 56 ab 2a 8e 0e 15 aa 4c d7 93 98 > cipher suites > TLS_RSA_WITH_RC4_128_MD5 > TLS_RSA_WITH_RC4_128_SHA > TLS_RSA_WITH_NULL_MD5 > TLS_RSA_WITH_NULL_SHA > TLS_DH_anon_WITH_3DES_EDE_CBC_SHA > TLS_DH_anon_WITH_RC4_128_MD5 > TLS_RSA_WITH_DES_CBC_SHA > TLS_RSA_EXPORT1024_WITH_RC4_56_SHA > TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA > TLS_DH_anon_WITH_DES_CBC_SHA > compression methods > NULL > 7 2 0.1219 (0.0005) S>CV3.1(74) Handshake > ServerHello > Version 3.1 > random[32]= > 4c 39 bc 05 60 47 db 31 51 42 29 ae 9f 45 c5 2f > 56 5b 02 dd 4b 3b c1 f7 f3 b7 80 9b 8b 16 7a a0 > session_id[32]= > f5 49 8b 05 df 1d 7a 30 6d b5 58 79 b6 a3 e4 1d > c2 c1 98 fc de f5 64 8f 85 cc 7c 7f a9 08 9f d7 > cipherSuite TLS_RSA_WITH_RC4_128_SHA > compressionMethod NULL > 7 3 0.1219 (0.0000) S>CV3.1(644) Handshake > Certificate > 7 4 0.1219 (0.0000) S>CV3.1(4) Handshake > ServerHelloDone > 7 5 0.4413 (0.3193) C>SV3.1(134) Handshake > ClientKeyExchange > EncryptedPreMasterSecret[128]= > 38 71 d5 4f 2f 4f 66 08 37 92 57 ef 4a bd 13 60 > 6f 7c 0f 23 47 83 a5 59 95 25 91 58 10 c2 b9 ea > 8d 12 88 d9 33 06 a8 8c aa e6 17 b8 91 35 0f 56 > 49 a7 9f 77 04 9c fe b1 17 70 c0 a3 5f bc 77 ff > 25 41 a5 2f cd f3 24 d7 3c da 9f a7 86 30 a8 64 > 7d ca cd 06 6c d4 a0 fc 18 b7 5a 04 69 45 ae 28 > 4e 0f bf 3c 49 87 fc f6 91 3b 1d 00 6e 81 46 5f > e3 7e cd a2 40 b2 37 3b c3 fc 0a 17 8b aa 76 88 > 7 6 0.4413 (0.0000) C>SV3.1(1) ChangeCipherSpec > 7 7 0.4413 (0.0000) C>SV3.1(36) Handshake > Finished > verify_data[12]= > 64 c2 1f cb 54 7c ec 60 0a 45 db e4 > > 7 8 0.4504 (0.0091) S>CV3.1(1) ChangeCipherSpec > 7 9 0.4504 (0.0000) S>CV3.1(36) Handshake > Finished > verify_data[12]= > 17 f3 73 52 66 96 4a 87 ba 6d 42 f5 > > 7 10 0.5550 (0.1046) C>SV3.1(406) application_data > --------------------------------------------------------------- > CANCEL sip:9999 at XXXX.dyndns.org;user=phone SIP/2.0 > Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport > From: "Matthias" ;tag=rllbt2d9pz > To: > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 2 CANCEL > Max-Forwards: 70 > Reason: SIP;cause=487;text="Request terminated by user" > Content-Length: 0 > > > > --------------------------------------------------------------- > 7 11 0.5559 (0.0008) S>CV3.1(163) application_data > --------------------------------------------------------------- > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/TLS > 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=78.53.13.206 > --------------------------------------------------------------- > 7 12 0.5560 (0.0000) S>CV3.1(84) application_data > --------------------------------------------------------------- > From: "Matthias" ;tag=rllbt2d9pz > --------------------------------------------------------------- > 7 13 0.5560 (0.0000) S>CV3.1(85) application_data > --------------------------------------------------------------- > To: ;tag=XUQm659jUQQvB > --------------------------------------------------------------- > 7 14 0.5560 (0.0000) S>CV3.1(72) application_data > --------------------------------------------------------------- > Call-ID: 5379263cf7de-1vkl60xxknr0 > CSeq: 2 CANCEL > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fraserredmond at gmail.com Sat Aug 14 14:32:47 2010 From: fraserredmond at gmail.com (Fraser Redmond) Date: Sat, 14 Aug 2010 22:32:47 +0100 Subject: [Freeswitch-users] Can you detect if a gateway exists from the dialplan? Message-ID: I want to have different gateways defined on dev and prod servers, but with the same dialplan code used between them. Is there an API function I can put in a condition to detect if a gateway exists? (I had a look through google and the wiki, and the closest looking thing was domain_exists.) Or am I meant to just try bridging to it, and dealing with a failure? Cheers, Fraser -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/89dfeb63/attachment.html From jaybinks at gmail.com Sat Aug 14 14:40:13 2010 From: jaybinks at gmail.com (jay binks) Date: Sun, 15 Aug 2010 07:40:13 +1000 Subject: [Freeswitch-users] Can you detect if a gateway exists from the dialplan? In-Reply-To: References: Message-ID: <6895625413001741156@unknownmsgid> Can a gateway set a channel variable ?? If so you can have the gateways define a var like "gateway-available-MYGATEWAY" then test for THAT in your dial plan .... Jay On 15/08/2010, at 7:32 AM, Fraser Redmond wrote: > I want to have different gateways defined on dev and prod servers, but with the same dialplan code used between them. > > Is there an API function I can put in a condition to detect if a gateway exists? (I had a look through google and the wiki, and the closest looking thing was domain_exists.) > > Or am I meant to just try bridging to it, and dealing with a failure? > > Cheers, > Fraser > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prometheus001 at gmx.net Sat Aug 14 14:47:58 2010 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 14 Aug 2010 23:47:58 +0200 Subject: [Freeswitch-users] valet park and event_socket Message-ID: <4C670F0E.6010204@gmx.net> Hello, before I try to implement and test this, I simply ask some question beforehand (see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park): - is there an event, which I can subscribe to via event socket, when a new parking lot is created? - is there an event, which I can subscribe to via event socket, when a used parking lot is emptied? - can a lot number be alphanumeric, e.g. peter_8001 ? Best regards Peter From fraserredmond at gmail.com Sat Aug 14 15:34:54 2010 From: fraserredmond at gmail.com (Fraser Redmond) Date: Sat, 14 Aug 2010 23:34:54 +0100 Subject: [Freeswitch-users] Can you detect if a gateway exists from the dialplan? In-Reply-To: <6895625413001741156@unknownmsgid> References: <6895625413001741156@unknownmsgid> Message-ID: Thanks Jay, I had a similar thought - I already set a global var in /conf/vars.xml so I'm checking that. Ideally I'd like to check if the gateway is registered too, though I guess that's different than whether it exists or not. Cheers, Fraser On Sat, Aug 14, 2010 at 10:40 PM, jay binks wrote: > Can a gateway set a channel variable ?? > > If so you can have the gateways define a var like > "gateway-available-MYGATEWAY" then test for THAT in your dial plan > .... > > > Jay > > > > On 15/08/2010, at 7:32 AM, Fraser Redmond wrote: > > > I want to have different gateways defined on dev and prod servers, but > with the same dialplan code used between them. > > > > Is there an API function I can put in a condition to detect if a gateway > exists? (I had a look through google and the wiki, and the closest looking > thing was domain_exists.) > > > > Or am I meant to just try bridging to it, and dealing with a failure? > > > > Cheers, > > Fraser > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/bc979b5e/attachment.html From steveayre at gmail.com Sat Aug 14 15:40:10 2010 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 14 Aug 2010 23:40:10 +0100 Subject: [Freeswitch-users] Corrupted name for profile internal In-Reply-To: References: Message-ID: This sounds like a bug. What version of freeswitch are you running? If you aren't on the latest git, upgrade and see if you can reproduce the problem. It may have already been fixed. If you are still able to reproduce the problem, open a Jira ticket so the bug can be tracked and try to identify what causes it. Likely one of the softphones is something peculiar which is triggering a bug that hadn't been reported yet, if you can find a way to quickly reproduce the problem then a packet trace might show what is causing it. Steve on iPhone On 14 Aug 2010, at 20:01, Fraser Redmond wrote: > I've had something strange happening on one of my installations of FreeSwitch. > > After it's been running for a while (couple of days, on fairly low usage) my softphone no longer connects. In the console/log the failure has some unicode chars in the profile name, where I assume it should have 'internal' > > Doing a "sofia profile internal restart reloadxml" doesn't resolve it, but doing a full stop & start of freeswitch does fix it. > > Any ideas? > > 2010-08-14 19:04:32.235578 [WARNING] sofia_reg.c:2086 SIP auth failure (REGISTER) due to user-agent-filter. Filter "nat.auto" User-Agent "(SIPphone here)" > 2010-08-14 19:04:32.235578 [WARNING] sofia_reg.c:1030 SIP auth failure (REGISTER) on sofia profile '??' for [username at 192.168.1.12] from ip 192.168.1.80 > > Cheers, > Fraser > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/a33b3649/attachment.html From steveayre at gmail.com Sat Aug 14 15:43:11 2010 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 14 Aug 2010 23:43:11 +0100 Subject: [Freeswitch-users] use of multiple templates and output files with mod_cdr_csv In-Reply-To: References: <1281605124.28815.239.camel@luna.tc.commsmundi.com> <1281688074.16764.25.camel@luna.tc.commsmundi.com> Message-ID: Indeed, there is no way to load a module multiple times. Modules register various api and app commands. Loading a module twice would register the same command twice, then how does fs know which module you mean? I would agree with Hassan - mod_xml_cdr is far more versatile, as the web server can read the account variable and use that to determine what to do with the cdr. Steve on iPhone On 14 Aug 2010, at 21:42, Nyamul Hassan wrote: > I don't think you can load the same module twice. > > I would suggest you take a look at mod_xml_cdr. It dumps a whole lot of information, which can be very useful for debugging as well. You're already using an external program to parse the CSV files. Just need it to parse the XML output, and you've got access to a whole lot more! > > Regards > HASSAN > > > > On Fri, Aug 13, 2010 at 14:27, Fran?ois Delawarde wrote: > Thanks for the answers, > > > > Look at the accountcode channel variable: > > > http://wiki.freeswitch.org/wiki/Mod_cdr_csv#accountcode > > I initially wanted to use the accountcode variable as an account code > for billing and not for template selection, but I guess I can use it for > template and create another variable (ex: realaccountcode) for the > actual account code. > > Only thing would be that I'd be limited to two templates. Say I want to > log to CSV and a couple of database tables with different fields, well > I'll have to do it "manually". > > > > what I would do is to log all the fields you need to Master.csv, and > > then do selective imports of the fields you require depending on the > > usage. > > mysqlimport (LOAD DATA INFILE) allows that if you put a dummy variable > > (@dummy) instead of a column name. > > Nice idea thanks, I could also have my own CDR script listening for > hangup events, or to log everything to say an sqlite db and use a cron > script to create other logs out of it, but the LOAD DATA INFILE method > seems quite cool. > > Last question: Is there any way to load this module more than one time > with different base directories to make it simpler? > > Thanks again, > Fran?ois. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100814/bea2d6ed/attachment-0001.html From brian at microcomaustralia.com.au Sat Aug 14 17:13:36 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Sun, 15 Aug 2010 10:13:36 +1000 Subject: [Freeswitch-users] freetdm dialtone In-Reply-To: References: Message-ID: Yes, also works for me. On 10 August 2010 11:44, Moises Silva wrote: > cool, do not hesitate in reporting freetdm issues here or at > jira.freeswitch.org (the latter is preferred and you have better chances of > having it fixed if you assign the issue to me right away). I am never sure which project to assign the issue to - does it matter if I get it wrong? -- Brian May From steveayre at gmail.com Sun Aug 15 00:17:59 2010 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 15 Aug 2010 08:17:59 +0100 Subject: [Freeswitch-users] Can you detect if a gateway exists from the dialplan? In-Reply-To: References: <6895625413001741156@unknownmsgid> Message-ID: <2F228271-44C6-4D9C-9D71-C3425DFA588D@gmail.com> The Sofia api command will show you the status of a gateway and if it exists. Perhaps you can use that in some way? Sofia status gateway Steve on iPhone On 14 Aug 2010, at 23:34, Fraser Redmond wrote: > Thanks Jay, I had a similar thought - I already set a global var in /conf/vars.xml so I'm checking that. > > Ideally I'd like to check if the gateway is registered too, though I guess that's different than whether it exists or not. > > Cheers, > Fraser > > > > > On Sat, Aug 14, 2010 at 10:40 PM, jay binks wrote: > Can a gateway set a channel variable ?? > > If so you can have the gateways define a var like > "gateway-available-MYGATEWAY" then test for THAT in your dial plan > .... > > > Jay > > > > On 15/08/2010, at 7:32 AM, Fraser Redmond wrote: > > > I want to have different gateways defined on dev and prod servers, but with the same dialplan code used between them. > > > > Is there an API function I can put in a condition to detect if a gateway exists? (I had a look through google and the wiki, and the closest looking thing was domain_exists.) > > > > Or am I meant to just try bridging to it, and dealing with a failure? > > > > Cheers, > > Fraser > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100815/900b46b6/attachment.html From petedao at gmail.com Sun Aug 15 03:03:27 2010 From: petedao at gmail.com (Pete Kay) Date: Sun, 15 Aug 2010 18:03:27 +0800 Subject: [Freeswitch-users] re-invite problem In-Reply-To: References: <201008122229.47612.sos@sokhapkin.dyndns.org> Message-ID: Hi, Is this a bug that should be filed in jira? On Fri, Aug 13, 2010 at 11:28 AM, Pete Kay wrote: > Hi, > > I have tried setting 3pcc to true and proxy but the result is the same. > > P > > On Fri, Aug 13, 2010 at 10:29 AM, Sergey Okhapkin > wrote: >> Looks like you have no 3rd party call control enabled in FS settings... >> >> On Thursday 12 August 2010, Pete Kay wrote: >>> Hi, >>> >>> Yes, here is the packet trace for the re-invite... >>> >>> recv 513 bytes from udp/[192.168.1.114]:5060 at 07:37:55.968109: >>> ? ?------------------------------------------------------------------------ >>> ? ?INVITE sip:sipp at 192.168.1.114:5070 SIP/2.0 >>> ? ?Record-Route: >>> ? ?Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 >>> ? ?Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 >>> ? ?From: sipp ;tag=1 >>> ? ?To: ;tag=4tj1e3m89ZSBj >>> ? ?Contact: sip:1899 at 192.168.1.3:9871 >>> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >>> ? ?CSeq: 1 INVITE >>> ? ?Session-Expires: 120;refresher=uac >>> ? ?Min-SE: 90 >>> ? ?Supported: 100rel, timer >>> ? ?Max-Forwards: 69 >>> ? ?Content-Length: 0 >>> >>> ? ?------------------------------------------------------------------------ >>> tport_deliver(0x7f7f50004670): msg 0x7f7f502518d0 (513 bytes) from >>> udp/192.168.1.114:5070/sip next=(nil) >>> nta: received INVITE sip:sipp at 192.168.1.114:5070 SIP/2.0 (CSeq 1) >>> nta: canonizing sip:sipp at 192.168.1.114:5070 with contact >>> nta: INVITE (1) going to existing leg >>> nta: timer shortened to 200 ms >>> nua: nua_stack_process_request: entering >>> soa_init_offer_answer(static::0x7f7f501de8e0) called >>> tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 >>> tport_resolve addrinfo = 192.168.1.114:5060 >>> tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 >>> tport_vsend(0x7f7f50004670): 387 bytes of 387 to udp/192.168.1.114:5060 >>> tport_vsend returned 387 >>> send 387 bytes to udp/[192.168.1.114]:5060 at 07:37:55.968290: >>> ? ?------------------------------------------------------------------------ >>> ? ?SIP/2.0 100 Trying >>> ? ?Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 >>> ? ?Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 >>> ? ?Record-Route: >>> ? ?From: sipp ;tag=1 >>> ? ?To: ;tag=4tj1e3m89ZSBj >>> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >>> ? ?CSeq: 1 INVITE >>> ? ?User-Agent: FreeSWITCH >>> ? ?Content-Length: 0 >>> >>> ? ?------------------------------------------------------------------------ >>> nta: sent 100 Trying for INVITE (1) >>> nua(0x7f7e3408b0a0): event i_invite 100 Trying >>> nua: nua_application_event: entering >>> nua: nua_handle_magic: entering >>> nua(0x7f7e3408b0a0): ready call updated: received >>> nua(0x7f7e3408b0a0): event i_state 100 Trying >>> nua: nua_application_event: entering >>> 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel >>> sofia/external/1899 at 192.168.1.3:9871 entering state [received][100] >>> nua: nua_respond: entering >>> nua(0x7f7e3408b0a0): sent signal r_respond >>> nua: nua_handle_magic: entering >>> nua(0x7f7e3408b0a0): recv signal r_respond 200 OK >>> nua: nua_stack_set_params: entering >>> soa_set_params(static::0x7f7f501de8e0, ...) called >>> nua: nua_invite_server_respond: entering >>> soa_generate_offer(static::0x7f7f501de8e0, 0) called >>> soa_static_offer_answer_action(0x7f7f501de8e0, soa_generate_offer): called >>> soa_sdp_mode_set(0x7f7f50226fe0, (nil), ""): called >>> soa_get_local_sdp(static::0x7f7f501de8e0, [(nil)], [0x7f7f60520d30], >>> [0x7f7f60520d3c]) called >>> tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 >>> tport_resolve addrinfo = 192.168.1.114:5060 >>> tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 >>> tport_vsend(0x7f7f50004670): 944 bytes of 944 to udp/192.168.1.114:5060 >>> tport_vsend returned 944 >>> send 944 bytes to udp/[192.168.1.114]:5060 at 07:37:55.968539: >>> ? ?------------------------------------------------------------------------ >>> ? ?SIP/2.0 200 OK >>> ? ?Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.0 >>> ? ?Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-5 >>> ? ?Record-Route: >>> ? ?From: sipp ;tag=1 >>> ? ?To: ;tag=4tj1e3m89ZSBj >>> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >>> ? ?CSeq: 1 INVITE >>> ? ?Contact: >>> ? ?User-Agent: FreeSWITCH >>> ? ?Accept: application/sdp >>> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >>> ? ?Require: timer >>> ? ?Supported: timer, precondition, path, replaces >>> ? ?Session-Expires: 120;refresher=uac >>> ? ?Min-SE: 120 >>> ? ?Content-Type: application/sdp >>> ? ?Content-Disposition: session >>> ? ?Content-Length: 213 >>> >>> ? ?v=0 >>> ? ?o=user1 53655765 2353687637 IN IP4 192.168.1.114 >>> ? ?s=- >>> ? ?c=IN IP4 192.168.1.114 >>> ? ?t=0 0 >>> ? ?m=audio 27938 RTP/AVP 0 18 101 >>> ? ?a=rtpmap:0 PCMU/8000 >>> ? ?a=rtpmap:18 G729/8000 >>> ? ?a=rtpmap:101 telephone-event/8000 >>> ? ?a=ptime:20 >>> ? ?------------------------------------------------------------------------ >>> nta: sent 200 OK for INVITE (1) >>> nua(0x7f7e3408b0a0): ready call updated: completed sent offer >>> soa_get_local_sdp(static::0x7f7f501de8e0, [0x7f7f60520e08], >>> [0x7f7f60520e00], [(nil)]) called >>> nua(0x7f7e3408b0a0): event i_state 200 OK >>> nua: nua_application_event: entering >>> 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel >>> sofia/external/1899 at 192.168.1.3:9871 entering state [completed][200] >>> nua: nua_handle_magic: entering >>> tport_wakeup_pri(0x7f7f50004670): events IN >>> tport_recv_event(0x7f7f50004670) >>> tport_recv_iovec(0x7f7f50004670) msg 0x7f7f5022be60 from >>> (udp/192.168.1.114:5070) has 525 bytes, veclen = 1 >>> recv 525 bytes from udp/[192.168.1.114]:5060 at 07:37:55.968843: >>> ? ?------------------------------------------------------------------------ >>> ? ?ACK sip:sipp at 192.168.1.114:5070 SIP/2.0 >>> ? ?Via: SIP/2.0/UDP 192.168.1.114;branch=z9hG4bKe076.4f957502.2 >>> ? ?Via: SIP/2.0/UDP 192.168.1.3:9871;branch=z9hG4bK-9171-1-8 >>> ? ?From: sipp ;tag=1 >>> ? ?To: ;tag=4tj1e3m89ZSBj >>> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >>> ? ?CSeq: 1 ACK >>> ? ?Max-Forwards: 69 >>> ? ?Content-Type: application/sdp >>> ? ?Content-Length: ? 133 >>> >>> ? ?v=0 >>> ? ?o=user1 53655765 2353687637 IN IP4 192.168.1.3 >>> ? ?s=- >>> ? ?c=IN IP4 192.168.1.3 >>> ? ?t=0 0 >>> ? ?m=audio 6000 RTP/AVP 0 >>> ? ?a=rtpmap:0 PCMU/8000 >>> ? ?------------------------------------------------------------------------ >>> tport_deliver(0x7f7f50004670): msg 0x7f7f5022be60 (525 bytes) from >>> udp/192.168.1.114:5070/sip next=(nil) >>> nta: received ACK sip:sipp at 192.168.1.114:5070 SIP/2.0 (CSeq 1) >>> nta: ACK (1) is going to INVITE (1) >>> nua: process_ack_or_cancel: entering >>> soa_set_remote_sdp(static::0x7f7f501de8e0, (nil), 0x7f7f501eb298, 133) >>> ?called soa_process_answer(static::0x7f7f501de8e0) called >>> soa_static_offer_answer_action(0x7f7f501de8e0, soa_process_answer): called >>> soa_sdp_mode_set(0x7f7f50226fe0, 0x7f7f50245750, ""): called >>> soa_activate(static::0x7f7f501de8e0, (nil)) called >>> soa_clear_remote_sdp(static::0x7f7f501de8e0) called >>> nua(0x7f7e3408b0a0): event i_ack 200 OK >>> nua: nua_application_event: entering >>> nua: nua_handle_magic: entering >>> nua(0x7f7e3408b0a0): ready call updated: ready received answer >>> soa_get_remote_sdp(static::0x7f7f501de8e0, [0x7f7f60520968], >>> [0x7f7f60520960], [(nil)]) called >>> soa_get_params(static::0x7f7f501de8e0, ...) called >>> nua(0x7f7e3408b0a0): event i_state 200 OK >>> nua: nua_application_event: entering >>> 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3289 Channel >>> sofia/external/1899 at 192.168.1.3:9871 entering state [ready][200] >>> 2010-08-10 00:37:55.968013 [DEBUG] sofia.c:3296 Remote SDP: >>> v=0 >>> o=user1 53655765 2353687637 IN IP4 192.168.1.3 >>> s=- >>> c=IN IP4 192.168.1.3 >>> t=0 0 >>> m=audio 6000 RTP/AVP 0 >>> a=rtpmap:0 PCMU/8000 >>> >>> nua: nua_respond: entering >>> nua(0x7f7e3408b0a0): sent signal r_respond >>> 2010-08-10 00:37:55.968013 [NOTICE] sofia.c:3712 Hangup >>> sofia/external/1899 at 192.168.1.3:9871 [CS_HIBERNATE] >>> [INCOMPATIBLE_DESTINATION] >>> 2010-08-10 00:37:55.969068 [DEBUG] switch_channel.c:1683 Send signal >>> sofia/external/1899 at 192.168.1.3:9871 [KILL] >>> 2010-08-10 00:37:55.969068 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/1899 at 192.168.1.3:9871 [BREAK] >>> nua: nua_handle_magic: entering >>> nua(0x7f7e3408b0a0): event i_active 200 Call active >>> nua: nua_application_event: entering >>> nua(): refresh session after 88 seconds (in [88..88]) >>> nua(0x7f7e3408b0a0): recv signal r_respond 488 Not Acceptable Here >>> nua(0x7f7e3408b0a0): event i_error 500 Responding to a Non-Existing Request >>> nua: nua_application_event: entering >>> 2010-08-10 00:37:55.969068 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/1899 at 192.168.1.3:9871) Running State Change CS_HANGUP >>> 2010-08-10 00:37:55.969068 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/external/1899 at 192.168.1.3:9871) State HANGUP >>> 2010-08-10 00:37:55.969068 [DEBUG] mod_sofia.c:338 Channel >>> sofia/external/1899 at 192.168.1.3:9871 hanging up, cause: >>> INCOMPATIBLE_DESTINATION >>> 2010-08-10 00:37:55.969068 [DEBUG] mod_sofia.c:396 Sending BYE to >>> sofia/external/1899 at 192.168.1.3:9871 >>> nua: nua_bye: entering >>> nua(0x7f7e3408b0a0): sent signal r_bye >>> nua(0x7f7e3408b0a0): recv signal r_bye >>> nua: nua_stack_set_params: entering >>> soa_set_params(static::0x7f7f501de8e0, ...) called >>> soa_terminate(static::0x7f7f501de8e0) called >>> 2010-08-10 00:37:55.969068 [NOTICE] switch_ivr_bridge.c:710 Hangup >>> sofia/external/sipp at 192.168.1.3:8970 [CS_HIBERNATE] >>> [INCOMPATIBLE_DESTINATION] >>> soa_init_offer_answer(static::0x7f7f501de8e0) called >>> nta: selecting scheme sip >>> tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 >>> tport_resolve addrinfo = 192.168.1.114:5060 >>> tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 >>> 2010-08-10 00:37:55.970127 [DEBUG] switch_channel.c:1683 Send signal >>> sofia/external/sipp at 192.168.1.3:8970 [KILL] >>> 2010-08-10 00:37:55.970127 [DEBUG] mod_limit.c:195 >>> originate_disposition=[SUCCESS] >>> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/sipp at 192.168.1.3:8970 [BREAK] >>> tport_vsend(0x7f7f50004670): 619 bytes of 619 to udp/192.168.1.114:5060 >>> tport_vsend returned 619 >>> send 619 bytes to udp/[192.168.1.114]:5060 at 07:37:55.970197: >>> ? ?------------------------------------------------------------------------ >>> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:46 >>> sofia/external/1899 at 192.168.1.3:9871 Standard HANGUP, cause: >>> INCOMPATIBLE_DESTINATION >>> ? ?BYE sip:1899 at 192.168.1.3:9871 SIP/2.0 >>> ? ?Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKeFg2t9SeUmHSN >>> ? ?Route: >>> ? ?Max-Forwards: 70 >>> ? ?From: "sipp" ;tag=4tj1e3m89ZSBj >>> ? ?To: ;tag=1 >>> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >>> ? ?CSeq: 376874 BYE >>> ? ?Contact: >>> ? ?User-Agent: FreeSWITCH >>> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >>> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/sipp at 192.168.1.3:8970) Running State Change CS_HANGUP >>> ? ?Supported: timer, precondition, path, replaces >>> ? ?Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>> ? ?Content-Length: 02010-08-10 00:37:55.970127 [DEBUG] >>> switch_core_state_machine.c:434 (sofia/external/1899 at 192.168.1.3:9871) >>> State HANGUP going to sleep >>> >>> >>> ? ?------------------------------------------------------------------------ >>> nta: sent BYE (376874) to UDP/192.168.1.114:5060 >>> tport_pend(0x7f7f50004670): pending 0x7f7f50258650 for >>> udp/192.168.1.114:5070 (already 0) >>> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:461 >>> Hangup Command decre_call_stat(44 60 43 59 14 429178 0): >>> >>> 2010-08-10 00:37:55.970127 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/external/sipp at 192.168.1.3:8970) State HANGUP >>> 2010-08-10 00:37:55.970127 [DEBUG] mod_sofia.c:338 Channel >>> sofia/external/sipp at 192.168.1.3:8970 hanging up, cause: >>> INCOMPATIBLE_DESTINATION >>> 2010-08-10 00:37:55.970127 [DEBUG] mod_sofia.c:396 Sending BYE to >>> sofia/external/sipp at 192.168.1.3:8970 >>> nua: nua_bye: entering >>> nua(0x7f7f5022abd0): sent signal r_bye >>> nua(0x7f7f5022abd0): recv signal r_bye >>> nua: nua_stack_set_params: entering >>> soa_set_params(static::0x7f7f50202440, ...) called >>> soa_terminate(static::0x7f7f50202440) called >>> soa_init_offer_answer(static::0x7f7f50202440) called >>> nta: selecting scheme sip >>> tport_tsend(0x7f7f50004670) tpn = */192.168.1.114:5060 >>> tport_resolve addrinfo = 192.168.1.114:5060 >>> tport_by_addrinfo(0x7f7f50004670): not found by name */192.168.1.114:5060 >>> tport_vsend(0x7f7f50004670): 637 bytes of 637 to udp/192.168.1.114:5060 >>> tport_vsend returned 637 >>> send 637 bytes to udp/[192.168.1.114]:5060 at 07:37:55.970869: >>> ? ?------------------------------------------------------------------------ >>> ? ?BYE sip:sipp at 192.168.1.3:8970 SIP/2.0 >>> ? ?Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKFr9tv4ajrX7BH >>> ? ?Route: >>> ? ?Max-Forwards: 70 >>> ? ?From: sut ;tag=3HS8c834cQ3rp >>> ? ?To: sipp ;tag=1 >>> ? ?Call-ID: 1-9173 at 192.168.1.3 >>> ? ?CSeq: 376873 BYE >>> ? ?Contact: >>> ? ?User-Agent: FreeSWITCH >>> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >>> ? ?Supported: timer, precondition, path, replaces >>> ? ?Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>> ? ?Content-Length: 0 >>> >>> ? ?------------------------------------------------------------------------ >>> nta: sent BYE (376873) to */192.168.1.114:5060 >>> tport_pend(0x7f7f50004670): pending 0x7f7f50254a40 for >>> udp/192.168.1.114:5070 (already 1) >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:46 >>> sofia/external/sipp at 192.168.1.3:8970 Standard HANGUP, cause: >>> INCOMPATIBLE_DESTINATION >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/external/sipp at 192.168.1.3:8970) State HANGUP going to sleep >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:476 >>> (sofia/external/sipp at 192.168.1.3:8970) State Change CS_HANGUP -> >>> CS_REPORTING >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/sipp at 192.168.1.3:8970 [BREAK] >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/sipp at 192.168.1.3:8970) Running State Change >>> CS_REPORTING >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 >>> (sofia/external/sipp at 192.168.1.3:8970) State REPORTING >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:53 >>> sofia/external/sipp at 192.168.1.3:8970 Standard REPORTING, cause: >>> INCOMPATIBLE_DESTINATION >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 >>> (sofia/external/sipp at 192.168.1.3:8970) State REPORTING going to sleep >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:411 >>> (sofia/external/sipp at 192.168.1.3:8970) State Change CS_REPORTING -> >>> CS_DESTROY >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:1068 Session >>> 17184 (sofia/external/sipp at 192.168.1.3:8970) Locked, Waiting on >>> external entities >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:476 >>> (sofia/external/1899 at 192.168.1.3:9871) State Change CS_HANGUP -> >>> CS_REPORTING >>> 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1086 Session >>> 17184 (sofia/external/sipp at 192.168.1.3:8970) Ended >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/external/1899 at 192.168.1.3:9871 [BREAK] >>> 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1088 Close >>> Channel sofia/external/sipp at 192.168.1.3:8970 [CS_DESTROY] >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:398 >>> (sofia/external/1899 at 192.168.1.3:9871) Running State Change >>> CS_REPORTING >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 >>> (sofia/external/1899 at 192.168.1.3:9871) State REPORTING >>> 2010-08-10 00:37:55.971183 [DEBUG] mod_cdr_bdb.c:139 Is number 1899 >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 >>> (sofia/external/sipp at 192.168.1.3:8970) State DESTROY >>> 2010-08-10 00:37:55.971183 [DEBUG] mod_sofia.c:255 >>> sofia/external/sipp at 192.168.1.3:8970 SOFIA DESTROY >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:60 >>> sofia/external/sipp at 192.168.1.3:8970 Standard DESTROY >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 >>> (sofia/external/sipp at 192.168.1.3:8970) State DESTROY going to sleep >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:53 >>> sofia/external/1899 at 192.168.1.3:9871 Standard REPORTING, cause: >>> INCOMPATIBLE_DESTINATION >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:612 >>> (sofia/external/1899 at 192.168.1.3:9871) State REPORTING going to sleep >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:411 >>> (sofia/external/1899 at 192.168.1.3:9871) State Change CS_REPORTING -> >>> CS_DESTROY >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_session.c:1068 Session >>> 17185 (sofia/external/1899 at 192.168.1.3:9871) Locked, Waiting on >>> external entities >>> 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1086 Session >>> 17185 (sofia/external/1899 at 192.168.1.3:9871) Ended >>> 2010-08-10 00:37:55.971183 [NOTICE] switch_core_session.c:1088 Close >>> Channel sofia/external/1899 at 192.168.1.3:9871 [CS_DESTROY] >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 >>> (sofia/external/1899 at 192.168.1.3:9871) State DESTROY >>> 2010-08-10 00:37:55.971183 [DEBUG] mod_sofia.c:255 >>> sofia/external/1899 at 192.168.1.3:9871 SOFIA DESTROY >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:60 >>> sofia/external/1899 at 192.168.1.3:9871 Standard DESTROY >>> 2010-08-10 00:37:55.971183 [DEBUG] switch_core_state_machine.c:564 >>> (sofia/external/1899 at 192.168.1.3:9871) State DESTROY going to sleep >>> nta: timer set next to 301 ms >>> nta: timer E fired, retransmit BYE (376874) >>> tport_release(0x7f7f50004670): 0x7f7f50258650 by 0x7f7f50275640 with (nil) >>> tport_tsend(0x7f7f50004670) tpn = UDP/192.168.1.114:5060 >>> tport_resolve addrinfo = 192.168.1.114:5060 >>> tport_by_addrinfo(0x7f7f50004670): not found by name UDP/192.168.1.114:5060 >>> tport_vsend(0x7f7f50004670): 619 bytes of 619 to udp/192.168.1.114:5060 >>> tport_vsend returned 619 >>> send 619 bytes to udp/[192.168.1.114]:5060 at 07:37:56.469550: >>> ? ?------------------------------------------------------------------------ >>> ? ?BYE sip:1899 at 192.168.1.3:9871 SIP/2.0 >>> ? ?Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKeFg2t9SeUmHSN >>> ? ?Route: >>> ? ?Max-Forwards: 70 >>> ? ?From: "sipp" ;tag=4tj1e3m89ZSBj >>> ? ?To: ;tag=1 >>> ? ?Call-ID: 07b4d02e-1ef5-122e-dd94-18a905453cf0 >>> ? ?CSeq: 376874 BYE >>> ? ?Contact: >>> ? ?User-Agent: FreeSWITCH >>> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >>> ? ?Supported: timer, precondition, path, replaces >>> ? ?Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>> ? ?Content-Length: 0 >>> >>> ? ?------------------------------------------------------------------------ >>> nta: resent BYE (376874) to UDP/192.168.1.114:5060 >>> tport_pend(0x7f7f50004670): pending 0x7f7f50258650 for >>> udp/192.168.1.114:5070 (already 1) >>> nta_outgoing_timer: 1/2 resent, 0/3 tout, 0/1 term, 0/4 free >>> nta: timer set next to 1 ms >>> nta: timer E fired, retransmit BYE (376873) >>> tport_release(0x7f7f50004670): 0x7f7f50254a40 by 0x7f7f50247670 with (nil) >>> tport_tsend(0x7f7f50004670) tpn = */192.168.1.114:5060 >>> tport_resolve addrinfo = 192.168.1.114:5060 >>> tport_by_addrinfo(0x7f7f50004670): not found by name */192.168.1.114:5060 >>> tport_vsend(0x7f7f50004670): 637 bytes of 637 to udp/192.168.1.114:5060 >>> tport_vsend returned 637 >>> send 637 bytes to udp/[192.168.1.114]:5060 at 07:37:56.470547: >>> ? ?------------------------------------------------------------------------ >>> ? ?BYE sip:sipp at 192.168.1.3:8970 SIP/2.0 >>> ? ?Via: SIP/2.0/UDP 192.168.1.114:5070;rport;branch=z9hG4bKFr9tv4ajrX7BH >>> ? ?Route: >>> ? ?Max-Forwards: 70 >>> ? ?From: sut ;tag=3HS8c834cQ3rp >>> ? ?To: sipp ;tag=1 >>> ? ?Call-ID: 1-9173 at 192.168.1.3 >>> ? ?CSeq: 376873 BYE >>> ? ?Contact: >>> ? ?User-Agent: FreeSWITCH >>> ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >>> ? ?Supported: timer, precondition, path, replaces >>> ? ?Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>> ? ?Content-Length: 0 >>> >>> ? ?------------------------------------------------------------------------ >>> nta: resent BYE (376873) to */192.168.1.114:5060 >>> tport_pend(0x7f7f50004670): pending 0x7f7f50254a40 for >>> udp/192.168.1.114:5070 (already 1) >>> nta_outgoing_timer: 1/2 resent, 0/3 tout, 0/1 term, 0/4 free >>> nta: timer set next to 638 ms >>> >>> On Fri, Aug 13, 2010 at 2:05 AM, Steven Ayre wrote: >>> > Do you have a packet trace available? >>> > >>> > -Steve >>> > >>> > On 12 August 2010 14:25, Pete Kay wrote: >>> >> Hi, >>> >> >>> >> I check tags and callid. ?It is the same dialog. ?Also, the invite is >>> >> accepted and I can see UAC does respond 200 OK . ?The freeswitch sends >>> >> out BYE after ACK. >>> >> >>> >> The problem I am seeing is that the re-invite triggers the dialplan >>> >> execution which based on its logic is responding with a 488 within the >>> >> dialplan using the respond app. ?When freeswitch receives the 488, it >>> >> can't recognize the dialog so it sends out BYE. >>> >> >>> >> Therefore, I think the way to solve this is to configure sofia so that >>> >> the invite won't trigger the execution of dialplan. >>> >> >>> >> Is there anyway to do that? >>> >> >>> >> Thanks, >>> >> P >>> >> >>> >> On Thu, Aug 12, 2010 at 9:00 PM, Brian West wrote: >>> >> > Chances are it is in fact a new dialog have you double checked the >>> >> > to/from tags and callid? ?Also if its without an SDP you would have to >>> >> > enable 3pcc on the profile to accept it possibly. >>> >> > >>> >> > /b >>> >> > >>> >> > On Aug 12, 2010, at 3:11 AM, Pete Kay wrote: >>> >> >> I am running a b2bua with freeswitch. ?It is fine until a Mitel UAS >>> >> >> starts sending INVITE without sdp and ACK with sdp. Freeswitch seems >>> >> >> to treat it as another dialog and sends it to dialplan handling. >>> >> >> >>> >> >> Within the dialplan, how can I recognize request as a re-invite and >>> >> >> possibly ignore it? >>> >> >> >>> >> >> Does anyone know how to resolve this problem? >>> >> > >>> >> > _______________________________________________ >>> >> > FreeSWITCH-users mailing list >>> >> > FreeSWITCH-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> >> >rs http://www.freeswitch.org >>> >> >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From msc at freeswitch.org Sun Aug 15 12:40:05 2010 From: msc at freeswitch.org (Michael Collins) Date: Sun, 15 Aug 2010 12:40:05 -0700 Subject: [Freeswitch-users] valet park and event_socket In-Reply-To: <4C670F0E.6010204@gmx.net> References: <4C670F0E.6010204@gmx.net> Message-ID: On Sat, Aug 14, 2010 at 2:47 PM, Peter P GMX wrote: > Hello, > > before I try to implement and test this, I simply ask some question > beforehand (see > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park): > - is there an event, which I can subscribe to via event socket, when a > new parking lot is created? > - is there an event, which I can subscribe to via event socket, when a > used parking lot is emptied? > I see 3 different events created by mod_valet_parking.c and none of them have to do with a valet being "created" or "emptied". They are: exit, hold, and bridge. If you want to see all of the events fired as callers are put into or taken out of a valet: fs_cli /log 0 /events plain all /filter Event-Subclass valet_parking::info - can a lot number be alphanumeric, e.g. peter_8001 ? > Yes, you can have alphanumerics in the lot name. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100815/e520ae5d/attachment.html From mike at jerris.com Sun Aug 15 14:51:35 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 17:51:35 -0400 Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1281809261831-5423774.post@n2.nabble.com> References: <1278469312148-5263368.post@n2.nabble.com> <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> <1278517888279-5265912.post@n2.nabble.com> <1278543145818-5267704.post@n2.nabble.com> <1281809261831-5423774.post@n2.nabble.com> Message-ID: <640CFC68-974D-4E28-AEE2-E82E20E54E62@jerris.com> Please open a bug on jira on this issue for me with all those details attached. Thanks Mike On Aug 14, 2010, at 2:07 PM, mazilo wrote: > > OpenWRT has created/included the patch for sofia.c file as pointed out above > by > http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5256137.html > Anthony Minessale . > > > Michael Jerris wrote: >> >> I had the same issue with other uclibc builds, we had a patch for uclibc >> in astlinux that fixes this, but I believe it was fixed in newer uclibc as >> well. This was some header bug in uclibc iirc. > > From prayersts at gmail.com Sat Aug 14 08:45:04 2010 From: prayersts at gmail.com (Tae-Sung Shin) Date: Sat, 14 Aug 2010 11:45:04 -0400 Subject: [Freeswitch-users] transfer a call to external phone In-Reply-To: References: Message-ID: <00d101cb3bc7$aaf53b40$00dfb1c0$@com> Guys I am having no problem with transferring a call to internal phones but I got hangup as soon as transferring an incoming (external) call to an external phone. I am mostly using default configuration. Can you guys think of any cause of this problem? Thanks From mike at jerris.com Sun Aug 15 15:10:27 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 18:10:27 -0400 Subject: [Freeswitch-users] setting a context per gateway in a single profile In-Reply-To: <1279270340.11951.11.camel@marces.tc.commsmundi.com> References: <1279270340.11951.11.camel@marces.tc.commsmundi.com> Message-ID: <7F89A34C-E573-4FC3-A365-8D45874FC0FD@jerris.com> Matching inbound calls to a gateway is at best unreliable. The reason is, we don't necessarily know what gateway an inbound call is coming from as many providers do not use the request uri as our registered contact. I am not sure there is any reliable solution to this problem. Mike On Jul 16, 2010, at 4:52 AM, Antonio wrote: > I want to set a incoming context per each gateway that i have register > in one profile only. so in the dialplan i can have rules per gateway, > without having to do transfers from de default context per gw. > > > Is it possible? > > Thanks, > Ant?nio > > the configuration: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Sun Aug 15 15:12:58 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 18:12:58 -0400 Subject: [Freeswitch-users] transfer a call to external phone In-Reply-To: <00d101cb3bc7$aaf53b40$00dfb1c0$@com> References: <00d101cb3bc7$aaf53b40$00dfb1c0$@com> Message-ID: <32ED90F8-AF0A-43E0-A912-BE658125096F@jerris.com> Take a look at debug logs and sip trace, it should show you something about what the issue is. Mike On Aug 14, 2010, at 11:45 AM, Tae-Sung Shin wrote: > Guys > > I am having no problem with transferring a call to internal phones but I got > hangup as soon as transferring an incoming (external) call to an external > phone. I am mostly using default configuration. Can you guys think of any > cause of this problem? From mike at jerris.com Sun Aug 15 15:20:50 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 18:20:50 -0400 Subject: [Freeswitch-users] Determining codec when using inbound-late-negotiation In-Reply-To: References: Message-ID: We never fully parse the sdp in this case, but the full sdp of each side should be stored as channel vars that you could parse if you like to figure out what codec was used. Its necessarily going to be definitive if there are multiple negotiated codecs,. Mike On Jul 16, 2010, at 6:20 PM, Roger Salloum wrote: > Correct, but is there a way to know what codec freeswitch is currently proxing for a given call even if proxy media is on > > > ----- Original Message ----- > From: Brian West > Date: Friday, July 16, 2010 7:27 am > Subject: Re: [Freeswitch-users] Determining codec when using inbound-late-negotiation > To: FreeSWITCH Users Help > > > Cuz you have proxy media on. > > > > /b > > > > On Jul 16, 2010, at 9:17 AM, Roger Salloum wrote: > > > > > When using inbound-late-negotiation, is it possible to > > determine what codec was negotiated when the A leg was answered? > > The read_codec and the write_codec are given the value of PROXY. From mike at jerris.com Sun Aug 15 15:26:04 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 18:26:04 -0400 Subject: [Freeswitch-users] CallCentric keeps changing expire time by +1 second In-Reply-To: <1279471317231-5309210.post@n2.nabble.com> References: <1279471317231-5309210.post@n2.nabble.com> Message-ID: <7F1DB29B-DA8F-4D93-96B8-399CCC5A0484@jerris.com> Looks like we just log it, I expect they always do this, but I have no idea why. Mike On Jul 18, 2010, at 12:41 PM, mazilo wrote: > > I have configured my FreeSWITCH PBX system with a CallCentric IP Freedom > account using this > http://wiki.freeswitch.org/wiki/Provider_Configuration:_Callcentric SIP > profile . It works just fine to place/receive calls to/from other > CallCentric numbers. However, what puzzles me is the log messages file > incidates CallCentric keeps changing the expire time by +1 second as shown > below. Honestly, I have never seen this behavior on my Asterisk PBX system. > Is this a new feature from FreeSWITCH? > > 2010-07-18 10:53:53.041636 [DEBUG] sofia_reg.c:1484 Changing expire time to > 85 by request of proxy sip:callcentric.com > 2010-07-18 10:54:38.038438 [DEBUG] sofia_reg.c:1484 Changing expire time to > 86 by request of proxy sip:callcentric.com From mike at jerris.com Sun Aug 15 15:38:18 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 18:38:18 -0400 Subject: [Freeswitch-users] Error Loading module mod_spidermonkey.so In-Reply-To: <4C442F63.7000608@gmail.com> References: <4C36BDCC.3020309@gmail.com> <2979973A-00C7-4E58-8C96-422F8225B157@freeswitch.org> <4C3AAC3D.3040209@gmail.com> <87B5CFA9-723F-458F-A387-8E30260A71E2@freeswitch.org> <4C442F63.7000608@gmail.com> Message-ID: This should be fixed now in git HEAD. Mike On Jul 19, 2010, at 6:56 AM, Goa wrote: > Hello, Brian. > > Can you send me link to the jira issue, please? > I tried to find but all issues I found was already fixed and closed. > > Thank you. > > 12.07.2010 18:33, Brian West ?????: >> A jira already is open on this issue. FreeBSD is not one of our top three target platforms... the top three are windows, mac and linux... anything else is just bonus. ;) >> From mike at jerris.com Sun Aug 15 15:49:46 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 18:49:46 -0400 Subject: [Freeswitch-users] Multiple Dialog Subscription NOTIFY Messages In-Reply-To: <4496CFEBB8FC4D398E1F1DDF2693524B@greyhawk.tonecommander.com> References: <765ECD85-0F04-45C9-936A-06C1503F881E@freeswitch.org> <4496CFEBB8FC4D398E1F1DDF2693524B@greyhawk.tonecommander.com> Message-ID: <382E60F4-8EC4-4441-867D-DB3A5945D08E@jerris.com> Please open a jira on this issue if you can confirm this still in current git. Mike On Jul 22, 2010, at 1:58 PM, Jerry Richards wrote: > I didn't check the subscription table, but I do have 5 phones subscribing to > "dialog" events of the same extension. It appears that FS is sending NOTIFY > messages for each subscriber to each subscriber, when it should only > generate NOTIFY messages for the extension being monitored. > > Jerry > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian > West > Sent: Wednesday, July 21, 2010 9:03 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Multiple Dialog Subscription NOTIFY Messages > > Because its subscribed to 5 times I suspect.. have you checked the sip > subscription table? > > /b > > On Jul 21, 2010, at 10:57 AM, Jerry Richards wrote: > >> Hello, >> >> If I configure 5 phones to subscribe to "dialog" events of a >> particular extension, and then make a call to that extension, each >> subscribing phone receives multiple NOTIFY messages with dialog state >> "early". I looked into the content of the NOTIFY messages received by >> one of the subscribers. It received 5 pairs of NOTIFY messages where >> each pair contained a different . What is the reason FS would > do this? >> >> Thanks, >> Jerry > > From mike at jerris.com Sun Aug 15 15:52:14 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 18:52:14 -0400 Subject: [Freeswitch-users] enable passthrough of "Privacy: id" header in sip In-Reply-To: <201007221436.42762.sos@sokhapkin.dyndns.org> References: <1a5cbae9-5bd2-4fd1-d1d3-88fae8b6bd15@me.com> <201007221436.42762.sos@sokhapkin.dyndns.org> Message-ID: is there a specification which defines this "uri" value? Mike On Jul 22, 2010, at 2:36 PM, Sergey Okhapkin wrote: > FS doesn't recognize "uri" value of privacy tag, the recognized values are > "yes", "full", "name" and "number". Any other value is interpreted as "off". > See sofia.c, lines around 6660. > > On Thursday 22 July 2010, mike.burlingame wrote: >> Ok so I am still kinda lost in trying to figure this one out here are the >> two headers that I am looking at - basically someone sends an invite to FS >> that ask's FS via RPID to hide the caller id info from downstream gateways >> so the A-LEG is the invite going to FS - FS takes the invite and spits out >> the B-LEG to go downstream HOWEVER FS does not copy the parameters >> correctly as you can see in the initial invite A-LEG requested in the RPID >> to be privacy=uri however on the B-LEG side FS set the RPID to privacy=off >> in turn telling gateways downstream to display the CID info. >> >> A-LEG >> Remote-Party-ID: ;party=calling;screen=yes;privacy=uri >> >> B-LEG >> Remote-Party-ID: "NAME" >> ;party=calling;screen=yes;privacy=off >> >> On Jul 22, 2010, at 09:35 AM, Michael Collins wrote: >> >> >> >> On Wed, Jul 21, 2010 at 4:18 PM, mike.burlingame >> wrote: yeah the variable I need from the A-LEG is privacy=uri the B-LEG of >> FS by default is putting privacy=off - so I would guess the variable that >> needs to be exported would be the privacy= correct? >> >> So the next question I would have is how do I export that that from the >> A-LEG to the B-LEG - I would have though FS would be Data in Data out so >> no need to change the RPID request from the original A-LEG? Is the >> variable simply named "privacy"? If so just use the export app before the >> bridge: From mike at jerris.com Sun Aug 15 15:58:08 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 18:58:08 -0400 Subject: [Freeswitch-users] SIP session timer In-Reply-To: <201007231542.22796.sos@sokhapkin.dyndns.org> References: <201007231526.40276.sos@sokhapkin.dyndns.org> <201007231542.22796.sos@sokhapkin.dyndns.org> Message-ID: <4D094510-C92D-4854-AFBC-E60663602FAB@jerris.com> Still awaiting your reply on that bug. Could you please respond to the questions posted? Mike On Jul 23, 2010, at 3:42 PM, Sergey Okhapkin wrote: > Well, let me provide some background info - I'm looking for a work around for > http://jira.freeswitch.org/browse/MODSOFIA-72 which bothers me a lot. 3 > crashes today. > > Can I enable SST on B-leg only? I.e. > in SIP profile and {sofia_session_timeout=NNN} in bridge command? > > On Friday 23 July 2010, Sergey Okhapkin wrote: >> Is it possible to use SIP UPDATE method to check session activity instead >> of re-INVITE? >> From mike at jerris.com Sun Aug 15 16:23:34 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 19:23:34 -0400 Subject: [Freeswitch-users] Native stacks In-Reply-To: References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C373A40.1020207@ewetel.de> Message-ID: You will need to get Apache to change its license as well. I find this unlikely to happen any time soon. Mike On Jul 27, 2010, at 5:26 PM, Mathieu Parent wrote: > I understand your position. IANAL but I want to go a bit deeper on some points. > > > On Tue, Jul 27, 2010 at 7:56 PM, Anthony Minessale > wrote: >> The GPL is the license that would require change not the MPL. (there already >> is such a change called LGPL btw) > > As Nyamul pointed, MPL "has some complex restrictions that make it > incompatible with the GNU GPL". It means that, according to FSF, you > can't link a module under current MPL 1.1 and another under GPL 2 or > 3. This part may be solved in MPL 2.0. From mike at jerris.com Sun Aug 15 16:24:24 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 19:24:24 -0400 Subject: [Freeswitch-users] Native stacks In-Reply-To: References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C373A40.1020207@ewetel.de> Message-ID: It is still my plan to merge the work From stkn back into freetdm, but time has not permitted as of yet to get this done. Mike On Jul 9, 2010, at 11:18 AM, Anthony Minessale wrote: > I want the native stack to succeed because our goal was to have an open source BSD licensed free ISDN stack so we can put an end to people selling it for ungodly fees. > > The libpri is fine to use if a user assembles it himself but ultimately it's GPL and there is a grey-area license conflict with using with FS. Technically OpenZAP/FreeTDM is BSD and compat with libpri but according to the greedy GPL, when you load it the GPL infects the whole code and makes it also GPL. FS is MPL and is happily compat with OpenZAP/FreeTDM but there is a philosophical debate as to if the BSD lib in the middle that completely abstracts the 2 entities, protects FS from the GPL > > FreeTDM is still OpenZAP, just with another name. Sangoma is only working on their own modules for FreeTDM and the API as a whole still supports everything OpenZAP does. I think stkn will merge his stack back into FreeTDM and it can continue to be developed for those who don't have Sangoma cards. > From mike at jerris.com Sun Aug 15 16:38:00 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 19:38:00 -0400 Subject: [Freeswitch-users] FreeSWITCH -- Router -- WAN -- Router -- OpenSBC -- Several Remote Spa5xx In-Reply-To: <20100729223409.6851F1F6904@c-in3ws--03-04.sv2.lotuslive.com> References: <20100729223409.6851F1F6904@c-in3ws--03-04.sv2.lotuslive.com> Message-ID: I have never seen any nat issues with SPA5xx phones in the field and we have many deployed. Can you nail down why they are failing? Try adjusting nat params in the SPA configs such as keepalive. Mike On Jul 29, 2010, at 6:34 PM, Loy Glenn wrote: > As the Subject outlines, this is the basic setup. Does anyone have any experience w/ FreeSWITCH and any session border controllers? The problem we are having seems to be related to NAT. We have several Cisco Spa5xxG phones which remotely register to a FS that is publicly accessible. The phones are behind NAT router, and begin to fail after more than 4 register. > I am thinking about using the SBC for a registration/media proxy, and am wondering how I would get FS and the SBC to work together. Any ideas? > > -- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100815/62675fb1/attachment.html From mike at jerris.com Sun Aug 15 16:38:59 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 19:38:59 -0400 Subject: [Freeswitch-users] Fedora 10 64bits last git In-Reply-To: <37CE1A44066C4271A5EA744D1915C51B@MOBILEE1705> References: <37CE1A44066C4271A5EA744D1915C51B@MOBILEE1705> Message-ID: Can you open a bug on jira for this if there isn't one already please. Mike On Jul 30, 2010, at 12:42 AM, Madovsky wrote: > success to compile install last git with Fedora 10 64bits. > > only a little load module error : > > 2010-07-30 00:33:45.401990 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so > **/lib64/libnssutil3.so: undefined symbol: PR_GetEnv** > And so what ? the book saiys : > > --without-libcurl for Fedora if you have this error > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100815/afa07a84/attachment.html From mike at jerris.com Sun Aug 15 16:42:15 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 19:42:15 -0400 Subject: [Freeswitch-users] can't make current In-Reply-To: <20100730125729.16ac7735@anubis.defcon1> References: <20100730125729.16ac7735@anubis.defcon1> Message-ID: <5015F33B-B5A7-4E08-86F0-7FB845D88D5B@jerris.com> You could remove some items from modules.conf, but i have never seen this before, whats different for you? What OS/distro etc? On Jul 30, 2010, at 6:57 AM, Jean-Yves F. Barbier wrote: > Hi list, > > make current > > Making clean in src > /bin/bash: line 26: /usr/bin/make: Liste d'arguments trop longue > (args list too long) > make: *** [clean-recursive] Erreur 1 From mike at jerris.com Sun Aug 15 16:45:16 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 19:45:16 -0400 Subject: [Freeswitch-users] Conference and Speex Wideband audio quality problem In-Reply-To: References: Message-ID: Stay tuned for mod_rtmp, coming soon to a freeswitch near you. Mike On Jul 30, 2010, at 5:47 PM, Richard Alam wrote: > OK...figured out what the problem. > > Flash Player by default sends 2 frames per packet with 20ms of audio > (320 samples) per frame. FS expects only a frame every packet. > So setting mic.framesPerPacket = 1; in the flash client worked. > > Here's the complete Actionscript setting in case somebody in the > future wants to work with Flash. > > private function setupMicrophone():void { > mic.setUseEchoSuppression(true); > mic.setLoopBack(false); > mic.setSilenceLevel(0,20000); > mic.codec = SoundCodec.SPEEX; > mic.gain = 60; > mic.framesPerPacket = 1; > mic.rate = 16; // use 8 for Nelly > LogUtil.debug("codec=SPEEX,gain=60,encodeQuality=10,framesPerPacket=2,rate=16"); > } > > Richard From mike at jerris.com Sun Aug 15 16:56:51 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 19:56:51 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008020625.37601.sos@sokhapkin.dyndns.org> References: <201008020625.37601.sos@sokhapkin.dyndns.org> Message-ID: I don't think there is ever a reason to use proxy_media anymore. If anyone can provide an example of a good reason to use it I would be interested to hear. On Aug 2, 2010, at 6:25 AM, Sergey Okhapkin wrote: > FS has 3 ways (not 2!) to handle media: > > 1. bypass_media=true - media path is set to run audio directly between leg a > and b endpoints. absolute_codec_string variable is ignored in this mode. > 2. proxy_media=true - media is proxied by FS, but without any transcoding etc, > frame in - frame out. absolute_codec_string variable is ignored in this mode. > 3. Neither bypass_media, no proxy_media variables are set. FS does full media > handling/transcoding, absolute_codec_string variable is honored. From mike at jerris.com Sun Aug 15 17:00:45 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 20:00:45 -0400 Subject: [Freeswitch-users] ODBC postgresql network and FS In-Reply-To: References: Message-ID: <747867B3-E80B-4453-958D-1755E3EC7B3F@jerris.com> pgpool can do this for you, however, we have found quite a few issues with its codebase. I'll work on pushing our patches up to pgpool next week sometime. Mike On Aug 3, 2010, at 5:17 PM, Madovsky wrote: > Hi Clue guys, > > hope you enjoy at Cluecon ! > Is there any setings in FS to restart it > in case of idle DB connection ? > exemple : if the DB on node1 fails and node2 starts DB in failover with the same shared IP, > FS continues to believe that node1 is yet the right node to contact.... > so all SIP communicatio with DB are stalled > maybe a restart or other would be nice or anything else to unblock the status. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100815/0b45c0dc/attachment-0001.html From sos at sokhapkin.dyndns.org Sun Aug 15 17:00:32 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 15 Aug 2010 20:00:32 -0400 Subject: [Freeswitch-users] Strange early media SDP behavior In-Reply-To: <201008141119.07759.sos@sokhapkin.dyndns.org> References: <201008141119.07759.sos@sokhapkin.dyndns.org> Message-ID: <201008152000.32846.sos@sokhapkin.dyndns.org> Any feedback from FS developers? On Saturday 14 August 2010, Sergey Okhapkin wrote: > Late negotiation is on, bypass_media=true. Leg B receives the following SDP > in SIP 183 message: > > v=0 > o=root 4913 4913 IN IP4 64.21.13.41 > s=session > c=IN IP4 64.21.13.41 > t=0 0 > m=audio 37650 RTP/AVP 0 8 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 3 codecs in SDP offer. Freeswitch send to leg A SIP 183 with the following > SDP: > > v=0 > o=root 4913 4913 IN IP4 64.21.13.41 > s=session > c=IN IP4 64.21.13.41 > t=0 0 > m=audio 37650 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 1 codec only. Why 2 other codecs are removed from SDP? Tested on today's > git, older versions have the same behavior. Is it expected behavior or a > bug? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From 12ukwn at gmail.com Sun Aug 15 17:02:45 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Mon, 16 Aug 2010 02:02:45 +0200 Subject: [Freeswitch-users] can't make current In-Reply-To: <5015F33B-B5A7-4E08-86F0-7FB845D88D5B@jerris.com> References: <20100730125729.16ac7735@anubis.defcon1> <5015F33B-B5A7-4E08-86F0-7FB845D88D5B@jerris.com> Message-ID: <20100816020245.7517a230@anubis.defcon1> Le Sun, 15 Aug 2010 19:42:15 -0400, Michael Jerris a ?crit : > You could remove some items from modules.conf, but i have never seen this Most of them are enabled as it is for tests and I don't wanna recompile 3 times a day especially on a 5 years old Athlon XP2600+. > before, whats different for you? What OS/distro etc? OS: Linux Distro: Debian Branch: sid (unstable) Former builds (~2-3 months ago) never had this PB, but I may have add other modules in the build and these modules may have a bunch of .o/.lo files. Just retested a make current: the same. JY > On Jul 30, 2010, at 6:57 AM, Jean-Yves F. Barbier wrote: > > > Hi list, > > > > make current > > > > Making clean in src > > /bin/bash: line 26: /usr/bin/make: Liste d'arguments trop longue > > (args list too long) > > make: *** [clean-recursive] Erreur 1 -- In Tennessee, it is illegal to shoot any game other than whales from a moving automobile. From sos at sokhapkin.dyndns.org Sun Aug 15 17:07:00 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 15 Aug 2010 20:07:00 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: <201008020625.37601.sos@sokhapkin.dyndns.org> Message-ID: <201008152007.00985.sos@sokhapkin.dyndns.org> To avoid transcoding if both endpoints have common codec, but one of endpoints is behind NAT. On Sunday 15 August 2010, Michael Jerris wrote: > I don't think there is ever a reason to use proxy_media anymore. If anyone > can provide an example of a good reason to use it I would be interested to > hear. > > On Aug 2, 2010, at 6:25 AM, Sergey Okhapkin wrote: > > FS has 3 ways (not 2!) to handle media: > > > > 1. bypass_media=true - media path is set to run audio directly between > > leg a and b endpoints. absolute_codec_string variable is ignored in this > > mode. 2. proxy_media=true - media is proxied by FS, but without any > > transcoding etc, frame in - frame out. absolute_codec_string variable is > > ignored in this mode. 3. Neither bypass_media, no proxy_media variables > > are set. FS does full media handling/transcoding, absolute_codec_string > > variable is honored. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Sun Aug 15 17:26:29 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 20:26:29 -0400 Subject: [Freeswitch-users] can't make current In-Reply-To: <20100816020245.7517a230@anubis.defcon1> References: <20100730125729.16ac7735@anubis.defcon1> <5015F33B-B5A7-4E08-86F0-7FB845D88D5B@jerris.com> <20100816020245.7517a230@anubis.defcon1> Message-ID: <1DED4D92-F36C-4C14-AD03-9D0014F04CBF@jerris.com> If you can provide remote access to the box for me, I can take a look at what is going on. Mike On Aug 15, 2010, at 8:02 PM, Jean-Yves F. Barbier wrote: > Le Sun, 15 Aug 2010 19:42:15 -0400, > Michael Jerris a ?crit : > >> You could remove some items from modules.conf, but i have never seen this > > Most of them are enabled as it is for tests and I don't wanna recompile 3 > times a day especially on a 5 years old Athlon XP2600+. > >> before, whats different for you? What OS/distro etc? > > OS: Linux > Distro: Debian > Branch: sid (unstable) > > Former builds (~2-3 months ago) never had this PB, but I may have add other > modules in the build and these modules may have a bunch of .o/.lo files. > > Just retested a make current: the same. > > JY > >> On Jul 30, 2010, at 6:57 AM, Jean-Yves F. Barbier wrote: >> >>> Hi list, >>> >>> make current >>> >>> Making clean in src >>> /bin/bash: line 26: /usr/bin/make: Liste d'arguments trop longue >>> (args list too long) >>> make: *** [clean-recursive] Erreur 1 > > From mike at jerris.com Sun Aug 15 17:28:07 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 20:28:07 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008152007.00985.sos@sokhapkin.dyndns.org> References: <201008020625.37601.sos@sokhapkin.dyndns.org> <201008152007.00985.sos@sokhapkin.dyndns.org> Message-ID: http://wiki.freeswitch.org/wiki/Codec_negotiation disable-transcoding On Aug 15, 2010, at 8:07 PM, Sergey Okhapkin wrote: > To avoid transcoding if both endpoints have common codec, but one of endpoints > is behind NAT. > > On Sunday 15 August 2010, Michael Jerris wrote: >> I don't think there is ever a reason to use proxy_media anymore. If anyone >> can provide an example of a good reason to use it I would be interested to >> hear. >> >> On Aug 2, 2010, at 6:25 AM, Sergey Okhapkin wrote: >>> FS has 3 ways (not 2!) to handle media: >>> >>> 1. bypass_media=true - media path is set to run audio directly between >>> leg a and b endpoints. absolute_codec_string variable is ignored in this >>> mode. 2. proxy_media=true - media is proxied by FS, but without any >>> transcoding etc, frame in - frame out. absolute_codec_string variable is >>> ignored in this mode. 3. Neither bypass_media, no proxy_media variables >>> are set. FS does full media handling/transcoding, absolute_codec_string >>> variable is honored. From jan.berger at video24.no Sun Aug 15 17:33:20 2010 From: jan.berger at video24.no (Jan Berger) Date: Mon, 16 Aug 2010 02:33:20 +0200 Subject: [Freeswitch-users] Native stacks In-Reply-To: References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C373A40.1020207@ewetel.de> Message-ID: <3189660FE9A44826B81101C17D6A44FC@dell9400> Has he actually done anything? Can I find the code somethere? Jan -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 16. august 2010 01:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Native stacks It is still my plan to merge the work From stkn back into freetdm, but time has not permitted as of yet to get this done. Mike On Jul 9, 2010, at 11:18 AM, Anthony Minessale wrote: > I want the native stack to succeed because our goal was to have an open source BSD licensed free ISDN stack so we can put an end to people selling it for ungodly fees. > > The libpri is fine to use if a user assembles it himself but ultimately it's GPL and there is a grey-area license conflict with using with FS. Technically OpenZAP/FreeTDM is BSD and compat with libpri but according to the greedy GPL, when you load it the GPL infects the whole code and makes it also GPL. FS is MPL and is happily compat with OpenZAP/FreeTDM but there is a philosophical debate as to if the BSD lib in the middle that completely abstracts the 2 entities, protects FS from the GPL > > FreeTDM is still OpenZAP, just with another name. Sangoma is only working on their own modules for FreeTDM and the API as a whole still supports everything OpenZAP does. I think stkn will merge his stack back into FreeTDM and it can continue to be developed for those who don't have Sangoma cards. > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mnhassan at usa.net Sun Aug 15 17:34:37 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 16 Aug 2010 06:34:37 +0600 Subject: [Freeswitch-users] mod_xml_cdr flags b-leg with "a_" Message-ID: Hi, When a call hangs up normally, like either of the user kept the phone down, then there are no issues. However, when the call is hung up through a transfer, like mod_nibblebill does when "low_balance" or "no_balance" is reached, then the b-leg cdr.xml file gets prefixed with "a_". Is this a desired feature? Or a bug? Regards HASSAN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/98e9ac4a/attachment.html From 12ukwn at gmail.com Sun Aug 15 17:40:38 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Mon, 16 Aug 2010 02:40:38 +0200 Subject: [Freeswitch-users] can't make current In-Reply-To: <1DED4D92-F36C-4C14-AD03-9D0014F04CBF@jerris.com> References: <20100730125729.16ac7735@anubis.defcon1> <5015F33B-B5A7-4E08-86F0-7FB845D88D5B@jerris.com> <20100816020245.7517a230@anubis.defcon1> <1DED4D92-F36C-4C14-AD03-9D0014F04CBF@jerris.com> Message-ID: <20100816024038.7930ae8a@anubis.defcon1> Le Sun, 15 Aug 2010 20:26:29 -0400, Michael Jerris a ?crit : > If you can provide remote access to the box for me, I can take a look at > what is going on. > > Mike Nope, I'll try to figure it out myself. Otherwise I'll keep my mod'op as it is: reinstall the source each time instead of upgrading it. JY -- From Nabble at slickdeals.endjunk.com Sun Aug 15 17:42:35 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Sun, 15 Aug 2010 17:42:35 -0700 (PDT) Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <640CFC68-974D-4E28-AEE2-E82E20E54E62@jerris.com> References: <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> <1278517888279-5265912.post@n2.nabble.com> <1278543145818-5267704.post@n2.nabble.com> <1281809261831-5423774.post@n2.nabble.com> <640CFC68-974D-4E28-AEE2-E82E20E54E62@jerris.com> Message-ID: <1281919355652-5426385.post@n2.nabble.com> Hi Mike, I don't think it is necessary because the problem has been fixed by OpenWRT. Michael Jerris wrote: > > Please open a bug on jira on this issue for me with all those details > attached. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5426385.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sos at sokhapkin.dyndns.org Sun Aug 15 17:48:38 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 15 Aug 2010 20:48:38 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: <201008152007.00985.sos@sokhapkin.dyndns.org> Message-ID: <201008152048.38275.sos@sokhapkin.dyndns.org> But what if endpoints have no common codec and disable-transcoding is set? Will FS ignore disable-transcoding parameter? On Sunday 15 August 2010, Michael Jerris wrote: > http://wiki.freeswitch.org/wiki/Codec_negotiation > > disable-transcoding > > On Aug 15, 2010, at 8:07 PM, Sergey Okhapkin wrote: > > To avoid transcoding if both endpoints have common codec, but one of > > endpoints is behind NAT. > > > > On Sunday 15 August 2010, Michael Jerris wrote: > >> I don't think there is ever a reason to use proxy_media anymore. If > >> anyone can provide an example of a good reason to use it I would be > >> interested to hear. > >> > >> On Aug 2, 2010, at 6:25 AM, Sergey Okhapkin wrote: > >>> FS has 3 ways (not 2!) to handle media: > >>> > >>> 1. bypass_media=true - media path is set to run audio directly between > >>> leg a and b endpoints. absolute_codec_string variable is ignored in > >>> this mode. 2. proxy_media=true - media is proxied by FS, but without > >>> any transcoding etc, frame in - frame out. absolute_codec_string > >>> variable is ignored in this mode. 3. Neither bypass_media, no > >>> proxy_media variables are set. FS does full media handling/transcoding, > >>> absolute_codec_string variable is honored. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Sun Aug 15 18:49:24 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 21:49:24 -0400 Subject: [Freeswitch-users] Native stacks In-Reply-To: <3189660FE9A44826B81101C17D6A44FC@dell9400> References: <5195B7332FD141F8950E89C65F3BE491@dell9400> <4C3717E4.602@ewetel.de> <70AA47CA5E7B4209A6ECA97A7F55CA23@dell9400> <4C373A40.1020207@ewetel.de> <3189660FE9A44826B81101C17D6A44FC@dell9400> Message-ID: http://oss.axsentis.de/gitweb/?p=libisdn.git;a=summary http://oss.axsentis.de/gitweb/?p=ftmod_isdn.git;a=summary Mike On Aug 15, 2010, at 8:33 PM, Jan Berger wrote: > Has he actually done anything? Can I find the code somethere? > > Jan > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael > Jerris > Sent: 16. august 2010 01:24 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Native stacks > > It is still my plan to merge the work From stkn back into freetdm, but time > has not permitted as of yet to get this done. > > Mike From mike at jerris.com Sun Aug 15 18:53:00 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 21:53:00 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008152048.38275.sos@sokhapkin.dyndns.org> References: <201008152007.00985.sos@sokhapkin.dyndns.org> <201008152048.38275.sos@sokhapkin.dyndns.org> Message-ID: <40C7FF38-D47F-4190-A0D6-44BA14390845@jerris.com> Proxy media does not handle this case now and the call will fail in the same way. We will try to avoid transcoding by default, so I think what you are suggesting is already the default action without proxy_media. On Aug 15, 2010, at 8:48 PM, Sergey Okhapkin wrote: > But what if endpoints have no common codec and disable-transcoding is set? > Will FS ignore disable-transcoding parameter? > > On Sunday 15 August 2010, Michael Jerris wrote: >> http://wiki.freeswitch.org/wiki/Codec_negotiation >> >> disable-transcoding >> >> On Aug 15, 2010, at 8:07 PM, Sergey Okhapkin wrote: >>> To avoid transcoding if both endpoints have common codec, but one of >>> endpoints is behind NAT. >>> >>> On Sunday 15 August 2010, Michael Jerris wrote: >>>> I don't think there is ever a reason to use proxy_media anymore. If >>>> anyone can provide an example of a good reason to use it I would be >>>> interested to hear. >>>> >>>> On Aug 2, 2010, at 6:25 AM, Sergey Okhapkin wrote: >>>>> FS has 3 ways (not 2!) to handle media: >>>>> >>>>> 1. bypass_media=true - media path is set to run audio directly between >>>>> leg a and b endpoints. absolute_codec_string variable is ignored in >>>>> this mode. 2. proxy_media=true - media is proxied by FS, but without >>>>> any transcoding etc, frame in - frame out. absolute_codec_string >>>>> variable is ignored in this mode. 3. Neither bypass_media, no >>>>> proxy_media variables are set. FS does full media handling/transcoding, >>>>> absolute_codec_string variable is honored. From Nabble at slickdeals.endjunk.com Sun Aug 15 18:56:18 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Sun, 15 Aug 2010 18:56:18 -0700 (PDT) Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008020625.37601.sos@sokhapkin.dyndns.org> References: <201008020625.37601.sos@sokhapkin.dyndns.org> Message-ID: <1281923778178-5426489.post@n2.nabble.com> My preference is to use this bypass_media=true option to relief FS from proxying the media. This has some drawbacks, i.e. MOH won't work, etc. Sergey Okhapkin wrote: > 1. bypass_media=true - media path is set to run audio directly between leg > a and b endpoints. absolute_codec_string variable is ignored in this mode. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Proxy-Media-and-Codec-Prefer-tp5362727p5426489.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sos at sokhapkin.dyndns.org Sun Aug 15 19:15:26 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 15 Aug 2010 22:15:26 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <40C7FF38-D47F-4190-A0D6-44BA14390845@jerris.com> References: <201008152048.38275.sos@sokhapkin.dyndns.org> <40C7FF38-D47F-4190-A0D6-44BA14390845@jerris.com> Message-ID: <201008152215.26181.sos@sokhapkin.dyndns.org> Sure? How FS behaves now if disable-transcoding is set, late negotiation is on and proxy_media is not set? Will it forward packets as is if there are common codecs and fall back to transcoding if there are no common codecs on a and b leg? If this is current behavior, then I will definitely use it... On Sunday 15 August 2010, Michael Jerris wrote: > Proxy media does not handle this case now and the call will fail in the > same way. We will try to avoid transcoding by default, so I think what > you are suggesting is already the default action without proxy_media. > > On Aug 15, 2010, at 8:48 PM, Sergey Okhapkin wrote: > > But what if endpoints have no common codec and disable-transcoding is > > set? Will FS ignore disable-transcoding parameter? > > > > On Sunday 15 August 2010, Michael Jerris wrote: > >> http://wiki.freeswitch.org/wiki/Codec_negotiation > >> > >> disable-transcoding > >> > >> On Aug 15, 2010, at 8:07 PM, Sergey Okhapkin wrote: > >>> To avoid transcoding if both endpoints have common codec, but one of > >>> endpoints is behind NAT. > >>> > >>> On Sunday 15 August 2010, Michael Jerris wrote: > >>>> I don't think there is ever a reason to use proxy_media anymore. If > >>>> anyone can provide an example of a good reason to use it I would be > >>>> interested to hear. > >>>> > >>>> On Aug 2, 2010, at 6:25 AM, Sergey Okhapkin wrote: > >>>>> FS has 3 ways (not 2!) to handle media: > >>>>> > >>>>> 1. bypass_media=true - media path is set to run audio directly > >>>>> between leg a and b endpoints. absolute_codec_string variable is > >>>>> ignored in this mode. 2. proxy_media=true - media is proxied by FS, > >>>>> but without any transcoding etc, frame in - frame out. > >>>>> absolute_codec_string variable is ignored in this mode. 3. Neither > >>>>> bypass_media, no proxy_media variables are set. FS does full media > >>>>> handling/transcoding, absolute_codec_string variable is honored. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Sun Aug 15 19:37:52 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 22:37:52 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008152215.26181.sos@sokhapkin.dyndns.org> References: <201008152048.38275.sos@sokhapkin.dyndns.org> <40C7FF38-D47F-4190-A0D6-44BA14390845@jerris.com> <201008152215.26181.sos@sokhapkin.dyndns.org> Message-ID: <191DFB82-27FE-4A4D-BD2A-7EC0B2DC1E12@jerris.com> Just set late negotiation and it should do what you want. If you set disable-transcoding it will ? disable transcoding, failing the call if no common codecs. On Aug 15, 2010, at 10:15 PM, Sergey Okhapkin wrote: > Sure? How FS behaves now if disable-transcoding is set, late negotiation is on > and proxy_media is not set? Will it forward packets as is if there are common > codecs and fall back to transcoding if there are no common codecs on a and b > leg? If this is current behavior, then I will definitely use it... > > On Sunday 15 August 2010, Michael Jerris wrote: >> Proxy media does not handle this case now and the call will fail in the >> same way. We will try to avoid transcoding by default, so I think what >> you are suggesting is already the default action without proxy_media. >> >> On Aug 15, 2010, at 8:48 PM, Sergey Okhapkin wrote: >>> But what if endpoints have no common codec and disable-transcoding is >>> set? Will FS ignore disable-transcoding parameter? >>> >>> On Sunday 15 August 2010, Michael Jerris wrote: >>>> http://wiki.freeswitch.org/wiki/Codec_negotiation >>>> >>>> disable-transcoding >>>> >>>> On Aug 15, 2010, at 8:07 PM, Sergey Okhapkin wrote: >>>>> To avoid transcoding if both endpoints have common codec, but one of >>>>> endpoints is behind NAT. >>>>> >>>>> On Sunday 15 August 2010, Michael Jerris wrote: >>>>>> I don't think there is ever a reason to use proxy_media anymore. If >>>>>> anyone can provide an example of a good reason to use it I would be >>>>>> interested to hear. From mike at jerris.com Sun Aug 15 19:39:46 2010 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Aug 2010 22:39:46 -0400 Subject: [Freeswitch-users] SPA8000 In-Reply-To: <201008101837.38236.errotan@elder.hu> References: <81C2CEF80046FB4F863A60D4347DD33A0C559E@server1.st.local> <81C2CEF80046FB4F863A60D4347DD33A0C55A1@server1.st.local> <201008101837.38236.errotan@elder.hu> Message-ID: I can confirm we have working SPA8800 with FreeSWITCH. On Aug 10, 2010, at 12:37 PM, Pusk?s Zsolt wrote: > Hi. > > SPA8000 works perfectly for me. > Do you have NAT involved ? If so have you set NAT mapping and NAT keep alive > on ? > Please tell how you configured the SPA8000 because without info we can't help. > > > 2010. augusztus 10. 13:00:49 d?tummal Erkan ?nl? az al?bbiakat ?rta: >> I think the problem is in FreeSwitch. >> >> If I use other voip server that works. From msc at freeswitch.org Sun Aug 15 21:09:21 2010 From: msc at freeswitch.org (Michael Collins) Date: Sun, 15 Aug 2010 21:09:21 -0700 Subject: [Freeswitch-users] mod_xml_cdr flags b-leg with "a_" In-Reply-To: References: Message-ID: On Sun, Aug 15, 2010 at 5:34 PM, Nyamul Hassan wrote: > Hi, > > When a call hangs up normally, like either of the user kept the phone down, > then there are no issues. > > However, when the call is hung up through a transfer, like mod_nibblebill > does when "low_balance" or "no_balance" is reached, then the b-leg cdr.xml > file gets prefixed with "a_". > > Is this a desired feature? Or a bug? > Step us through the entire call flow. Also, pastebin an example set of files (a and b legs) for working vs. not working calls. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100815/9cb3ccf0/attachment.html From babak.freeswitch at gmail.com Sun Aug 15 22:52:03 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Mon, 16 Aug 2010 10:22:03 +0430 Subject: [Freeswitch-users] playing a message while bridging Message-ID: Hi is it possible to play a message while bridge is taking place, like I've tried many combinations but sometimes I get server failure and in the best case the message file is played like beep beep . .. (but it's a voice message)!!! and it never works for now I'm using answer and playback and then I bridge but I wanted to see if it is possible not to answer the call thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/6ef153a6/attachment.html From b_ball_henry at hotmail.com Sun Aug 15 23:00:30 2010 From: b_ball_henry at hotmail.com (Henry Huang) Date: Mon, 16 Aug 2010 14:00:30 +0800 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <40C7FF38-D47F-4190-A0D6-44BA14390845@jerris.com> References: <201008152007.00985.sos@sokhapkin.dyndns.org> <201008152048.38275.sos@sokhapkin.dyndns.org> <40C7FF38-D47F-4190-A0D6-44BA14390845@jerris.com> Message-ID: Michael: Since what version does this become a default behavior? I was using 1.0.5 and I had to use proxy_media=true for the A -> FS -> B to avoid g729 codec priority issue. Let me explain: A has codec priority as g711, g729 FS has codec priority as g711, g729(passthrough) B has codec priority as g729 , then whatever esle *Without* proxy media= true: 1. A will negotiate with FS about which codec to use, since they both have g711 as the first priority, they will agree on using g711 to communicate. 2. Now FS will negotiate codec with B, and since B has g729 as priority, and FS has g729(passthrough) available, they will agree on using g729 as the codec. 3. Now the calls will bridge and drop right away since FS does not do codec translation. With proxy media = true 1. A will skip FS and negotiate codec with B directly. And they will use whatever matches they can find, and this will leave FS not worrying about codec translation at all. please correct me if the above scenario has been changed in the newer FS. Thanks, Henry Huang Unified Communication System R&D Project Manager US: +1 (626) 606-3306 Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me [image: Linkedin][image: Facebook] [image: Twitter] On Mon, Aug 16, 2010 at 9:53 AM, Michael Jerris wrote: > Proxy media does not handle this case now and the call will fail in the > same way. We will try to avoid transcoding by default, so I think what you > are suggesting is already the default action without proxy_media. > > On Aug 15, 2010, at 8:48 PM, Sergey Okhapkin wrote: > > > But what if endpoints have no common codec and disable-transcoding is > set? > > Will FS ignore disable-transcoding parameter? > > > > On Sunday 15 August 2010, Michael Jerris wrote: > >> http://wiki.freeswitch.org/wiki/Codec_negotiation > >> > >> disable-transcoding > >> > >> On Aug 15, 2010, at 8:07 PM, Sergey Okhapkin wrote: > >>> To avoid transcoding if both endpoints have common codec, but one of > >>> endpoints is behind NAT. > >>> > >>> On Sunday 15 August 2010, Michael Jerris wrote: > >>>> I don't think there is ever a reason to use proxy_media anymore. If > >>>> anyone can provide an example of a good reason to use it I would be > >>>> interested to hear. > >>>> > >>>> On Aug 2, 2010, at 6:25 AM, Sergey Okhapkin wrote: > >>>>> FS has 3 ways (not 2!) to handle media: > >>>>> > >>>>> 1. bypass_media=true - media path is set to run audio directly > between > >>>>> leg a and b endpoints. absolute_codec_string variable is ignored in > >>>>> this mode. 2. proxy_media=true - media is proxied by FS, but without > >>>>> any transcoding etc, frame in - frame out. absolute_codec_string > >>>>> variable is ignored in this mode. 3. Neither bypass_media, no > >>>>> proxy_media variables are set. FS does full media > handling/transcoding, > >>>>> absolute_codec_string variable is honored. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/38ae4ccb/attachment-0001.html From nagalenoj at gmail.com Sun Aug 15 23:15:36 2010 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Mon, 16 Aug 2010 11:45:36 +0530 Subject: [Freeswitch-users] playing a message while bridging In-Reply-To: References: Message-ID: May be you want this, http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_aleg_app On Mon, Aug 16, 2010 at 11:22 AM, babak yakhchali < babak.freeswitch at gmail.com> wrote: > Hi is it possible to play a message while bridge is taking place, like > > > I've tried many combinations but sometimes I get server failure and in the > best case the message file is played like beep beep . .. (but it's a voice > message)!!! and it never works > for now I'm using answer and playback and then I bridge but I wanted to see > if it is possible not to answer the call > thanx > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Nagalenoj H. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/e0d5f63c/attachment.html From mike at jerris.com Sun Aug 15 23:20:38 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Aug 2010 02:20:38 -0400 Subject: [Freeswitch-users] playing a message while bridging In-Reply-To: References: Message-ID: You can playback without answer. On Aug 16, 2010, at 1:52 AM, babak yakhchali wrote: > Hi is it possible to play a message while bridge is taking place, like > > > I've tried many combinations but sometimes I get server failure and in the best case the message file is played like beep beep . .. (but it's a voice message)!!! and it never works > for now I'm using answer and playback and then I bridge but I wanted to see if it is possible not to answer the call From mike at jerris.com Sun Aug 15 23:21:36 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Aug 2010 02:21:36 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: <201008152007.00985.sos@sokhapkin.dyndns.org> <201008152048.38275.sos@sokhapkin.dyndns.org> <40C7FF38-D47F-4190-A0D6-44BA14390845@jerris.com> Message-ID: <1D562E19-9B73-4B02-959A-5EE77A11F466@jerris.com> Feel free to confirm for yourself, check out latest git and report back to all. On Aug 16, 2010, at 2:00 AM, Henry Huang wrote: > Michael: > > Since what version does this become a default behavior? I was using 1.0.5 and I had to use proxy_media=true for the A -> FS -> B to avoid g729 codec priority issue. Let me explain: > A has codec priority as g711, g729 > FS has codec priority as g711, g729(passthrough) > B has codec priority as g729 , then whatever esle > > Without proxy media= true: > 1. A will negotiate with FS about which codec to use, since they both have g711 as the first priority, they will agree on using g711 to communicate. > 2. Now FS will negotiate codec with B, and since B has g729 as priority, and FS has g729(passthrough) available, they will agree on using g729 as the codec. > 3. Now the calls will bridge and drop right away since FS does not do codec translation. > > With proxy media = true > 1. A will skip FS and negotiate codec with B directly. And they will use whatever matches they can find, and this will leave FS not worrying about codec translation at all. > > please correct me if the above scenario has been changed in the newer FS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/b66b2600/attachment.html From a.afzali2003 at gmail.com Sun Aug 15 23:31:39 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Mon, 16 Aug 2010 11:01:39 +0430 Subject: [Freeswitch-users] playing a message while bridging In-Reply-To: References: Message-ID: babak, use bridge_pre_execute_aleg_app / bridge_pre_execute_bleg_app to play message. -- afshin On Mon, Aug 16, 2010 at 10:22 AM, babak yakhchali < babak.freeswitch at gmail.com> wrote: > Hi is it possible to play a message while bridge is taking place, like > > > I've tried many combinations but sometimes I get server failure and in the > best case the message file is played like beep beep . .. (but it's a voice > message)!!! and it never works > for now I'm using answer and playback and then I bridge but I wanted to see > if it is possible not to answer the call > thanx > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/496a6b2f/attachment.html From babak.freeswitch at gmail.com Sun Aug 15 23:34:14 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Mon, 16 Aug 2010 11:04:14 +0430 Subject: [Freeswitch-users] playing a message while bridging In-Reply-To: References: Message-ID: thanx but when I playback before answer, message and early media (beep.....beep...) are mixed together! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/54db0ea3/attachment.html From lists at infosecurity.ch Mon Aug 16 01:39:57 2010 From: lists at infosecurity.ch (Fabio Pietrosanti (naif)) Date: Mon, 16 Aug 2010 10:39:57 +0200 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: <201008020625.37601.sos@sokhapkin.dyndns.org> Message-ID: <4C68F95D.6060102@infosecurity.ch> I use proxy_media as a default because i use only mobile clients. In mobile networks there are good chance that FS NAT detection does not work. That's because several mobile carriers use as "internal ip address" IP that are not inside the standard 192.168/16 10/8 and 172.16/16 networks. For example H3G in italy use 1/8 networks and in other countries there are other fancy approach that are unpredictable. In such condition proxy_media is the only solution to have a 100% working solution, otherwise in a X% of the case on some mobile operator it will just don't work. And following all the "particular" internal networks that mobile operators follow is not a good way to manage the issue as there are too many mobile operators in the world and too many different internal network conditions. It would be really nice to have some kind of "dynamic" approach, like the ones provided by ICE, that are client-independent and server-side based to handle such particular issues (IE: without the ability to rely on client IP address being inside a particular network). Fabio On 16/08/10 01.56, Michael Jerris wrote: > I don't think there is ever a reason to use proxy_media anymore. If anyone can provide an example of a good reason to use it I would be interested to hear. > From steveayre at gmail.com Mon Aug 16 01:59:41 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 16 Aug 2010 09:59:41 +0100 Subject: [Freeswitch-users] Failed to compile sofia.c for ARM In-Reply-To: <1281919355652-5426385.post@n2.nabble.com> References: <1278472465333-5263442.post@n2.nabble.com> <1278472842929-5263450.post@n2.nabble.com> <1278507728660-5265110.post@n2.nabble.com> <1278517888279-5265912.post@n2.nabble.com> <1278543145818-5267704.post@n2.nabble.com> <1281809261831-5423774.post@n2.nabble.com> <640CFC68-974D-4E28-AEE2-E82E20E54E62@jerris.com> <1281919355652-5426385.post@n2.nabble.com> Message-ID: <2E75D3E7-A5C2-4F62-BED3-7C18A9976932@gmail.com> Do it anyway... It allows the problem to be logged and tracked, and gives an id to use in commits. Freeswitch can then be patched so openwrt no longer need to. Then the patch can be pushed upstream to nokia referencing the Jira ticket which will contain a full description of the issue with patches attached, which will result in all copies of Sofia being fixed, not just openwrt. That then makes less work for openwrt and freeswitch as they no longer have to maintain a patch which might break against each new release of Sofia. In short, just because openwrt have fixed it doesn't mean it's not necessary to fix it elsewhere. Steve on iPhone On 16 Aug 2010, at 01:42, mazilo wrote: > > Hi Mike, > > I don't think it is necessary because the problem has been fixed by OpenWRT. > > > Michael Jerris wrote: >> >> Please open a bug on jira on this issue for me with all those details >> attached. > > > ----- > don't and stop are the ONLY two 4-letter words considered offensive to men, > but not when used together. > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Failed-to-compile-sofia-c-for-ARM-tp5252219p5426385.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveayre at gmail.com Mon Aug 16 02:02:18 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 16 Aug 2010 10:02:18 +0100 Subject: [Freeswitch-users] playing a message while bridging In-Reply-To: References: Message-ID: <55192CF9-A167-4BC7-B282-B794BAEAA690@gmail.com> Try using ignore early media too. Steve on iPhone On 16 Aug 2010, at 07:34, babak yakhchali wrote: > thanx > but when I playback before answer, message and early media (beep.....beep...) are mixed together! > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From babak.freeswitch at gmail.com Mon Aug 16 02:15:26 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Mon, 16 Aug 2010 13:45:26 +0430 Subject: [Freeswitch-users] playing a message while bridging In-Reply-To: <55192CF9-A167-4BC7-B282-B794BAEAA690@gmail.com> References: <55192CF9-A167-4BC7-B282-B794BAEAA690@gmail.com> Message-ID: tried this: still my message and early media are mixing together :( -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/3ccce3b3/attachment.html From ktngl at yahoo.co.uk Mon Aug 16 02:25:26 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Mon, 16 Aug 2010 09:25:26 +0000 (GMT) Subject: [Freeswitch-users] listen for key presses during a live call In-Reply-To: <8C74E9E1-7188-4631-AD63-2424C3EF1BC4@ipeva.fr> Message-ID: <764220.38091.qm@web29218.mail.ird.yahoo.com> Is there a way to listen for key presses during a live call -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/5f58e255/attachment.html From david.ponzone at ipeva.fr Mon Aug 16 02:39:48 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 16 Aug 2010 11:39:48 +0200 Subject: [Freeswitch-users] listen for key presses during a live call In-Reply-To: <764220.38091.qm@web29218.mail.ird.yahoo.com> References: <764220.38091.qm@web29218.mail.ird.yahoo.com> Message-ID: bind_meta_app David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 16/08/2010 ? 11:25, Nigel Kent a ?crit : > Is there a way to listen for key presses during a live call > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/7a530566/attachment.html From daniel.neubert at solomo.de Mon Aug 16 02:42:44 2010 From: daniel.neubert at solomo.de (Daniel Neubert) Date: Mon, 16 Aug 2010 11:42:44 +0200 Subject: [Freeswitch-users] listen for key presses during a live call In-Reply-To: <764220.38091.qm@web29218.mail.ird.yahoo.com> References: <764220.38091.qm@web29218.mail.ird.yahoo.com> Message-ID: <4C690814.4080709@solomo.de> You only need to execute start_dtmf from dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf Regards Daniel Neubert On 16.08.2010 11:25, Nigel Kent wrote: > Is there a way to listen for key presses during a live call > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/799c934b/attachment.html From david.ponzone at ipeva.fr Mon Aug 16 02:54:29 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 16 Aug 2010 11:54:29 +0200 Subject: [Freeswitch-users] listen for key presses during a live call In-Reply-To: <4C690814.4080709@solomo.de> References: <764220.38091.qm@web29218.mail.ird.yahoo.com> <4C690814.4080709@solomo.de> Message-ID: <25991561-7D5B-4F68-926A-13077762BDB7@ipeva.fr> start_dtmf, according the wiki, only enables inband DTMF detection. "OOB" DTMF detection is enabled by default. You then have to do something with the keypresses, either with your own script talking with event socket, or with bind_meta_app. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 16/08/2010 ? 11:42, Daniel Neubert a ?crit : > You only need to execute start_dtmf from dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > > Regards > Daniel Neubert > > > On 16.08.2010 11:25, Nigel Kent wrote: >> >> Is there a way to listen for key presses during a live call >> >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/505d7ab1/attachment-0001.html From Nabble at slickdeals.endjunk.com Mon Aug 16 05:21:06 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Mon, 16 Aug 2010 05:21:06 -0700 (PDT) Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008020625.37601.sos@sokhapkin.dyndns.org> References: <201008020625.37601.sos@sokhapkin.dyndns.org> Message-ID: <1281961266262-5427635.post@n2.nabble.com> Currently, I included bypass_media=true to every dialplan file, but prefer to make it a default setting. So, which file should I edit to include bypass_media=true as a default setting for the whole FS system? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Proxy-Media-and-Codec-Prefer-tp5362727p5427635.html Sent from the freeswitch-users mailing list archive at Nabble.com. From odermann at googlemail.com Mon Aug 16 05:56:49 2010 From: odermann at googlemail.com (Dennis) Date: Mon, 16 Aug 2010 14:56:49 +0200 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> Message-ID: phillip, it seems that we have somehow the same problem, but we have an additional strange behavior. we used wireshark on the incoming network-card and on the outgoing side. there we see two strange things, we can not figure out: 1.) on the incoming side, we receive the in-band tone from the cirpack with a tiny gap, but fs recognizes them as 2 tones. in wireshark it look something like this: | |||||||| as i said, fs sees this as two tones. 2.) fs sends the tone above to the outgoing side as two tones, but seems to cut the first ms of the second part. in wireshark it looks something like this: | ||| i feel, that we have to find out the following: 1.) why does the tone already have a gap, when we receive it? 2.) why does fs not leave the tone untouched? something happens with the tone, while passing fs. is there a setting to avoid this? 3.) why is there no easy way to completely delete dtmf tones, if it is possible with vuvuzela-noise? ;-) dennis From mnhassan at usa.net Mon Aug 16 06:05:13 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 16 Aug 2010 19:05:13 +0600 Subject: [Freeswitch-users] listen for key presses during a live call In-Reply-To: <25991561-7D5B-4F68-926A-13077762BDB7@ipeva.fr> References: <764220.38091.qm@web29218.mail.ird.yahoo.com> <4C690814.4080709@solomo.de> <25991561-7D5B-4F68-926A-13077762BDB7@ipeva.fr> Message-ID: David is right, I'm using "bind_meta_app" for this, and it works like a charm. Regards HASSAN 2010/8/16 David Ponzone > start_dtmf, according the wiki, only enables inband DTMF detection. > > "OOB" DTMF detection is enabled by default. > > You then have to do something with the keypresses, either with your own > script talking with event socket, or with bind_meta_app. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 16/08/2010 ? 11:42, Daniel Neubert a ?crit : > > You only need to execute start_dtmf from dialplan: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > > Regards > Daniel Neubert > > > On 16.08.2010 11:25, Nigel Kent wrote: > > Is there a way to listen for key presses during a live call > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/8eaa5296/attachment.html From brian at freeswitch.org Mon Aug 16 06:20:22 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Aug 2010 08:20:22 -0500 Subject: [Freeswitch-users] Strange early media SDP behavior In-Reply-To: <201008152000.32846.sos@sokhapkin.dyndns.org> References: <201008141119.07759.sos@sokhapkin.dyndns.org> <201008152000.32846.sos@sokhapkin.dyndns.org> Message-ID: <349A61FB-7C45-4212-A266-17C258ED6AF4@freeswitch.org> I suspect you have FreeSWITCH generating ringback..... because we NEVER answer with a 200 OK with more then one codec. /b On Aug 15, 2010, at 7:00 PM, Sergey Okhapkin wrote: > Any feedback from FS developers? From sos at sokhapkin.dyndns.org Mon Aug 16 06:29:25 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 16 Aug 2010 09:29:25 -0400 Subject: [Freeswitch-users] Strange early media SDP behavior In-Reply-To: <349A61FB-7C45-4212-A266-17C258ED6AF4@freeswitch.org> References: <201008141119.07759.sos@sokhapkin.dyndns.org> <201008152000.32846.sos@sokhapkin.dyndns.org> <349A61FB-7C45-4212-A266-17C258ED6AF4@freeswitch.org> Message-ID: <201008160929.25403.sos@sokhapkin.dyndns.org> No ringbacks, it's bypass_media mode. On Monday 16 August 2010, Brian West wrote: > I suspect you have FreeSWITCH generating ringback..... because we NEVER > answer with a 200 OK with more then one codec. > > /b > > On Aug 15, 2010, at 7:00 PM, Sergey Okhapkin wrote: > > Any feedback from FS developers? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From patrick.violin at gmail.com Mon Aug 16 03:16:00 2010 From: patrick.violin at gmail.com (Patrick Hsieh) Date: Mon, 16 Aug 2010 12:16:00 +0200 Subject: [Freeswitch-users] Did FreeSWITCH modules provides Vicidials features? Message-ID: Hello, I am a new user of FreeSWITCH. I have installed the Asterisk and Vicidial applications in my notebook. Q1: I would like to know are there any similar features which already be implemented by FreeSWITCH itself? Q2: Which will be a efficient way to have the features in FreeSWITCH? Rewrite all features in FreeSWITCH or create a plug-in modules to Vicidial directly. Q3: Did FreeSWITCH already support the ACD and Predictive Dialing features? Which modules releated to the features? Please advices! I am willing to contribute myself to extend the FreeSWITCH features. Best Regards, Patrick MAJOR VICIDIAL FEATURES: - Inbound, Outbound and Blended call handling - Outbound agent-controlled, broadcast and predictive dialing - Full USA FTC-compliance capability - Web-based agent and administrative interfaces - Ability to have agents operate remotely - Integrated call recording - Three-Way calling within the agent application - Scheduled Callbacks: Agent-Only and Anyone - Web-configurable IVRs and Voicemail boxes - Scalable to hundreds of seats - Ability to use standard Telco lines and VOIP trunks - Open-Source AGPLv2 licensed, with no software licensing cost -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/f7bd1fa4/attachment-0001.html From gianluca.varisco at privatewave.com Mon Aug 16 06:33:34 2010 From: gianluca.varisco at privatewave.com (Gianluca Varisco) Date: Mon, 16 Aug 2010 15:33:34 +0200 Subject: [Freeswitch-users] TLS problem using SNOM phones In-Reply-To: References: Message-ID: <4C693E2E.4080402@privatewave.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 08/13/2010 11:25 PM, Matthias Reinacher wrote: > Hello all, > i have a problem using different Snom phones (300/320/820/821) with > Freeswitch and TLS in a internet-wide setup (phones being registered via > internet at a FS w/ public IP). Sometimes (25-50% of cases) the first > try to call someone (internal) won't go through. Logs from phone, > Freeswitch and ssldump show that the phone sends an INVITE, the FS asks > for more credentials (digest auth b/c phone IP not in ACL), the phone > answers with an INVITE w/ more auth credentials -- and this second > INVITE package is not received by the FS. It does arrive at the server > though, as verified by ssldump. Interestingly, if one presses "Cancel" > on the phone, a new TCP/SSL connection is created, apparently the old > one died -- see logs from phone, FS, and ssldump below. Also, Snom > 820/821 show the registrar as not registered after such an action > (presumably b/c original SSL connection is dead). > > Has anybody encountered this behaviour and can verify it? Are there any > ideas what causes the problem here (FS, OpenSSL, phone?) and if there is > any remedy for it? I was planning on using FS as a production phone > system w/ encrypted signalling and audio using Snom phones. I can't do > that right now b/c of the abovementioned problem, and i have not yet > found a solution. > Hello Matthias, I'm facing a similar issue and already reported it on http://jira.freeswitch.org/browse/FSCONFIG-28 . My environment is made up of mobile clients and, therefore, unreliable networks. It happens that, if "userA" unexpectedly looses the connection and - thus - his TCP connection is not closed correctly (no UNREGISTER being sent), further calls to him 'float in the air' and are not delivered. User remains registered but his socket is - simply - dead. As ugly workaround, I've to use http://pastebin.com/sBUraBw9 Let me know if you find any other way to 'handle' this problem. Cheers, Gianluca -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.14 (GNU/Linux) iQEcBAEBAgAGBQJMaT4uAAoJEOPkCSSH2GC78C8H/A2HK6FDMrlyyncQUyb2eMeM 8vL4zXWtL5XzvdxVHChbz6TZvDTNXrz8FbU+Kqa6VpAFyXnRj9fMz9Ayqy7CMsC3 ed+9nxX8hmghlrVaStfmCyfsMW1UOiNEYHiELjzZBaNVQvkUrVx0ezTS4ZCk5RFq em9rV3ooO3be1qDP25AeSycB3YRPp6ev9SXg9KgccGpQhllEx5cHgbH9P9YvTfjN u2ubm/Zf6waV8VMeY0t5nqq5ihtTO2b3/SvTOrIdEvzVqboSnu5XAeSZOhsZz3SN 3Moc58bvb1ikYtZ1ZXt5f+r5Tq/nPQ3IhN7ZQih4vh61a3+KsW8uhFAH0rx0x7Q= =H1pI -----END PGP SIGNATURE----- From mike at jerris.com Mon Aug 16 06:44:24 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Aug 2010 09:44:24 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <4C68F95D.6060102@infosecurity.ch> References: <201008020625.37601.sos@sokhapkin.dyndns.org> <4C68F95D.6060102@infosecurity.ch> Message-ID: <4F16FEFA-6152-4A2C-9D47-6C04050CA087@jerris.com> This is a case of your nat settings simply not being right on the FreeSWITCH side I suspect, study the acl settings and come up with an example of a failure here with full sip trace and debug. Mike On Aug 16, 2010, at 4:39 AM, Fabio Pietrosanti (naif) wrote: > I use proxy_media as a default because i use only mobile clients. > > In mobile networks there are good chance that FS NAT detection does not > work. > > That's because several mobile carriers use as "internal ip address" IP > that are not inside the standard 192.168/16 10/8 and 172.16/16 networks. > > For example H3G in italy use 1/8 networks and in other countries there > are other fancy approach that are unpredictable. > In such condition proxy_media is the only solution to have a 100% > working solution, otherwise in a X% of the case on some mobile operator > it will just don't work. > > And following all the "particular" internal networks that mobile > operators follow is not a good way to manage the issue as there are too > many mobile operators in the world and too many different internal > network conditions. > > It would be really nice to have some kind of "dynamic" approach, like > the ones provided by ICE, that are client-independent and server-side > based to handle such particular issues (IE: without the ability to rely > on client IP address being inside a particular network). > > Fabio > > On 16/08/10 01.56, Michael Jerris wrote: >> I don't think there is ever a reason to use proxy_media anymore. If anyone can provide an example of a good reason to use it I would be interested to hear. >> > From sos at sokhapkin.dyndns.org Mon Aug 16 07:02:28 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 16 Aug 2010 10:02:28 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <4F16FEFA-6152-4A2C-9D47-6C04050CA087@jerris.com> References: <4C68F95D.6060102@infosecurity.ch> <4F16FEFA-6152-4A2C-9D47-6C04050CA087@jerris.com> Message-ID: <201008161002.28245.sos@sokhapkin.dyndns.org> http://pastebin.freeswitch.org/13643 , note lines 185 (SDP from B leg, 3 codecs) and 260 (SDP sent to A leg, only 1 codec). The call is running with bypass_media=true. On Monday 16 August 2010, Michael Jerris wrote: > This is a case of your nat settings simply not being right on the > FreeSWITCH side I suspect, study the acl settings and come up with an > example of a failure here with full sip trace and debug. > > Mike > > On Aug 16, 2010, at 4:39 AM, Fabio Pietrosanti (naif) wrote: > > I use proxy_media as a default because i use only mobile clients. > > > > In mobile networks there are good chance that FS NAT detection does not > > work. > > > > That's because several mobile carriers use as "internal ip address" IP > > that are not inside the standard 192.168/16 10/8 and 172.16/16 networks. > > > > For example H3G in italy use 1/8 networks and in other countries there > > are other fancy approach that are unpredictable. > > In such condition proxy_media is the only solution to have a 100% > > working solution, otherwise in a X% of the case on some mobile operator > > it will just don't work. > > > > And following all the "particular" internal networks that mobile > > operators follow is not a good way to manage the issue as there are too > > many mobile operators in the world and too many different internal > > network conditions. > > > > It would be really nice to have some kind of "dynamic" approach, like > > the ones provided by ICE, that are client-independent and server-side > > based to handle such particular issues (IE: without the ability to rely > > on client IP address being inside a particular network). > > > > Fabio > > > > On 16/08/10 01.56, Michael Jerris wrote: > >> I don't think there is ever a reason to use proxy_media anymore. If > >> anyone can provide an example of a good reason to use it I would be > >> interested to hear. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Mon Aug 16 08:00:01 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Aug 2010 11:00:01 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008161002.28245.sos@sokhapkin.dyndns.org> References: <4C68F95D.6060102@infosecurity.ch> <4F16FEFA-6152-4A2C-9D47-6C04050CA087@jerris.com> <201008161002.28245.sos@sokhapkin.dyndns.org> Message-ID: In bypass media we pass the sdp through directly. I am unsure what you are trying to show me here and what it has to do with the problem you describe. Mike On Aug 16, 2010, at 10:02 AM, Sergey Okhapkin wrote: > http://pastebin.freeswitch.org/13643 , note lines 185 (SDP from B leg, 3 > codecs) and 260 (SDP sent to A leg, only 1 codec). The call is running with > bypass_media=true. > > On Monday 16 August 2010, Michael Jerris wrote: >> This is a case of your nat settings simply not being right on the >> FreeSWITCH side I suspect, study the acl settings and come up with an >> example of a failure here with full sip trace and debug. >> >> Mike >> >> On Aug 16, 2010, at 4:39 AM, Fabio Pietrosanti (naif) wrote: >>> I use proxy_media as a default because i use only mobile clients. >>> >>> In mobile networks there are good chance that FS NAT detection does not >>> work. >>> >>> That's because several mobile carriers use as "internal ip address" IP >>> that are not inside the standard 192.168/16 10/8 and 172.16/16 networks. >>> >>> For example H3G in italy use 1/8 networks and in other countries there >>> are other fancy approach that are unpredictable. >>> In such condition proxy_media is the only solution to have a 100% >>> working solution, otherwise in a X% of the case on some mobile operator >>> it will just don't work. >>> >>> And following all the "particular" internal networks that mobile >>> operators follow is not a good way to manage the issue as there are too >>> many mobile operators in the world and too many different internal >>> network conditions. >>> >>> It would be really nice to have some kind of "dynamic" approach, like >>> the ones provided by ICE, that are client-independent and server-side >>> based to handle such particular issues (IE: without the ability to rely >>> on client IP address being inside a particular network). >>> >>> Fabio >>> >>> On 16/08/10 01.56, Michael Jerris wrote: >>>> I don't think there is ever a reason to use proxy_media anymore. If >>>> anyone can provide an example of a good reason to use it I would be >>>> interested to hear. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pjintheusa at gmail.com Mon Aug 16 08:03:47 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 16 Aug 2010 11:03:47 -0400 Subject: [Freeswitch-users] Enterprise Originate and group_confirm In-Reply-To: References: <23CC36CD-05AF-4612-A21A-CA3013DC2668@freeswitch.org> Message-ID: I created a feature request in jira for this http://jira.freeswitch.org/browse/FSCORE-651 On Fri, Aug 13, 2010 at 3:24 PM, Phillip Jones wrote: > Ok - thanks Brian > > I have some further testing and it does not look like > "fail_on_single_reject" applys to the | (OR) , just to the , (AND) > So with , > > > data="{leg_timeout=10,ignore_early_media=true}sofia/gateway/bad_gateway/2129996599,sofia/gateway/broadvox1/2129996599"/> > > - you will not get the call because bad_gateway fails > > But with | > > > data="{leg_timeout=10,ignore_early_media=true}sofia/gateway/bad_gateway/2129996599|sofia/gateway/broadvox1/2129996599"/> > > You do get the call - fail_on_single_reject is ignored. I tried all > combination with export, {} etc > > Make sense? > > > > On Fri, Aug 13, 2010 at 1:09 PM, Brian West wrote: > >> try export instead of set... but you should set all those inside the {} or >> export nolocal >> >> /b >> >> On Aug 13, 2010, at 11:57 AM, Phillip Jones wrote: >> >> > Ok I know am talking to myself here but perhaps this useful to someone. >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/fe925238/attachment.html From sos at sokhapkin.dyndns.org Mon Aug 16 08:12:32 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 16 Aug 2010 11:12:32 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: References: <201008161002.28245.sos@sokhapkin.dyndns.org> Message-ID: <201008161112.32497.sos@sokhapkin.dyndns.org> Yes, it's supposed to be passed directly (and it is passed directly on 200 OK), but only 1st codec is passed to leg A on SIP 183! On Monday 16 August 2010, Michael Jerris wrote: > In bypass media we pass the sdp through directly. I am unsure what you are > trying to show me here and what it has to do with the problem you > describe. > > Mike > > On Aug 16, 2010, at 10:02 AM, Sergey Okhapkin wrote: > > http://pastebin.freeswitch.org/13643 , note lines 185 (SDP from B leg, 3 > > codecs) and 260 (SDP sent to A leg, only 1 codec). The call is running > > with bypass_media=true. > > > > On Monday 16 August 2010, Michael Jerris wrote: > >> This is a case of your nat settings simply not being right on the > >> FreeSWITCH side I suspect, study the acl settings and come up with an > >> example of a failure here with full sip trace and debug. > >> > >> Mike > >> > >> On Aug 16, 2010, at 4:39 AM, Fabio Pietrosanti (naif) wrote: > >>> I use proxy_media as a default because i use only mobile clients. > >>> > >>> In mobile networks there are good chance that FS NAT detection does not > >>> work. > >>> > >>> That's because several mobile carriers use as "internal ip address" IP > >>> that are not inside the standard 192.168/16 10/8 and 172.16/16 > >>> networks. > >>> > >>> For example H3G in italy use 1/8 networks and in other countries there > >>> are other fancy approach that are unpredictable. > >>> In such condition proxy_media is the only solution to have a 100% > >>> working solution, otherwise in a X% of the case on some mobile operator > >>> it will just don't work. > >>> > >>> And following all the "particular" internal networks that mobile > >>> operators follow is not a good way to manage the issue as there are too > >>> many mobile operators in the world and too many different internal > >>> network conditions. > >>> > >>> It would be really nice to have some kind of "dynamic" approach, like > >>> the ones provided by ICE, that are client-independent and server-side > >>> based to handle such particular issues (IE: without the ability to rely > >>> on client IP address being inside a particular network). > >>> > >>> Fabio > >>> > >>> On 16/08/10 01.56, Michael Jerris wrote: > >>>> I don't think there is ever a reason to use proxy_media anymore. If > >>>> anyone can provide an example of a good reason to use it I would be > >>>> interested to hear. > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon Aug 16 08:19:36 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Aug 2010 10:19:36 -0500 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008161112.32497.sos@sokhapkin.dyndns.org> References: <201008161002.28245.sos@sokhapkin.dyndns.org> <201008161112.32497.sos@sokhapkin.dyndns.org> Message-ID: <8B4767A4-8C7C-45FF-86CE-C296C48CB96C@freeswitch.org> Who sends the 183? the far side? And please limit your issue to ONE thread on the list not two please. I still haven't seen a full console log with sip trace. /b On Aug 16, 2010, at 10:12 AM, Sergey Okhapkin wrote: > Yes, it's supposed to be passed directly (and it is passed directly on 200 > OK), but only 1st codec is passed to leg A on SIP 183! > > On Monday 16 August 2010, Michael Jerris wrote: >> In bypass media we pass the sdp through directly. I am unsure what you are >> trying to show me here and what it has to do with the problem you >> describe. >> >> Mike From sos at sokhapkin.dyndns.org Mon Aug 16 08:23:56 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 16 Aug 2010 11:23:56 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <201008161112.32497.sos@sokhapkin.dyndns.org> References: <201008161112.32497.sos@sokhapkin.dyndns.org> Message-ID: <201008161123.56754.sos@sokhapkin.dyndns.org> I wonder why the last line in the log says skipping state [early][183] for leg A? Code in sofia.c do not copy SDP in this case. On Monday 16 August 2010, Sergey Okhapkin wrote: > Yes, it's supposed to be passed directly (and it is passed directly on 200 > OK), but only 1st codec is passed to leg A on SIP 183! > > On Monday 16 August 2010, Michael Jerris wrote: > > In bypass media we pass the sdp through directly. I am unsure what you > > are trying to show me here and what it has to do with the problem you > > describe. > > > > Mike > > > > On Aug 16, 2010, at 10:02 AM, Sergey Okhapkin wrote: > > > http://pastebin.freeswitch.org/13643 , note lines 185 (SDP from B leg, > > > 3 codecs) and 260 (SDP sent to A leg, only 1 codec). The call is > > > running with bypass_media=true. > > > > > > On Monday 16 August 2010, Michael Jerris wrote: > > >> This is a case of your nat settings simply not being right on the > > >> FreeSWITCH side I suspect, study the acl settings and come up with an > > >> example of a failure here with full sip trace and debug. > > >> > > >> Mike > > >> > > >> On Aug 16, 2010, at 4:39 AM, Fabio Pietrosanti (naif) wrote: > > >>> I use proxy_media as a default because i use only mobile clients. > > >>> > > >>> In mobile networks there are good chance that FS NAT detection does > > >>> not work. > > >>> > > >>> That's because several mobile carriers use as "internal ip address" > > >>> IP that are not inside the standard 192.168/16 10/8 and 172.16/16 > > >>> networks. > > >>> > > >>> For example H3G in italy use 1/8 networks and in other countries > > >>> there are other fancy approach that are unpredictable. > > >>> In such condition proxy_media is the only solution to have a 100% > > >>> working solution, otherwise in a X% of the case on some mobile > > >>> operator it will just don't work. > > >>> > > >>> And following all the "particular" internal networks that mobile > > >>> operators follow is not a good way to manage the issue as there are > > >>> too many mobile operators in the world and too many different > > >>> internal network conditions. > > >>> > > >>> It would be really nice to have some kind of "dynamic" approach, like > > >>> the ones provided by ICE, that are client-independent and server-side > > >>> based to handle such particular issues (IE: without the ability to > > >>> rely on client IP address being inside a particular network). > > >>> > > >>> Fabio > > >>> > > >>> On 16/08/10 01.56, Michael Jerris wrote: > > >>>> I don't think there is ever a reason to use proxy_media anymore. If > > >>>> anyone can provide an example of a good reason to use it I would be > > >>>> interested to hear. > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > >>rs http://www.freeswitch.org > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > > >s http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sos at sokhapkin.dyndns.org Mon Aug 16 08:31:09 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 16 Aug 2010 11:31:09 -0400 Subject: [Freeswitch-users] Proxy Media and Codec Prefer In-Reply-To: <8B4767A4-8C7C-45FF-86CE-C296C48CB96C@freeswitch.org> References: <201008161112.32497.sos@sokhapkin.dyndns.org> <8B4767A4-8C7C-45FF-86CE-C296C48CB96C@freeswitch.org> Message-ID: <201008161131.09600.sos@sokhapkin.dyndns.org> The log is on pastebin. You're correct, 183 comes from far side, from called PSTN gateway on leg B. On Monday 16 August 2010, Brian West wrote: > Who sends the 183? the far side? And please limit your issue to ONE > thread on the list not two please. I still haven't seen a full console > log with sip trace. > > /b > > On Aug 16, 2010, at 10:12 AM, Sergey Okhapkin wrote: > > Yes, it's supposed to be passed directly (and it is passed directly on > > 200 OK), but only 1st codec is passed to leg A on SIP 183! > > > > On Monday 16 August 2010, Michael Jerris wrote: > >> In bypass media we pass the sdp through directly. I am unsure what you > >> are trying to show me here and what it has to do with the problem you > >> describe. > >> > >> Mike > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mnhassan at usa.net Mon Aug 16 09:13:43 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 16 Aug 2010 22:13:43 +0600 Subject: [Freeswitch-users] mod_xml_cdr flags b-leg with "a_" In-Reply-To: References: Message-ID: I just found another problem: One CDR file of a b_leg is showing "xx_uuid" of the originator... but the CDR file for the a_leg (which has the uuid of "xx_uuid") does not show any mention of the "originatee". And, this is one those cases where the b_leg cdr.xml was prefixed with a "a_" Pastebin of both CDR files is in: http://pastebin.freeswitch.org/13646 Regards HASSAN Note: I'm a new user to pastebin. I've mistakenly put two more pastebin entries: http://pastebin.freeswitch.org/13645 http://pastebin.freeswitch.org/13647 Cannot find a way to delete them. Please delete them, if not too much of a hassle, as they are useless. On Mon, Aug 16, 2010 at 10:09, Michael Collins wrote: > > > On Sun, Aug 15, 2010 at 5:34 PM, Nyamul Hassan wrote: > >> Hi, >> >> When a call hangs up normally, like either of the user kept the phone >> down, then there are no issues. >> >> However, when the call is hung up through a transfer, like mod_nibblebill >> does when "low_balance" or "no_balance" is reached, then the b-leg cdr.xml >> file gets prefixed with "a_". >> >> Is this a desired feature? Or a bug? >> > Step us through the entire call flow. Also, pastebin an example set of > files (a and b legs) for working vs. not working calls. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/bae10aa4/attachment.html From gmaruzz at celliax.org Mon Aug 16 09:31:00 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 16 Aug 2010 18:31:00 +0200 Subject: [Freeswitch-users] mod_skypopen (skype endpoint) changes, please test (REPOST) Message-ID: Just in case you missed the weekend mailz: ======= Hi FreeSWITCHers, I've made some long due modifications to mod_skypopen, that maybe introduced bugs. Please test with the latest git and report any *new* problem (ok, old problems too), here in the mailing list, or - much better - in the Jira. ========= commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 Author: Giovanni Maruzzelli Date: Fri Aug 13 16:19:20 2010 -0500 skypopen: now answer a call only when directed to do it (before was trying to answer any incoming call). Lot of changes to a messy part, so maybe some problem will come out... Signed-off-by: Giovanni Maruzzelli =========== Thank you to all, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From ritzalam at gmail.com Mon Aug 16 10:20:22 2010 From: ritzalam at gmail.com (Richard Alam) Date: Mon, 16 Aug 2010 13:20:22 -0400 Subject: [Freeswitch-users] Conference and Speex Wideband audio quality problem In-Reply-To: References: Message-ID: On Sun, Aug 15, 2010 at 7:45 PM, Michael Jerris wrote: > Stay tuned for mod_rtmp, coming soon to a freeswitch near you. > Is there a link where I can follow the progress on this module? We are be very interested in helping test this out. Thanks. Richard > Mike > > On Jul 30, 2010, at 5:47 PM, Richard Alam wrote: > >> OK...figured out what the problem. >> >> Flash Player by default sends 2 frames per packet with 20ms of audio >> (320 samples) per frame. FS expects only a frame every packet. >> So setting mic.framesPerPacket = 1; in the flash client worked. >> >> Here's the complete Actionscript setting in case somebody in the >> future wants to work with Flash. >> >> ? ? ? ? ? ? ? private function setupMicrophone():void { >> ? ? ? ? ? ? ? ? ? ? ? mic.setUseEchoSuppression(true); >> ? ? ? ? ? ? ? ? ? ? ? mic.setLoopBack(false); >> ? ? ? ? ? ? ? ? ? ? ? mic.setSilenceLevel(0,20000); >> ? ? ? ? ? ? ? ? ? ? ? mic.codec = SoundCodec.SPEEX; >> ? ? ? ? ? ? ? ? ? ? ? mic.gain = 60; >> ? ? ? ? ? ? ? ? ? ? ? mic.framesPerPacket = 1; >> ? ? ? ? ? ? ? ? ? ? ? mic.rate = 16; // use 8 for Nelly >> ? ? ? ? ? ? ? ? ? ? ? LogUtil.debug("codec=SPEEX,gain=60,encodeQuality=10,framesPerPacket=2,rate=16"); >> ? ? ? ? ? ? ? } >> >> Richard > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton From d at d-man.org Mon Aug 16 10:48:03 2010 From: d at d-man.org (Darren Schreiber) Date: Mon, 16 Aug 2010 10:48:03 -0700 Subject: [Freeswitch-users] SF Telephony Meetup Tonight Message-ID: <059A533B-57B1-451A-87DA-BAB673CDCD96@d-man.org> Hi folks, Sorry this is so last minute. If you'd like to join, I'm hosting a FreeSWITCH install-fest as part of the SF Telephony Meetup Group run by Zhao Lu. It's tonight in downtown San Francisco. If you're in town, come on by and install some FreeSWITCH goodies :-) Details are here: http://www.meetup.com/sftelephony/calendar/14161219/ There's no charge or anything for this event - it's just a general phone-geek meeting session. Thanks, Darren Schreiber -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/ea07aab1/attachment.html From msc at freeswitch.org Mon Aug 16 11:02:57 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Aug 2010 11:02:57 -0700 Subject: [Freeswitch-users] Did FreeSWITCH modules provides Vicidials features? In-Reply-To: References: Message-ID: On Mon, Aug 16, 2010 at 3:16 AM, Patrick Hsieh wrote: > Hello, > I am a new user of FreeSWITCH. > I have installed the Asterisk and Vicidial applications in my notebook. > > Q1: > I would like to know are there any similar features which already be > implemented by FreeSWITCH itself? > Not directly. FreeSWITCH has all the hooks and tools, like proverbial Lego bricks. "Some assembly required." :) > Q2: > Which will be a efficient way to have the features in FreeSWITCH? > Rewrite all features in FreeSWITCH or create a plug-in modules to Vicidial > directly. > Matt says making Vicidial work with FS will be a big challenge. It's probably better to build from the ground up a new solution. > Q3: > Did FreeSWITCH already support the ACD and Predictive Dialing features? > Which modules releated to the features? > Moc has started writing mod_callcenter which handles the ACD side of things and will (hopefully) have some outbound dialing capabilities. Check with him in the IRC channel. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/288a8fcb/attachment-0001.html From Victor at isptelecom.net Mon Aug 16 12:20:55 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Mon, 16 Aug 2010 15:20:55 -0400 Subject: [Freeswitch-users] NAT ACL and security In-Reply-To: References: Message-ID: <4C698F97.5020505@isptelecom.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/9299f21e/attachment.html From msc at freeswitch.org Mon Aug 16 12:27:12 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Aug 2010 12:27:12 -0700 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> Message-ID: On Mon, Aug 16, 2010 at 5:56 AM, Dennis wrote: > phillip, it seems that we have somehow the same problem, but we have > an additional strange behavior. > > we used wireshark on the incoming network-card and on the outgoing > side. there we see two strange things, we can not figure out: > > 1.) on the incoming side, we receive the in-band tone from the cirpack > with a tiny gap, but fs recognizes them as 2 tones. > in wireshark it look something like this: | |||||||| > as i said, fs sees this as two tones. > > 2.) fs sends the tone above to the outgoing side as two tones, but > seems to cut the first ms of the second part. > in wireshark it looks something like this: | ||| > > > i feel, that we have to find out the following: > > 1.) why does the tone already have a gap, when we receive it? > That would probably be better answered by looking at the sending side. I would capture the packets leaving the sending side to see what they look like. > > 2.) why does fs not leave the tone untouched? something happens with > the tone, while passing fs. is there a setting to avoid this? > 3.) why is there no easy way to completely delete dtmf tones, if it is > possible with vuvuzela-noise? ;-) > I'll defer to those more knowledgeable than I on these other two questions... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/6c7df591/attachment.html From brian at freeswitch.org Mon Aug 16 12:31:09 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Aug 2010 14:31:09 -0500 Subject: [Freeswitch-users] NAT ACL and security In-Reply-To: <4C698F97.5020505@isptelecom.net> References: <4C698F97.5020505@isptelecom.net> Message-ID: <368C3633-0829-4E84-84DE-9D73265E7068@freeswitch.org> You're treating everything as if it were nat.... including public addresses... /b On Aug 16, 2010, at 2:20 PM, Victor Chukalovskiy wrote: > I'm using > > > in my SIP profile in order to make Freeswitch ping every phone registered to it. > This works well for keeping phones on remote LANs reachable. > > My_nat ACL is defined as following: > > > That is, it allows everybody. > > Question: am I making my system insecure by doing so? > I believe "No" since ACL list "my_nat" is only used by appl-nat-acl parameter, > but I don't know FreeSWITCH well enough to grantee that nothing else is affected. > E.g. does anything else change if phone is considered NATed / non-NATed? > > Regards, > Victor From msc at freeswitch.org Mon Aug 16 12:36:46 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Aug 2010 12:36:46 -0700 Subject: [Freeswitch-users] playing a message while bridging In-Reply-To: References: <55192CF9-A167-4BC7-B282-B794BAEAA690@gmail.com> Message-ID: On Mon, Aug 16, 2010 at 2:15 AM, babak yakhchali wrote: > tried this: > > > > > > > > still my message and early media are mixing together :( > How about this: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/a5edb075/attachment.html From andrew at hijacked.us Mon Aug 16 16:01:15 2010 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 16 Aug 2010 19:01:15 -0400 Subject: [Freeswitch-users] Did FreeSWITCH modules provides Vicidials features? In-Reply-To: References: Message-ID: <20100816230115.GD18924@hijacked.us> On Mon, Aug 16, 2010 at 12:16:00PM +0200, Patrick Hsieh wrote: Not to toot my own horn, but the only solution that I'm aware of that's open source is my own project; OpenACD. FreeSWITCH has mod_fifo and mod_distributor, but that only routes calls, it doesn't provide the rest of what you're looking for (the CudaTel builds on that and provides a web interface, but I'm not sure that's enough). However, OpenACD doesn't fully meet your feature requirements, I'll detail what is present/missing below. > MAJOR VICIDIAL FEATURES: > > - Inbound, Outbound and Blended call handling Agents can take inbound calls and make outbound calls (tagged to a brand/client). I don't know what blended is (agents doing both?). > - Outbound agent-controlled, broadcast and predictive dialing Outbound is only agent controlled right now. > - Full USA FTC-compliance capability No. > - Web-based agent and administrative interfaces Yes, the entire system is configured from the web UI, and agents have a web UI where they can make outbound calls, manage their release state, do dynamic wrapup, etc. > - Ability to have agents operate remotely Yes. > - Integrated call recording Yes. > - Three-Way calling within the agent application No, but it wouldn't be hard (its planned at some point). > - Scheduled Callbacks: Agent-Only and Anyone Not yet but, again, its planned (and not very hard). > - Web-configurable IVRs and Voicemail boxes No, but sipXecs is planning to integrate OpenACD and provide a comprehensive admin UI for configuring dialplans AND the ACD, I believe. > - Scalable to hundreds of seats Yes, although I'm still working on the scalability. > - Ability to use standard Telco lines and VOIP trunks Yes. > - Open-Source AGPLv2 licensed, with no software licensing cost CPAL (trying to get it changed to the MPL), so pretty close. The real hangup is the predictive dialer. Personally I'm not terribly interested in enabling companies to robo dial my phone (because that happens enough already). I'm of the opinion that if you want to do predictive dialing, you should pay to play. Andrew From chris.veazey at gmail.com Mon Aug 16 20:13:01 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Mon, 16 Aug 2010 22:13:01 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: References: <4508448D788749F6B25886D626C17C28@left><563B3EF6EAFE41BC8506DDA53B45F7F8@left><396D3026B6034431B93A6B3DC0803461@left><1281542398575-5412733.post@n2.nabble.com><55D75CA26DF8494B9FD88112BEAE8211@left><1281554134702-5413561.post@n2.nabble.com> Message-ID: <1D954802FAF148468C766D9636EF95F1@left> After this SIP debugging, I can only assume FS cannot correctly process the maddr. The initial INVITE sent out results in a 302 from the redirect server with a contact of: Contact:;q=0.5;ton=PUBLIC;ct=NIL;cat=OTHER FS immediately sends out a New Invite back to the redirect server using the contact info of the 302 in the Req URI. send 1145 bytes to udp/[10.11.0.20]:5060 at 03:04:36.240400: ------------------------------------------------------------------------ INVITE sip:9995552000000 at sip.blinkmind.net SIP/2.0 Via: SIP/2.0/UDP 10.11.0.80:5080;rport;branch=z9hG4bKS42Dt0Qj0StQm Max-Forwards: 68 From: "9995551000000" ;tag=S2XD2mKva2vHQ To: Call-ID: 023221b8-244f-122e-3fa7-0018512d505c CSeq: 671074 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 287 X-FS-Support: update_display Remote-Party-ID: "9995551000000" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1281981574 1281981575 IN IP4 10.11.0.80 s=FreeSWITCH c=IN IP4 10.11.0.80 t=0 0 m=audio 32702 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ recv 449 bytes from udp/[10.11.0.20]:32774 at 03:04:36.243214: ------------------------------------------------------------------------ SIP/2.0 302 Moved temporarily Via:SIP/2.0/UDP 10.11.0.80:5080;branch=z9hG4bKS42Dt0Qj0StQm;rport From:"9995551000000";tag=S2XD2mKva2vHQ To:;tag=1852618065-1282014276242 Call-ID:023221b8-244f-122e-3fa7-0018512d505c CSeq:671074 INVITE Contact:;q=0.5;ton=PUBLIC;ct=NIL;cat=OTHER Content-Length:0 ------------------------------------------------------------------------ send 372 bytes to udp/[10.11.0.20]:5060 at 03:04:36.243414: ------------------------------------------------------------------------ ACK sip:9995552000000 at sip.blinkmind.net SIP/2.0 Via: SIP/2.0/UDP 10.11.0.80:5080;rport;branch=z9hG4bKS42Dt0Qj0StQm Max-Forwards: 68 From: "9995551000000" ;tag=S2XD2mKva2vHQ To: ;tag=1852618065-1282014276242 Call-ID: 023221b8-244f-122e-3fa7-0018512d505c CSeq: 671074 ACK Content-Length: 0 ------------------------------------------------------------------------ send 1184 bytes to udp/[10.11.0.20]:5060 at 03:04:36.243999: ------------------------------------------------------------------------ INVITE sip:9995552000000 at sip.blinkmind.net:5060;transport=udp;maddr=10.11.0.81 SIP/2.0 Via: SIP/2.0/UDP 10.11.0.80:5080;rport;branch=z9hG4bKtDv6UU8NX2gag Max-Forwards: 68 From: "9995551000000" ;tag=tBQ63F4Z7aK4j To: Call-ID: 0232b20c-244f-122e-3fa7-0018512d505c CSeq: 671074 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 290 X-FS-Support: update_display Remote-Party-ID: "9995551000000" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1281981574 1281981576 IN IP4 10.11.0.80 s=FreeSWITCH c=IN IP4 10.11.0.80 t=0 0 m=audio 32702 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 448 bytes from udp/[10.11.0.20]:32774 at 03:04:36.246197: ------------------------------------------------------------------------ SIP/2.0 302 Moved temporarily Via:SIP/2.0/UDP 10.11.0.80:5080;branch=z9hG4bKtDv6UU8NX2gag;rport From:"9995551000000";tag=tBQ63F4Z7aK4j To:;tag=343356216-1282014276245 Call-ID:0232b20c-244f-122e-3fa7-0018512d505c CSeq:671074 INVITE Contact:;q=0.5;ton=PUBLIC;ct=NIL;cat=OTHER Content-Length:0 ------------------------------------------------------------------------ send 407 bytes to udp/[10.11.0.20]:5060 at 03:04:36.246339: ------------------------------------------------------------------------ ACK sip:9995552000000 at sip.blinkmind.net:5060;transport=udp;maddr=10.11.0.81 SIP/2.0 Via: SIP/2.0/UDP 10.11.0.80:5080;rport;branch=z9hG4bKtDv6UU8NX2gag Max-Forwards: 68 From: "9995551000000" ;tag=tBQ63F4Z7aK4j To: ;tag=343356216-1282014276245 Call-ID: 0232b20c-244f-122e-3fa7-0018512d505c CSeq: 671074 ACK Content-Length: 0 _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Wednesday, August 11, 2010 4:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS processing 302 AFAIK 302 reenters the dialplan. If the destination number is unchanged, it will execute the same commands after the redirect as it did before the redirect, which would likely cause it to continously bridge, redirect, bridge, redirect.... That could explain what you're experiencing. How do you want to handle the 302? It's advisable to be careful what to do with a 302 which is why it reenters the dialplan. You can try dialing a $0.01 route but have it redirect you to a $1.00 route leaving you sending the call out on a route costing you $1.00 but charging your customer $0.01, or if you do bill them correctly leaving them with a bill of $1.00 when they expected only $0.01. It's possible to handle a 302 manually by sending them to a dialplan context specifically for it, which keeps the redirect logic away from your main dialplan. See http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect -Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100816/96ccc4ba/attachment-0001.html From neilp at cs.stanford.edu Mon Aug 16 22:41:17 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Tue, 17 Aug 2010 11:11:17 +0530 Subject: [Freeswitch-users] outbound call with hunting number as the originating caller In-Reply-To: References: Message-ID: That did it! Thanks Michael! -Neil On Fri, Aug 13, 2010 at 12:30 AM, Michael Collins wrote: > > > On Thu, Aug 12, 2010 at 1:45 AM, Neil Patel wrote: > >> Hi All, >> >> I have an app running on FS making outbound calls over a PRI line. The >> calls all caller-id at the endpoints with the PRI's pilot number. Say I >> wanted to change it so that the endpoint receives the call from one of the >> block of hunting numbers that my line comes with (instead of the >> primary/pilot number), how do I originate a call to do that? Here's the >> command on fs_cli that I'm trying, that doesn't seem to work: >> >> > originate >> {destination_number=XXXXXXXXXX}openzap/smg_prid/a/YYYYYYYYYY at g2 &echo >> >> > Neil, use "origination_caller_id_number=xxx" like this: > > > originate > {origination_caller_id_number=1234567890,destination_number=XXXXXXXXXX}openzap/smg_prid/a/YYYYYYYYYY at g2&echo > > NOTE: The telco must support custom sending of "calling party number" > sometimes called CPN. It won't hurt to try it, though, so give it a whirl > and let us know. > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/3607b558/attachment.html From babak.freeswitch at gmail.com Tue Aug 17 01:25:25 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Tue, 17 Aug 2010 12:55:25 +0430 Subject: [Freeswitch-users] playing a message while bridging In-Reply-To: References: <55192CF9-A167-4BC7-B282-B794BAEAA690@gmail.com> Message-ID: thanx Michael it's working when I'm using cisco 7941 ip phones but when I'm calling using 3cx soft phone early media is mixed with my message (both of them are heard). may be some thing is wrong with my 3cx soft phone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/5ce14b61/attachment.html From erkan at speedingtrade.com Tue Aug 17 03:03:02 2010 From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=) Date: Tue, 17 Aug 2010 13:03:02 +0300 Subject: [Freeswitch-users] SPA8000 References: <81C2CEF80046FB4F863A60D4347DD33A0C559E@server1.st.local><81C2CEF80046FB4F863A60D4347DD33A0C55A1@server1.st.local><201008101837.38236.errotan@elder.hu> Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C55AD@server1.st.local> You have right. It works. The problem is the modem. I have select NAT = Yes in SPA8000 but if I change the installed modem by the customer, so all things works. It's interesting, the modem is a cheap modem and with that Airties modem doesn't work. But with Thomson and Zyxel that I tested, they work without problems. The confused thing is. If I try it with a softphone they work without problem. But the SPA8000 can't registered to the freeswitch server with the Airties Modem. The connection to other servers works. Thank you very much. Kind regards Erkan -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, August 16, 2010 5:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SPA8000 I can confirm we have working SPA8800 with FreeSWITCH. On Aug 10, 2010, at 12:37 PM, Pusk?s Zsolt wrote: > Hi. > > SPA8000 works perfectly for me. > Do you have NAT involved ? If so have you set NAT mapping and NAT keep alive > on ? > Please tell how you configured the SPA8000 because without info we can't help. > > > 2010. augusztus 10. 13:00:49 d?tummal Erkan ?nl? az al?bbiakat ?rta: >> I think the problem is in FreeSwitch. >> >> If I use other voip server that works. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jeff at jefflenk.com Tue Aug 17 07:11:55 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 17 Aug 2010 07:11:55 -0700 (PDT) Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <1D954802FAF148468C766D9636EF95F1@left> References: <563B3EF6EAFE41BC8506DDA53B45F7F8@left> <396D3026B6034431B93A6B3DC0803461@left> <1281542398575-5412733.post@n2.nabble.com> <55D75CA26DF8494B9FD88112BEAE8211@left> <1281554134702-5413561.post@n2.nabble.com> <1D954802FAF148468C766D9636EF95F1@left> Message-ID: <1282054315191-5432276.post@n2.nabble.com> Chris are you still running with the line I suggested commented out - I assume yes. Are you setting any specific variables as Brian asked for above? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p5432276.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ktngl at yahoo.co.uk Tue Aug 17 08:11:02 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Tue, 17 Aug 2010 15:11:02 +0000 (GMT) Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: Message-ID: <342834.88194.qm@web29214.mail.ird.yahoo.com> I want to capture # key in playandgetdigits and copy it to the varible but "\\d+|\/#/" is not working. The varible does not contain the '#' key -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/8a727dec/attachment.html From brian at freeswitch.org Tue Aug 17 08:18:55 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Aug 2010 10:18:55 -0500 Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <342834.88194.qm@web29214.mail.ird.yahoo.com> References: <342834.88194.qm@web29214.mail.ird.yahoo.com> Message-ID: In the future please click NEW message and input the address for the list. You have Hijacked your own thread three times already... Please try not to do that as your emails can get lost in some people's mail readers that thread based on the headers in the email. Do you have # set as the terminator? /b On Aug 17, 2010, at 10:11 AM, Nigel Kent wrote: > I want to capture # key in playandgetdigits and copy it to the varible but "\\d+|\/#/" is not working. The varible does not contain the '#' key > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris.veazey at gmail.com Tue Aug 17 08:35:14 2010 From: chris.veazey at gmail.com (Chris Veazey) Date: Tue, 17 Aug 2010 10:35:14 -0500 Subject: [Freeswitch-users] FS processing 302 In-Reply-To: <1282054315191-5432276.post@n2.nabble.com> References: <563B3EF6EAFE41BC8506DDA53B45F7F8@left><396D3026B6034431B93A6B3DC0803461@left><1281542398575-5412733.post@n2.nabble.com><55D75CA26DF8494B9FD88112BEAE8211@left><1281554134702-5413561.post@n2.nabble.com><1D954802FAF148468C766D9636EF95F1@left> <1282054315191-5432276.post@n2.nabble.com> Message-ID: We are using a Git from last week with this commented out: Line 3998 of Sofia.c switch_channel_hangup(channel, SWITCH_CAUSE_REDIRECTION_TO_NEW_DESTINATION) How do I set the variables described in http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect? The concerning issue is that the source code doesn't seem to provide a variable for the maddr, just the host portion sip_redirect_contact_ Chris -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Tuesday, August 17, 2010 9:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS processing 302 Chris are you still running with the line I suggested commented out - I assume yes. Are you setting any specific variables as Brian asked for above? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p54 32276.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ktngl at yahoo.co.uk Tue Aug 17 08:38:09 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Tue, 17 Aug 2010 15:38:09 +0000 (GMT) Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: Message-ID: <984277.29644.qm@web29210.mail.ird.yahoo.com> Sorry about that. Thanks for the tip Yes the terminator is set as # but I want to capture a seperate # aswell. In effect the user will enter ##. The second instance is the terminator and I am checking the dtmf varible for the which should have the the last key as #. But it appears # is not being copied to the dtmf varible. So is my regex pattern correct or is it some other issue --- On Tue, 17/8/10, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' To: "FreeSWITCH Users Help" Date: Tuesday, 17 August, 2010, 15:18 In the future please click NEW message and input the address for the list.? You have Hijacked your own thread three times already... Please try not to do that as your emails can get lost in some people's mail readers that thread based on the headers in the email. Do you have # set as the terminator? /b On Aug 17, 2010, at 10:11 AM, Nigel Kent wrote: > I want to capture # key in playandgetdigits and copy it to the varible but "\\d+|\/#/" is not working. The varible does not contain the '#' key > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/2f7b7c33/attachment-0001.html From david.ponzone at ipeva.fr Tue Aug 17 08:59:05 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Tue, 17 Aug 2010 17:59:05 +0200 Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <984277.29644.qm@web29210.mail.ird.yahoo.com> References: <984277.29644.qm@web29210.mail.ird.yahoo.com> Message-ID: <3C8C28F1-41D9-4E34-A698-A635CA2E9350@ipeva.fr> Nigel, how do you want to capture a # if # is the terminator ? :) if you want to capture #, you need to use * as the terminator. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 17/08/2010 ? 17:38, Nigel Kent a ?crit : > Sorry about that. Thanks for the tip > > Yes the terminator is set as # but I want to capture a seperate # > aswell. In effect the user will enter ##. The second instance is the > terminator and I am checking the dtmf varible for the which should > have the the last key as #. But it appears # is not being copied to > the dtmf varible. > > So is my regex pattern correct or is it some other issue > > --- On Tue, 17/8/10, Brian West wrote: > > From: Brian West > Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' > To: "FreeSWITCH Users Help" > Date: Tuesday, 17 August, 2010, 15:18 > > In the future please click NEW message and input the address for the > list. You have Hijacked your own thread three times already... > Please try not to do that as your emails can get lost in some > people's mail readers that thread based on the headers in the email. > > Do you have # set as the terminator? > > /b > > > On Aug 17, 2010, at 10:11 AM, Nigel Kent wrote: > > > I want to capture # key in playandgetdigits and copy it to the > varible but "\\d+|\/#/" is not working. The varible does not contain > the '#' key > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/4f4d4609/attachment.html From ktngl at yahoo.co.uk Tue Aug 17 09:19:33 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Tue, 17 Aug 2010 16:19:33 +0000 (GMT) Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <3C8C28F1-41D9-4E34-A698-A635CA2E9350@ipeva.fr> Message-ID: <263402.10260.qm@web29206.mail.ird.yahoo.com> I see, how clumsy of me. I have been focusing too much on capturing two #'s. Still is this not possible some way. I want the two ##'s because it is already the common usage in what I want to do so would prefer to keep the same --- On Tue, 17/8/10, David Ponzone wrote: From: David Ponzone Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' To: "FreeSWITCH Users Help" Date: Tuesday, 17 August, 2010, 15:59 Nigel, how do you want to capture a # if # is the terminator ? :)if you want to capture #, you need to use * as the terminator. David Ponzone ?Direction Techniqueemail: david.ponzone at ipeva.frtel: ? ? ?01 74 03 18 97gsm: ? 06 66 98 76 34 Service Client?IPevatel: ? ? ?0811 46 26 26www.ipeva.fr? -? ?www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 17/08/2010 ? 17:38, Nigel Kent a ?crit : Sorry about that. Thanks for the tip Yes the terminator is set as # but I want to capture a seperate # aswell. In effect the user will enter ##. The second instance is the terminator and I am checking the dtmf varible for the which should have the the last key as #. But it appears # is not being copied to the dtmf varible. So is my regex pattern correct or is it some other issue --- On Tue, 17/8/10, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' To: "FreeSWITCH Users Help" Date: Tuesday, 17 August, 2010, 15:18 In the future please click NEW message and input the address for the list.? You have Hijacked your own thread three times already... Please try not to do that as your emails can get lost in some people's mail readers that thread based on the headers in the email. Do you have # set as the terminator? /b On Aug 17, 2010, at 10:11 AM, Nigel Kent wrote: > I want to capture # key in playandgetdigits and copy it to the varible but "\\d+|\/#/" is not working. The varible does not contain the '#' key > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/c78522ea/attachment-0001.html From brian at freeswitch.org Tue Aug 17 09:27:42 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Aug 2010 11:27:42 -0500 Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <263402.10260.qm@web29206.mail.ird.yahoo.com> References: <263402.10260.qm@web29206.mail.ird.yahoo.com> Message-ID: <4792A7AF-3F25-44A5-86CD-21BC7B6238A7@freeswitch.org> I have to say I have never seen a ## be used in any IVR ... Has anyone else? /b On Aug 17, 2010, at 11:19 AM, Nigel Kent wrote: > I want the two ##'s because it is already the common usage in what I want to do so would prefer to keep the same From david.ponzone at ipeva.fr Tue Aug 17 09:36:32 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Tue, 17 Aug 2010 18:36:32 +0200 Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <263402.10260.qm@web29206.mail.ird.yahoo.com> References: <263402.10260.qm@web29206.mail.ird.yahoo.com> Message-ID: <422DA915-A215-477B-951C-2399C6C93726@ipeva.fr> I dont know if it's possible in FS, but did you think about not defining a terminator, defining max-length to 1, and then getting the DTMF one by one, concatenating them in a variable where you would look for ## ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 17/08/2010 ? 18:19, Nigel Kent a ?crit : > I see, how clumsy of me. > > I have been focusing too much on capturing two #'s. Still is this > not possible some way. > > I want the two ##'s because it is already the common usage in what I > want to do so would prefer to keep the same > > > > --- On Tue, 17/8/10, David Ponzone wrote: > > From: David Ponzone > Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' > To: "FreeSWITCH Users Help" > Date: Tuesday, 17 August, 2010, 15:59 > > Nigel, how do you want to capture a # if # is the terminator ? :) > if you want to capture #, you need to use * as the terminator. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 17/08/2010 ? 17:38, Nigel Kent a ?crit : > >> Sorry about that. Thanks for the tip >> >> Yes the terminator is set as # but I want to capture a seperate # >> aswell. In effect the user will enter ##. The second instance is >> the terminator and I am checking the dtmf varible for the which >> should have the the last key as #. But it appears # is not being >> copied to the dtmf varible. >> >> So is my regex pattern correct or is it some other issue >> >> --- On Tue, 17/8/10, Brian West wrote: >> >> From: Brian West >> Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' >> To: "FreeSWITCH Users Help" >> Date: Tuesday, 17 August, 2010, 15:18 >> >> In the future please click NEW message and input the address for >> the list. You have Hijacked your own thread three times already... >> Please try not to do that as your emails can get lost in some >> people's mail readers that thread based on the headers in the email. >> >> Do you have # set as the terminator? >> >> /b >> >> >> On Aug 17, 2010, at 10:11 AM, Nigel Kent wrote: >> >> > I want to capture # key in playandgetdigits and copy it to the >> varible but "\\d+|\/#/" is not working. The varible does not >> contain the '#' key >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/eb2d87d1/attachment.html From msc at freeswitch.org Tue Aug 17 10:41:58 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Aug 2010 10:41:58 -0700 Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <263402.10260.qm@web29206.mail.ird.yahoo.com> References: <3C8C28F1-41D9-4E34-A698-A635CA2E9350@ipeva.fr> <263402.10260.qm@web29206.mail.ird.yahoo.com> Message-ID: On Tue, Aug 17, 2010 at 9:19 AM, Nigel Kent wrote: > I see, how clumsy of me. > > I have been focusing too much on capturing two #'s. Still is this not > possible some way. > > I want the two ##'s because it is already the common usage in what I want > to do so would prefer to keep the same > Well, you can capture everything with a regex of ".*" and then just rely on the timeout. It's the only way to do it without a lot of hassle. If you're into hassle then you have other options: Capture digits one at a time and do fun/crazy logic to see if you have what you need Hack playandgetdigits to allow ## to be a terminator. I'd start with eliminating the terminator characters first and testing it out. You can shorten your timeout value so the user isn't waiting too long after dialing ##. Give it a test drive and see if it will work. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/a6e912f4/attachment-0001.html From ktngl at yahoo.co.uk Tue Aug 17 10:45:51 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Tue, 17 Aug 2010 17:45:51 +0000 (GMT) Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <422DA915-A215-477B-951C-2399C6C93726@ipeva.fr> Message-ID: <786917.85201.qm@web29219.mail.ird.yahoo.com> Are you suggesting using something other then playandgetdigits --- On Tue, 17/8/10, David Ponzone wrote: From: David Ponzone Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' To: "FreeSWITCH Users Help" Date: Tuesday, 17 August, 2010, 16:36 I dont know if it's possible in FS, but did you think about not defining a terminator, defining max-length to 1, and then getting the DTMF one by one, concatenating them in a variable where you would look for ## ? David Ponzone ?Direction Techniqueemail: david.ponzone at ipeva.frtel: ? ? ?01 74 03 18 97gsm: ? 06 66 98 76 34 Service Client?IPevatel: ? ? ?0811 46 26 26www.ipeva.fr? -? ?www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 17/08/2010 ? 18:19, Nigel Kent a ?crit : I see, how clumsy of me. I have been focusing too much on capturing two #'s. Still is this not possible some way. I want the two ##'s because it is already the common usage in what I want to do so would prefer to keep the same --- On Tue, 17/8/10, David Ponzone wrote: From: David Ponzone Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' To: "FreeSWITCH Users Help" Date: Tuesday, 17 August, 2010, 15:59 Nigel, how do you want to capture a # if # is the terminator ? :)if you want to capture #, you need to use * as the terminator. David Ponzone ?Direction Techniqueemail: david.ponzone at ipeva.frtel: ? ? ?01 74 03 18 97gsm: ? 06 66 98 76 34 Service Client?IPevatel: ? ? ?0811 46 26 26www.ipeva.fr? -? ?www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 17/08/2010 ? 17:38, Nigel Kent a ?crit : Sorry about that. Thanks for the tip Yes the terminator is set as # but I want to capture a seperate # aswell. In effect the user will enter ##. The second instance is the terminator and I am checking the dtmf varible for the which should have the the last key as #. But it appears # is not being copied to the dtmf varible. So is my regex pattern correct or is it some other issue --- On Tue, 17/8/10, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' To: "FreeSWITCH Users Help" Date: Tuesday, 17 August, 2010, 15:18 In the future please click NEW message and input the address for the list.? You have Hijacked your own thread three times already... Please try not to do that as your emails can get lost in some people's mail readers that thread based on the headers in the email. Do you have # set as the terminator? /b On Aug 17, 2010, at 10:11 AM, Nigel Kent wrote: > I want to capture # key in playandgetdigits and copy it to the varible but "\\d+|\/#/" is not working. The varible does not contain the '#' key > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/b64b278d/attachment.html From anthony.minessale at gmail.com Tue Aug 17 11:14:13 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Aug 2010 13:14:13 -0500 Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <786917.85201.qm@web29219.mail.ird.yahoo.com> References: <422DA915-A215-477B-951C-2399C6C93726@ipeva.fr> <786917.85201.qm@web29219.mail.ird.yahoo.com> Message-ID: you could set no terminator instead and just look for the ## with your regex. It's uncommon, so it's only not possible because there has been no demand for it at all. On Tue, Aug 17, 2010 at 12:45 PM, Nigel Kent wrote: > Are you suggesting using something other then playandgetdigits > > > --- On *Tue, 17/8/10, David Ponzone * wrote: > > > From: David Ponzone > Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' > To: "FreeSWITCH Users Help" > Date: Tuesday, 17 August, 2010, 16:36 > > > I dont know if it's possible in FS, but did you think about not defining a > terminator, defining max-length to 1, and then getting the DTMF one by one, > concatenating them in a variable where you would look for ## ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 17/08/2010 ? 18:19, Nigel Kent a ?crit : > > I see, how clumsy of me. > > I have been focusing too much on capturing two #'s. Still is this not > possible some way. > > I want the two ##'s because it is already the common usage in what I want > to do so would prefer to keep the same > > > > --- On *Tue, 17/8/10, David Ponzone > >* wrote: > > > From: David Ponzone > > > Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' > To: "FreeSWITCH Users Help" > > > Date: Tuesday, 17 August, 2010, 15:59 > > Nigel, how do you want to capture a # if # is the terminator ? :) > if you want to capture #, you need to use * as the terminator. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 17/08/2010 ? 17:38, Nigel Kent a ?crit : > > Sorry about that. Thanks for the tip > > Yes the terminator is set as # but I want to capture a seperate # aswell. > In effect the user will enter ##. The second instance is the terminator and > I am checking the dtmf varible for the which should have the the last key as > #. But it appears # is not being copied to the dtmf varible. > > So is my regex pattern correct or is it some other issue > > --- On *Tue, 17/8/10, Brian West * wrote: > > > From: Brian West > Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' > To: "FreeSWITCH Users Help" > Date: Tuesday, 17 August, 2010, 15:18 > > In the future please click NEW message and input the address for the list. > You have Hijacked your own thread three times already... Please try not to > do that as your emails can get lost in some people's mail readers that > thread based on the headers in the email. > > Do you have # set as the terminator? > > /b > > > On Aug 17, 2010, at 10:11 AM, Nigel Kent wrote: > > > I want to capture # key in playandgetdigits and copy it to the varible > but "\\d+|\/#/" is not working. The varible does not contain the '#' key > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/4c3fc8a2/attachment-0001.html From sprice at gmail.com Tue Aug 17 11:25:30 2010 From: sprice at gmail.com (SP) Date: Tue, 17 Aug 2010 13:25:30 -0500 Subject: [Freeswitch-users] enable passthrough of "Privacy: id" header in sip In-Reply-To: References: <1a5cbae9-5bd2-4fd1-d1d3-88fae8b6bd15@me.com> <201007221436.42762.sos@sokhapkin.dyndns.org> Message-ID: http://tools.ietf.org/html/draft-ietf-sip-privacy-00 5.2 Anonymity Header Field Definition blah blah blah..."If privacy is requested, it MUST be one or more of "full", "uri", "name", or "ipaddr"." On Sun, Aug 15, 2010 at 17:52, Michael Jerris wrote: > is there a specification which defines this "uri" value? > > Mike > > On Jul 22, 2010, at 2:36 PM, Sergey Okhapkin wrote: > >> FS doesn't recognize "uri" value of privacy tag, the recognized values are >> "yes", "full", "name" and "number". Any other value is interpreted as "off". >> See sofia.c, lines around 6660. >> >> On Thursday 22 July 2010, mike.burlingame wrote: >>> Ok so I am still kinda lost in trying to figure this one out here are the >>> two headers that I am looking at - basically someone sends an invite to FS >>> that ask's FS via RPID to hide the caller id info from downstream gateways >>> so the A-LEG is the invite going to FS - FS takes the invite and spits out >>> the B-LEG to go downstream HOWEVER FS does not copy the parameters >>> correctly as you can see in the initial invite A-LEG requested in the RPID >>> to be privacy=uri however on the B-LEG side FS set the RPID to privacy=off >>> in turn telling gateways downstream to display the CID info. >>> >>> A-LEG >>> Remote-Party-ID: ;party=calling;screen=yes;privacy=uri >>> >>> B-LEG >>> Remote-Party-ID: "NAME" >>> ;party=calling;screen=yes;privacy=off >>> >>> On Jul 22, 2010, at 09:35 AM, Michael Collins wrote: >>> >>> >>> >>> On Wed, Jul 21, 2010 at 4:18 PM, mike.burlingame >>> wrote: yeah the variable I need from the A-LEG is privacy=uri the B-LEG of >>> FS by default is putting privacy=off - so I would guess the variable that >>> needs to be exported would be the privacy= correct? >>> >>> So the next question I would have is how do I export that that from the >>> A-LEG to the B-LEG - I would have though FS would be Data in Data out so >>> no need to change the RPID request from the original A-LEG? Is the >>> variable simply named "privacy"? If so just use the export app before the >>> bridge: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From david.ponzone at ipeva.fr Tue Aug 17 11:44:14 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Tue, 17 Aug 2010 20:44:14 +0200 Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: <786917.85201.qm@web29219.mail.ird.yahoo.com> References: <786917.85201.qm@web29219.mail.ird.yahoo.com> Message-ID: Perhaps getDigits ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 17/08/2010 ? 19:45, Nigel Kent a ?crit : > Are you suggesting using something other then playandgetdigits > > --- On Tue, 17/8/10, David Ponzone wrote: > > From: David Ponzone > Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' > To: "FreeSWITCH Users Help" > Date: Tuesday, 17 August, 2010, 16:36 > > I dont know if it's possible in FS, but did you think about not > defining a terminator, defining max-length to 1, and then getting > the DTMF one by one, concatenating them in a variable where you > would look for ## ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 17/08/2010 ? 18:19, Nigel Kent a ?crit : > >> I see, how clumsy of me. >> >> I have been focusing too much on capturing two #'s. Still is this >> not possible some way. >> >> I want the two ##'s because it is already the common usage in what >> I want to do so would prefer to keep the same >> >> >> >> --- On Tue, 17/8/10, David Ponzone wrote: >> >> From: David Ponzone >> Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' >> To: "FreeSWITCH Users Help" >> Date: Tuesday, 17 August, 2010, 15:59 >> >> Nigel, how do you want to capture a # if # is the terminator ? :) >> if you want to capture #, you need to use * as the terminator. >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? l'intention exclusive de ses destinataires. Toute >> utilisation ou diffusion non autoris?e est interdite. Tout message >> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute >> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou >> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de >> le d?truire imm?diatement et d'avertir l'exp?diteur. >> >> >> >> >> Le 17/08/2010 ? 17:38, Nigel Kent a ?crit : >> >>> Sorry about that. Thanks for the tip >>> >>> Yes the terminator is set as # but I want to capture a seperate # >>> aswell. In effect the user will enter ##. The second instance is >>> the terminator and I am checking the dtmf varible for the which >>> should have the the last key as #. But it appears # is not being >>> copied to the dtmf varible. >>> >>> So is my regex pattern correct or is it some other issue >>> >>> --- On Tue, 17/8/10, Brian West wrote: >>> >>> From: Brian West >>> Subject: Re: [Freeswitch-users] playandgetdigits regex to get '#' >>> To: "FreeSWITCH Users Help" >>> Date: Tuesday, 17 August, 2010, 15:18 >>> >>> In the future please click NEW message and input the address for >>> the list. You have Hijacked your own thread three times >>> already... Please try not to do that as your emails can get lost >>> in some people's mail readers that thread based on the headers in >>> the email. >>> >>> Do you have # set as the terminator? >>> >>> /b >>> >>> >>> On Aug 17, 2010, at 10:11 AM, Nigel Kent wrote: >>> >>> > I want to capture # key in playandgetdigits and copy it to the >>> varible but "\\d+|\/#/" is not working. The varible does not >>> contain the '#' key >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> -----Inline Attachment Follows----- >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/c2aacdd7/attachment-0001.html From msc at freeswitch.org Tue Aug 17 12:45:44 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Aug 2010 12:45:44 -0700 Subject: [Freeswitch-users] playandgetdigits regex to get '#' In-Reply-To: References: <422DA915-A215-477B-951C-2399C6C93726@ipeva.fr> <786917.85201.qm@web29219.mail.ird.yahoo.com> Message-ID: On Tue, Aug 17, 2010 at 11:14 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you could set no terminator instead and just look for the ## with your > regex. > It's uncommon, so it's only not possible because there has been no demand > for it at all. > This is essentially what I was suggesting. Use playandgetdigits, have no terminators, and then have your regex capture the trailing ## characters. It will require a bit of logic on your part but I don't see why you couldn't do it. Just remember what I said about the timeout value because your users will have to wait a few seconds after pressing the ## before the system accepts that as the finished input. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/b4d5dffc/attachment.html From gavin.henry at gmail.com Tue Aug 17 15:08:46 2010 From: gavin.henry at gmail.com (Gavin Henry) Date: Tue, 17 Aug 2010 23:08:46 +0100 Subject: [Freeswitch-users] Howler - Gone? In-Reply-To: <87168847915b78750eb7c4def0258461@mail.gmail.com> References: <87168847915b78750eb7c4def0258461@mail.gmail.com> Message-ID: On 25 July 2010 18:48, Kristian Kielhofner wrote: > The only reason I ask is because I just bought a 480 for testing ;)... This one? http://www.voipon.co.uk/sangoma-d100480e-p-2566.html for testing??? -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From pjintheusa at gmail.com Tue Aug 17 15:11:17 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 17 Aug 2010 18:11:17 -0400 Subject: [Freeswitch-users] start_dtmf on BLeg Channel from the dialplan Message-ID: Hi there, Is there a way to execute start_dtmf on the B Leg, from the dialplan, before you bridge call? I am using group_confirm, so that a called user is bridged when they answer and press 1. With one of my carriers, DTMF is not working - pressing 1 does nothing. If I use group_confirm=exec and send the B leg to a script on answer, and execute "start_dtmf" on the b leg session in the script, all works as expected and pressing 1 bridges the call. I would perfer not to use a script if there is a way of achieving this via the dialplan. Any help appreciated. Thanks Pj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/7db0a710/attachment.html From mike at jerris.com Tue Aug 17 15:44:03 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 17 Aug 2010 18:44:03 -0400 Subject: [Freeswitch-users] enable passthrough of "Privacy: id" header in sip In-Reply-To: References: <1a5cbae9-5bd2-4fd1-d1d3-88fae8b6bd15@me.com> <201007221436.42762.sos@sokhapkin.dyndns.org> Message-ID: <2D0E16F0-C234-4C80-9174-5350C3EA7387@jerris.com> Expiration 5/31/01 An actual RFC as opposed to a 10 year old now expired draft? Mike On Aug 17, 2010, at 2:25 PM, SP wrote: > http://tools.ietf.org/html/draft-ietf-sip-privacy-00 > > 5.2 Anonymity Header Field Definition > blah blah blah..."If privacy is requested, it MUST be one or more of > "full", "uri", > "name", or "ipaddr"." > > > On Sun, Aug 15, 2010 at 17:52, Michael Jerris wrote: >> is there a specification which defines this "uri" value? >> >> Mike >> >> On Jul 22, 2010, at 2:36 PM, Sergey Okhapkin wrote: >> >>> FS doesn't recognize "uri" value of privacy tag, the recognized values are >>> "yes", "full", "name" and "number". Any other value is interpreted as "off". >>> See sofia.c, lines around 6660. >>> >>> On Thursday 22 July 2010, mike.burlingame wrote: >>>> Ok so I am still kinda lost in trying to figure this one out here are the >>>> two headers that I am looking at - basically someone sends an invite to FS >>>> that ask's FS via RPID to hide the caller id info from downstream gateways >>>> so the A-LEG is the invite going to FS - FS takes the invite and spits out >>>> the B-LEG to go downstream HOWEVER FS does not copy the parameters >>>> correctly as you can see in the initial invite A-LEG requested in the RPID >>>> to be privacy=uri however on the B-LEG side FS set the RPID to privacy=off >>>> in turn telling gateways downstream to display the CID info. >>>> >>>> A-LEG >>>> Remote-Party-ID: ;party=calling;screen=yes;privacy=uri >>>> >>>> B-LEG >>>> Remote-Party-ID: "NAME" >>>> ;party=calling;screen=yes;privacy=off >>>> >>>> On Jul 22, 2010, at 09:35 AM, Michael Collins wrote: >>>> >>>> >>>> >>>> On Wed, Jul 21, 2010 at 4:18 PM, mike.burlingame >>>> wrote: yeah the variable I need from the A-LEG is privacy=uri the B-LEG of >>>> FS by default is putting privacy=off - so I would guess the variable that >>>> needs to be exported would be the privacy= correct? >>>> >>>> So the next question I would have is how do I export that that from the >>>> A-LEG to the B-LEG - I would have though FS would be Data in Data out so >>>> no need to change the RPID request from the original A-LEG? Is the >>>> variable simply named "privacy"? If so just use the export app before the >>>> bridge: >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Shannon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/d10fe025/attachment.html From msc at freeswitch.org Tue Aug 17 16:17:08 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Aug 2010 16:17:08 -0700 Subject: [Freeswitch-users] start_dtmf on BLeg Channel from the dialplan In-Reply-To: References: Message-ID: Try this and see if it helps: http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_app -MC On Tue, Aug 17, 2010 at 3:11 PM, Phillip Jones wrote: > Hi there, > > Is there a way to execute start_dtmf on the B Leg, from the > dialplan, before you bridge call? > > > I am using group_confirm, so that a called user is bridged when they answer > and press 1. > > With one of my carriers, DTMF is not working - pressing 1 does nothing. > > If I use group_confirm=exec and send the B leg to a script on answer, and > execute "start_dtmf" on the b leg session in the script, all works as > expected and pressing 1 bridges the call. > > I would perfer not to use a script if there is a way of achieving this via > the dialplan. > > Any help appreciated. > > Thanks > > Pj > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/fd8647f9/attachment.html From xyangni at gmail.com Tue Aug 17 18:26:38 2010 From: xyangni at gmail.com (xuyan yang) Date: Wed, 18 Aug 2010 09:26:38 +0800 Subject: [Freeswitch-users] Is mod_java supported on windows? Message-ID: In the latest git I didn't find any VC project file for mod_java. From the makefile I found that swig is needed to conpile this. And swig does have windows version. Is it possible to compile mod_java for windows from latest git tree? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/bf66a96e/attachment-0001.html From brian at freeswitch.org Tue Aug 17 18:37:06 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Aug 2010 20:37:06 -0500 Subject: [Freeswitch-users] Is mod_java supported on windows? In-Reply-To: References: Message-ID: swig is NOT required to compile it... but nobody has tried it on windows. /b On Aug 17, 2010, at 8:26 PM, xuyan yang wrote: > In the latest git I didn't find any VC project file for mod_java. From the makefile I found that swig is needed to conpile this. And swig does have windows version. Is it possible to compile mod_java for windows from latest git tree? _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at microcomaustralia.com.au Tue Aug 17 18:58:09 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Wed, 18 Aug 2010 11:58:09 +1000 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: On 7 August 2010 18:35, Brian May wrote: >> As for how to do sqlite data recovery, try googling. e.g. >> http://www.mail-archive.com/sqlite-users at sqlite.org/msg17538.html > > Thanks for the reference. I tried looking but obviously wasn't looking > for the right search term. Curiously I had the same problem reoccur. Unfortunately freeswitch appears to work until some time after the initial problem, makes it hard to find out what happened. The power has failed recently though, maybe connected? [ etc ] 2010-08-17 04:30:40.728489 [CRIT] switch_core_sqldb.c:938 SQL thread unable to commit transaction, records lost! 2010-08-17 04:31:00.653705 [ERR] switch_core_sqldb.c:448 SQL ERR [database disk image is malformed] [ etc ] huey:/opt/freeswitch/db# ls -l core2.db -rw-r--r-- 1 freeswitch daemon 1502208 2010-08-15 17:11 core2.db huey:/opt/freeswitch/db# sqlite3 core2.db SQLite version 3.5.9 Enter ".help" for instructions sqlite> .dump BEGIN TRANSACTION; COMMIT; This is ext2 on flash memory. Maybe I should move core.db on to tmpfs? I get the impression persistence isn't required for this file... -- Brian May From sprice at gmail.com Tue Aug 17 19:16:20 2010 From: sprice at gmail.com (SP) Date: Tue, 17 Aug 2010 21:16:20 -0500 Subject: [Freeswitch-users] enable passthrough of "Privacy: id" header in sip In-Reply-To: <2D0E16F0-C234-4C80-9174-5350C3EA7387@jerris.com> References: <1a5cbae9-5bd2-4fd1-d1d3-88fae8b6bd15@me.com> <201007221436.42762.sos@sokhapkin.dyndns.org> <2D0E16F0-C234-4C80-9174-5350C3EA7387@jerris.com> Message-ID: Oh you want current. :) No I got nothing. On Tuesday, August 17, 2010, Michael Jerris wrote: > Expiration 5/31/01 > An actual RFC as opposed to a 10 year old now expired draft? > Mike > On Aug 17, 2010, at 2:25 PM, SP wrote: > http://tools.ietf.org/html/draft-ietf-sip-privacy-00 > > 5.2 Anonymity Header Field Definition > blah blah blah..."If privacy is requested, it MUST be one or more of > "full", "uri", > ??"name", or "ipaddr"." > > > On Sun, Aug 15, 2010 at 17:52, Michael Jerris wrote: > is there a specification which defines this "uri" value? > > Mike > > On Jul 22, 2010, at 2:36 PM, Sergey Okhapkin wrote: > > FS doesn't recognize "uri" value of privacy tag, the recognized values are > "yes", "full", "name" and "number". Any other value is interpreted as "off". > See sofia.c, lines around 6660. > > On Thursday 22 July 2010, mike.burlingame wrote: > Ok so I am still kinda lost in trying to figure this one out here are the > two headers that I am looking at - basically someone sends an invite to FS > that ask's FS via RPID to hide the caller id info from downstream gateways > so the A-LEG is the invite going to FS - FS takes the invite and spits out > the B-LEG to go downstream HOWEVER FS does not copy the parameters > correctly as you can see in the initial invite A-LEG requested in the RPID > to be privacy=uri however on the B-LEG side FS set the RPID to privacy=off > in turn telling gateways downstream to display the CID info. > > A-LEG > Remote-Party-ID: ;party=calling;screen=yes;privacy=uri > > B-LEG > Remote-Party-ID: "NAME" > ;party=calling;screen=yes;privacy=off > > On Jul 22, 2010, at 09:35 AM, Michael Collins wrote: > > > > On Wed, Jul 21, 2010 at 4:18 PM, mike.burlingame < -- Shannon From brian at freeswitch.org Tue Aug 17 19:26:08 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Aug 2010 21:26:08 -0500 Subject: [Freeswitch-users] enable passthrough of "Privacy: id" header in sip In-Reply-To: References: <1a5cbae9-5bd2-4fd1-d1d3-88fae8b6bd15@me.com> <201007221436.42762.sos@sokhapkin.dyndns.org> <2D0E16F0-C234-4C80-9174-5350C3EA7387@jerris.com> Message-ID: <0D3AB67B-DCDE-4D69-9063-61EFB4077F91@freeswitch.org> you forgot to use big text! On Aug 17, 2010, at 9:16 PM, SP wrote: > Oh you want current. :) > > No I got nothing. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100817/1d6effa3/attachment.html From Victor at isptelecom.net Tue Aug 17 19:54:21 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Tue, 17 Aug 2010 22:54:21 -0400 Subject: [Freeswitch-users] NAT ACL and security In-Reply-To: <368C3633-0829-4E84-84DE-9D73265E7068@freeswitch.org> References: <4C698F97.5020505@isptelecom.net> <368C3633-0829-4E84-84DE-9D73265E7068@freeswitch.org> Message-ID: <4C6B4B5D.4040008@isptelecom.net> Brian, I understand that in treats everything as NAT. But what are consequences apart from pinging them every minute? I've looked into Wiki, but didn't see clear definition of "NAT behaviour" Thank you, Victor On -10/01/37 02:59 PM, Brian West wrote: > You're treating everything as if it were nat.... including public addresses... > > /b > > On Aug 16, 2010, at 2:20 PM, Victor Chukalovskiy wrote: > > >> I'm using >> >> >> in my SIP profile in order to make Freeswitch ping every phone registered to it. >> This works well for keeping phones on remote LANs reachable. >> >> My_nat ACL is defined as following: >> >> >> That is, it allows everybody. >> >> Question: am I making my system insecure by doing so? >> I believe "No" since ACL list "my_nat" is only used by appl-nat-acl parameter, >> but I don't know FreeSWITCH well enough to grantee that nothing else is affected. >> E.g. does anything else change if phone is considered NATed / non-NATed? >> >> Regards, >> Victor >> > > > From xyangni at gmail.com Tue Aug 17 20:47:16 2010 From: xyangni at gmail.com (xuyan yang) Date: Wed, 18 Aug 2010 11:47:16 +0800 Subject: [Freeswitch-users] Is mod_java supported on windows? In-Reply-To: References: Message-ID: I am trying to compile it and meet a undefined SWITCH_PREFIX_DIR. So I added #ifndef SWITCH_PREFIX_DIR #define SWITCH_PREFIX_DIR "." #endif It can be compiled now, but a link error stopped me: switch_swig_wrap.obj : error LNK2001: unresolved external symbol "struct JavaVM_ * javaVM" (?javaVM@@3PAUJavaVM_@@A) And I have already added jvm.lib from jdk to the link option. Googled it, and no answer found. On Wed, Aug 18, 2010 at 9:37 AM, Brian West wrote: > swig is NOT required to compile it... but nobody has tried it on windows. > > /b > > On Aug 17, 2010, at 8:26 PM, xuyan yang wrote: > > > In the latest git I didn't find any VC project file for mod_java. From > the makefile I found that swig is needed to conpile this. And swig does have > windows version. Is it possible to compile mod_java for windows from latest > git tree? _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/365c77d6/attachment.html From steveayre at gmail.com Wed Aug 18 01:11:48 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 18 Aug 2010 09:11:48 +0100 Subject: [Freeswitch-users] NAT ACL and security In-Reply-To: <4C6B4B5D.4040008@isptelecom.net> References: <4C698F97.5020505@isptelecom.net> <368C3633-0829-4E84-84DE-9D73265E7068@freeswitch.org> <4C6B4B5D.4040008@isptelecom.net> Message-ID: Mostly it changes the way URIs and Contact headers are generated because the actual port numbers used can be different those within the packet when using NAT. -Steve On 18 August 2010 03:54, Victor Chukalovskiy wrote: > Brian, > > I understand that in treats everything as NAT. > But what are consequences apart from pinging them every minute? > I've looked into Wiki, but didn't see clear definition of "NAT behaviour" > > Thank you, > Victor > > On -10/01/37 02:59 PM, Brian West wrote: > > You're treating everything as if it were nat.... including public > addresses... > > > > /b > > > > On Aug 16, 2010, at 2:20 PM, Victor Chukalovskiy wrote: > > > > > >> I'm using > >> > >> > >> in my SIP profile in order to make Freeswitch ping every phone > registered to it. > >> This works well for keeping phones on remote LANs reachable. > >> > >> My_nat ACL is defined as following: > >> > >> > >> That is, it allows everybody. > >> > >> Question: am I making my system insecure by doing so? > >> I believe "No" since ACL list "my_nat" is only used by appl-nat-acl > parameter, > >> but I don't know FreeSWITCH well enough to grantee that nothing else is > affected. > >> E.g. does anything else change if phone is considered NATed / non-NATed? > >> > >> Regards, > >> Victor > >> > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/2f4c72ab/attachment.html From steveayre at gmail.com Wed Aug 18 01:36:32 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 18 Aug 2010 09:36:32 +0100 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: "The power has failed recently though, maybe connected?" Quite likely. Anything in the disk write cache will be lost on a power failure. Two ways to avoid that are: 1) Battery backup for your RAID 2) Disable write cache (which will of course have a performance penalty) The database files would be actively changed, so any changes would be in the write cache resulting in a only partially written database when power returns. Note that each db file is a separate sqlite database. You will only need to repair/delete the damaged one(s). You might find that moving everything into a ODBC database with transaction support will result in less corruption if a power fails. -Steve On 18 August 2010 02:58, Brian May wrote: > On 7 August 2010 18:35, Brian May wrote: > >> As for how to do sqlite data recovery, try googling. e.g. > >> http://www.mail-archive.com/sqlite-users at sqlite.org/msg17538.html > > > > Thanks for the reference. I tried looking but obviously wasn't looking > > for the right search term. > > Curiously I had the same problem reoccur. Unfortunately freeswitch > appears to work until some time after the initial problem, makes it > hard to find out what happened. The power has failed recently though, > maybe connected? > > [ etc ] > 2010-08-17 04:30:40.728489 [CRIT] switch_core_sqldb.c:938 SQL thread > unable to commit transaction, records lost! > 2010-08-17 04:31:00.653705 [ERR] switch_core_sqldb.c:448 SQL ERR > [database disk image is malformed] > [ etc ] > > huey:/opt/freeswitch/db# ls -l core2.db > -rw-r--r-- 1 freeswitch daemon 1502208 2010-08-15 17:11 core2.db > huey:/opt/freeswitch/db# sqlite3 core2.db > SQLite version 3.5.9 > Enter ".help" for instructions > sqlite> .dump > BEGIN TRANSACTION; > COMMIT; > > This is ext2 on flash memory. Maybe I should move core.db on to tmpfs? > I get the impression persistence isn't required for this file... > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/16e7038a/attachment-0001.html From steveayre at gmail.com Wed Aug 18 01:38:32 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 18 Aug 2010 09:38:32 +0100 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: Oh, I didn't spot the ext2 mention... I guess this is an embedded system? The tmpfs option might work well, although it'll be all or nothing since everything is stored in the same directory. A symlink to a tmpfs mount probably wouldn't work as FS will try to create a new file in the db directory. -Steve On 18 August 2010 09:36, Steven Ayre wrote: > "The power has failed recently though, maybe connected?" > > Quite likely. Anything in the disk write cache will be lost on a power > failure. Two ways to avoid that are: > 1) Battery backup for your RAID > 2) Disable write cache (which will of course have a performance penalty) > > The database files would be actively changed, so any changes would be in > the write cache resulting in a only partially written database when power > returns. > > Note that each db file is a separate sqlite database. You will only need to > repair/delete the damaged one(s). > > You might find that moving everything into a ODBC database with transaction > support will result in less corruption if a power fails. > > -Steve > > > > On 18 August 2010 02:58, Brian May wrote: > >> On 7 August 2010 18:35, Brian May wrote: >> >> As for how to do sqlite data recovery, try googling. e.g. >> >> http://www.mail-archive.com/sqlite-users at sqlite.org/msg17538.html >> > >> > Thanks for the reference. I tried looking but obviously wasn't looking >> > for the right search term. >> >> Curiously I had the same problem reoccur. Unfortunately freeswitch >> appears to work until some time after the initial problem, makes it >> hard to find out what happened. The power has failed recently though, >> maybe connected? >> >> [ etc ] >> 2010-08-17 04:30:40.728489 [CRIT] switch_core_sqldb.c:938 SQL thread >> unable to commit transaction, records lost! >> 2010-08-17 04:31:00.653705 [ERR] switch_core_sqldb.c:448 SQL ERR >> [database disk image is malformed] >> [ etc ] >> >> huey:/opt/freeswitch/db# ls -l core2.db >> -rw-r--r-- 1 freeswitch daemon 1502208 2010-08-15 17:11 core2.db >> huey:/opt/freeswitch/db# sqlite3 core2.db >> SQLite version 3.5.9 >> Enter ".help" for instructions >> sqlite> .dump >> BEGIN TRANSACTION; >> COMMIT; >> >> This is ext2 on flash memory. Maybe I should move core.db on to tmpfs? >> I get the impression persistence isn't required for this file... >> -- >> Brian May >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/963e682a/attachment.html From david.ponzone at ipeva.fr Wed Aug 18 01:52:28 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 18 Aug 2010 10:52:28 +0200 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: Brian, I think there is a good reason to move your db/ dir to tmpfs. If you leave that on flash, you gonna do a lot of writes to the flash, and you could eventually wear it out if the box is installed for several years. How quick it could be worn out will depend on the quality of flash you put in that. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 18/08/2010 ? 03:58, Brian May a ?crit : > This is ext2 on flash memory. Maybe I should move core.db on to tmpfs? > I get the impression persistence isn't required for this file... > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/50c29eb3/attachment.html From brian at microcomaustralia.com.au Wed Aug 18 02:29:43 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Wed, 18 Aug 2010 19:29:43 +1000 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: On 18 August 2010 18:38, Steven Ayre wrote: > Oh, I didn't spot the ext2 mention... I guess this is an embedded system? Yes. > The tmpfs option might work well, although it'll be all or nothing since > everything is stored in the same directory. A symlink to a tmpfs mount > probably wouldn't work as FS will try to create a new file in the db > directory. Believe it or not, it seems to work, it seems to work. I created a dangling symlink to /tmp/core.db, and it put the file in the correct spot, under /tmp. Not sure if this is the best way or not... -- Brian May From brian at microcomaustralia.com.au Wed Aug 18 02:32:01 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Wed, 18 Aug 2010 19:32:01 +1000 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: On 18 August 2010 18:52, David Ponzone wrote: > I think there is a good reason to move your db/ dir to tmpfs. > If you leave that on flash, you gonna do a lot of writes to the flash, and > you could eventually wear it out if the box is installed for several years. > How quick it could be worn out will depend on the quality of flash you put > in that. Yes, was thinking that too. However, I suspect I can't put voicemail_default.db onto tmpfs, I suspect that needs to have persistence - I assume it stores voicemail stuff not found anywhere else. Fortunately this file doesn't change often. -- Brian May From steveayre at gmail.com Wed Aug 18 02:53:00 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 18 Aug 2010 10:53:00 +0100 Subject: [Freeswitch-users] database corruption errors In-Reply-To: References: Message-ID: Checking the source code it contains two tables. One stores a list of voicemail messages including time, caller information and the location of the recorded voicemail file. The other stores the user preferences (password etc). I'd say both of those will want to be persistent. -Steve On 18 August 2010 10:32, Brian May wrote: > On 18 August 2010 18:52, David Ponzone wrote: > > I think there is a good reason to move your db/ dir to tmpfs. > > If you leave that on flash, you gonna do a lot of writes to the flash, > and > > you could eventually wear it out if the box is installed for several > years. > > How quick it could be worn out will depend on the quality of flash you > put > > in that. > > Yes, was thinking that too. > > However, I suspect I can't put voicemail_default.db onto tmpfs, I > suspect that needs to have persistence - I assume it stores voicemail > stuff not found anywhere else. > > Fortunately this file doesn't change often. > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/d37b3746/attachment-0001.html From chaitanya at vivainfomedia.com Wed Aug 18 03:02:29 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Wed, 18 Aug 2010 15:32:29 +0530 Subject: [Freeswitch-users] Freeswitch: issue in making simultaneous call Message-ID: Hey I am facing problem in making few outbound calls simultaneously. I am executing "api originate" command from perl over ESL. After calling originate ,same perl file listens for CHANNEL_ORIGINATE CHANNEL_ANSWER CHANNEL_HANGUP_COMPLETE events. When perl file receives CHANNEL_HANGUP_COMPLETE event, it get terminates. I have done this code of originate & hangup listen event in perl cgi. When i do 10-20 request (10-20 simultaneous call), at freeswitch only 2-3 calls get serviced. To find out issue when i commented the code of event listen in cgi, i was able to make all 10-20 calls simulateneously. I have already checked apache configuration & is working well. Now i want to know whether freeswitch has limitation on number of clients listening to its events or there is some other issue of ESL. Please guide me what could be the issue. Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/69785075/attachment.html From abu.4000 at gmail.com Wed Aug 18 04:20:14 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Wed, 18 Aug 2010 16:50:14 +0530 Subject: [Freeswitch-users] How many calls we can park in a freeswitch Message-ID: Dear All, I just want to know how many call we can park in a freeswitch at the maximum ?, -- *Best Regards, **Abubacker systems engineer bk systems (p) ltd** * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/a12aa7b8/attachment.html From mnhassan at usa.net Wed Aug 18 06:08:41 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Wed, 18 Aug 2010 19:08:41 +0600 Subject: [Freeswitch-users] How many calls we can park in a freeswitch In-Reply-To: References: Message-ID: I don't think there is a maximum limit bound by the software, other than any theoretical limitation from hardware, OS, etc. Regards HASSAN On Wed, Aug 18, 2010 at 17:20, Abubacker siddiq wrote: > Dear All, > I just want to know how many call we can park in a freeswitch at the > maximum ?, > > -- > *Best Regards, > **Abubacker > systems engineer > bk systems (p) ltd** > * > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/6ae9e07d/attachment.html From alanlo at commverge.com Wed Aug 18 02:39:54 2010 From: alanlo at commverge.com (alan lo) Date: Wed, 18 Aug 2010 17:39:54 +0800 Subject: [Freeswitch-users] IVR menu failure/timeout action Message-ID: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> Dear all, Referring to the threat below, where should the dp extension stored? In the same XML of ivr or should store it in other locations? Thx! Alan I have confirmed this behavior in the latest git as of last night. I don't know if this is by design or not. I would open JIRA as a feature request to have this behavior modified. The other alternative would be to have each IVR submenu called via transfer instead of menu-sub: Then have a dp extension to handle that: -MC On Mon, Jul 19, 2010 at 2:37 AM, Raymond Chan >wrote: > Hi all, > > > > I am experiencing an IVR problem. Except the top menu, all other sub menu > will not hang up after reached max invalid input or timeout limit. It will > go back up upper menu. I want it to load the exit-sound and then hang the > call when it reached max invalid input or timeout limit. Do you tell how to > configure? > > > > > > greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" > > > greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" > > > invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" > > exit-sound="voicemail/vm-goodbye-m.wav" > > timeout="5000" > > max-failures="2"> > > > > > > > > > > Thanks > > > > Raymond -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/215c2a79/attachment-0001.html From pjintheusa at gmail.com Wed Aug 18 06:23:41 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 18 Aug 2010 09:23:41 -0400 Subject: [Freeswitch-users] start_dtmf on BLeg Channel from the dialplan In-Reply-To: References: Message-ID: Thanks. That didn't do it. Neither did export nolocal:execute_on_answer=start_dtmf umm... On Tue, Aug 17, 2010 at 7:17 PM, Michael Collins wrote: > Try this and see if it helps: > > http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_app > -MC > > On Tue, Aug 17, 2010 at 3:11 PM, Phillip Jones wrote: > >> Hi there, >> >> Is there a way to execute start_dtmf on the B Leg, from the >> dialplan, before you bridge call? >> >> >> I am using group_confirm, so that a called user is bridged when they >> answer and press 1. >> >> With one of my carriers, DTMF is not working - pressing 1 does nothing. >> >> If I use group_confirm=exec and send the B leg to a script on answer, and >> execute "start_dtmf" on the b leg session in the script, all works as >> expected and pressing 1 bridges the call. >> >> I would perfer not to use a script if there is a way of achieving this via >> the dialplan. >> >> Any help appreciated. >> >> Thanks >> >> Pj >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/d4552015/attachment.html From 12ukwn at gmail.com Wed Aug 18 06:34:37 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 18 Aug 2010 15:34:37 +0200 Subject: [Freeswitch-users] How many calls we can park in a freeswitch In-Reply-To: References: Message-ID: <20100818153437.62793e51@anubis.defcon1> Le Wed, 18 Aug 2010 16:50:14 +0530, Abubacker siddiq a ?crit : > Dear All, > I just want to know how many call we can park in a freeswitch at the > maximum ?, So many. JY -- From david.ponzone at ipeva.fr Wed Aug 18 08:16:50 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 18 Aug 2010 17:16:50 +0200 Subject: [Freeswitch-users] IVR menu failure/timeout action In-Reply-To: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> References: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> Message-ID: I guess if you use: transfer sub_menu_1 XML it will stay in the current context. And if you use: transfer sub_menu_1 XML foo it will transfer to context foo. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 18/08/2010 ? 11:39, alan lo a ?crit : > Dear all, > > Referring to the threat below, where should the dp extension stored? > In the same XML of ivr or should store it in other locations? > > Thx! > > Alan > > I have confirmed this behavior in the latest git as of last night. I > don't > know if this is by design or not. I would open JIRA as a feature > request to > have this behavior modified. The other alternative would be to have > each IVR > submenu called via transfer instead of menu-sub: > > > > > > > Then have a dp extension to handle that: > > > > > > > -MC > > On Mon, Jul 19, 2010 at 2:37 AM, Raymond Chan commverge.com>wrote: > > > Hi all, > > > > > > > > I am experiencing an IVR problem. Except the top menu, all other > sub menu > > will not hang up after reached max invalid input or timeout limit. > It will > > go back up upper menu. I want it to load the exit-sound and then > hang the > > call when it reached max invalid input or timeout limit. Do you > tell how to > > configure? > > > > > > > > > > > > > greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" > > > > > > greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" > > > > > > invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" > > > > exit-sound="voicemail/vm-goodbye-m.wav" > > > > timeout="5000" > > > > max-failures="2"> > > > > param="playback > > C:/FreeSWITCH/recordings/800-ivr-announcement-m_ch.wav"/> > > > > > > > > > > > > > > > > Thanks > > > > > > > > Raymond > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/eda274f3/attachment-0001.html From david.ponzone at ipeva.fr Wed Aug 18 08:25:59 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 18 Aug 2010 17:25:59 +0200 Subject: [Freeswitch-users] IVR menu failure/timeout action In-Reply-To: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> References: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> Message-ID: <0326E15F-D932-4F58-98DD-2A4353E1A9AF@ipeva.fr> I guess if you use: transfer sub_menu_1 XML it will stay in the current context. And if you use: transfer sub_menu_1 XML foo it will transfer to context foo. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 18/08/2010 ? 11:39, alan lo a ?crit : > Dear all, > > Referring to the threat below, where should the dp extension stored? > In the same XML of ivr or should store it in other locations? > > Thx! > > Alan > > I have confirmed this behavior in the latest git as of last night. I > don't > know if this is by design or not. I would open JIRA as a feature > request to > have this behavior modified. The other alternative would be to have > each IVR > submenu called via transfer instead of menu-sub: > > > > > > > Then have a dp extension to handle that: > > > > > > > -MC > > On Mon, Jul 19, 2010 at 2:37 AM, Raymond Chan commverge.com>wrote: > > > Hi all, > > > > > > > > I am experiencing an IVR problem. Except the top menu, all other > sub menu > > will not hang up after reached max invalid input or timeout limit. > It will > > go back up upper menu. I want it to load the exit-sound and then > hang the > > call when it reached max invalid input or timeout limit. Do you > tell how to > > configure? > > > > > > > > > > > > > greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" > > > > > > greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" > > > > > > invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" > > > > exit-sound="voicemail/vm-goodbye-m.wav" > > > > timeout="5000" > > > > max-failures="2"> > > > > param="playback > > C:/FreeSWITCH/recordings/800-ivr-announcement-m_ch.wav"/> > > > > > > > > > > > > > > > > Thanks > > > > > > > > Raymond > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/d45912ab/attachment-0001.html From msc at freeswitch.org Wed Aug 18 09:42:17 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Aug 2010 09:42:17 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly! Message-ID: C'mon down! http://wiki.freeswitch.org/wiki/FS_weekly_2010_08_18 Brian K West will be by to talk about NAT and VoIP with FreeSWITCH. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/8642e8ed/attachment.html From pjintheusa at gmail.com Wed Aug 18 09:52:53 2010 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 18 Aug 2010 12:52:53 -0400 Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly! In-Reply-To: References: Message-ID: What happened to the drk chat about CDR tracking. Did I miss it somehow? On Wed, Aug 18, 2010 at 12:42 PM, Michael Collins wrote: > C'mon down! > > http://wiki.freeswitch.org/wiki/FS_weekly_2010_08_18 > > Brian K West will be by to talk about NAT and VoIP with FreeSWITCH. > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/49f37878/attachment.html From moises.silva at gmail.com Wed Aug 18 11:00:18 2010 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 18 Aug 2010 14:00:18 -0400 Subject: [Freeswitch-users] Howler - Gone? In-Reply-To: References: <87168847915b78750eb7c4def0258461@mail.gmail.com> Message-ID: On Tue, Aug 17, 2010 at 6:08 PM, Gavin Henry wrote: > On 25 July 2010 18:48, Kristian Kielhofner wrote: > > The only reason I ask is because I just bought a 480 for testing ;)... > > This one? http://www.voipon.co.uk/sangoma-d100480e-p-2566.html for > testing??? > > Yes, that's the D100. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com > -- > http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/72841a76/attachment.html From ovvenkatesan at gmail.com Wed Aug 18 09:13:28 2010 From: ovvenkatesan at gmail.com (ovvenkat) Date: Wed, 18 Aug 2010 21:43:28 +0530 Subject: [Freeswitch-users] How many calls we can park in a freeswitch In-Reply-To: <20100818153437.62793e51@anubis.defcon1> References: <20100818153437.62793e51@anubis.defcon1> Message-ID: Hi to all, Any one plz suggest me, If I want to handle 200 channels, what will be me the server and memory configuration required? On Wed, Aug 18, 2010 at 7:04 PM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote: > Le Wed, 18 Aug 2010 16:50:14 +0530, > Abubacker siddiq a ?crit : > > > Dear All, > > I just want to know how many call we can park in a freeswitch at the > > maximum ?, > > So many. > > JY > -- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/15ce9158/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Aug 18 11:17:13 2010 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 18 Aug 2010 20:17:13 +0200 Subject: [Freeswitch-users] Which db/*.db files are not suited for tmpfs? Message-ID: <4C6C23A9.5@puzzled.xs4all.nl> Hi, Earlier I read the interesting postings about the use of tmpfs. Looking at the files in db/ I wonder if anyone knows which ones needs to be persistent across FreeSWITCH restarts or which are unsuitable for use with tmpfs e.g. in case of a power failure. From Steven Ayre's comment I understand that voicemail_default.db should not be put on tmpfs. Are there any other db files that should *not* be put on tmpfs? Overview of files I have in db/ on my test box: call_limit.db core.db directory.db fifo.db sofia_reg_external.db sofia_reg_internal.db sofia_reg_internal-ipv6.db voicemail_default.db zrtp.dat Regards, Patrick From dipankarsarkar at gmail.com Wed Aug 18 11:20:56 2010 From: dipankarsarkar at gmail.com (Dipankar Sarkar) Date: Wed, 18 Aug 2010 23:50:56 +0530 Subject: [Freeswitch-users] How many calls we can park in a freeswitch In-Reply-To: References: Message-ID: Hi*,* * * *You should anyways try to benchmark your system, so that you have some idea. It should not be very tough to see the limits when you play around. * * * Thanks Dipankar Sarkar http://desinerd.com On Wed, Aug 18, 2010 at 4:50 PM, Abubacker siddiq wrote: > Dear All, > I just want to know how many call we can park in a freeswitch at the > maximum ?, > > -- > *Best Regards, > **Abubacker > systems engineer > bk systems (p) ltd** > * > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/444a2e7b/attachment.html From msc at freeswitch.org Wed Aug 18 12:13:18 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Aug 2010 12:13:18 -0700 Subject: [Freeswitch-users] Freeswitch: issue in making simultaneous call In-Reply-To: References: Message-ID: Comments inline... On Wed, Aug 18, 2010 at 3:02 AM, Chaitanya Bhatt // Viva < chaitanya at vivainfomedia.com> wrote: > Hey > > I am facing problem in making few outbound calls simultaneously. I am > executing "api originate" command from perl over ESL. After calling > originate ,same perl file listens for CHANNEL_ORIGINATE CHANNEL_ANSWER > CHANNEL_HANGUP_COMPLETE events. When perl file receives > CHANNEL_HANGUP_COMPLETE event, it get terminates. > > I have done this code of originate & hangup listen event in perl cgi. When > i do 10-20 request (10-20 simultaneous call), at freeswitch only 2-3 calls > get serviced. > To find out issue when i commented the code of event listen in cgi, i was > able to make all 10-20 calls simulateneously. > I wonder if your script isn't able to handle all the events quickly enough? > > I have already checked apache configuration & is working well. > Now i want to know whether freeswitch has limitation on number of clients > listening to its events or there is some other issue of ESL. > Using Perl, ESL, and POE I've done upwards of 50-60 simultaneous calls with originate. I used bgapi originate xxxx. It's not the most efficient way but it did work. However, the best way to handle something like this is to have several processes each with a specific purpose: one to send the bgapi originate commands and one to listen for the CHANNEL_XXX events and act accordingly. By having each process do just one thing you can scale better and it's easier to identify trouble spots. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/6eaf3afa/attachment.html From msc at freeswitch.org Wed Aug 18 12:21:08 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Aug 2010 12:21:08 -0700 Subject: [Freeswitch-users] Which db/*.db files are not suited for tmpfs? In-Reply-To: <4C6C23A9.5@puzzled.xs4all.nl> References: <4C6C23A9.5@puzzled.xs4all.nl> Message-ID: I was surprised at the lack of information on this topic in the wiki. If anyone has information on how to put SQLite db's into a ramdisk and any pointers on that let me know so that we can properly document it on the wiki. Also, I'm sure there are snippets in past mail list conversations so be sure to search them for nuggets of wisdom... -MC On Wed, Aug 18, 2010 at 11:17 AM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > Hi, > > Earlier I read the interesting postings about the use of tmpfs. Looking > at the files in db/ I wonder if anyone knows which ones needs to be > persistent across FreeSWITCH restarts or which are unsuitable for use > with tmpfs e.g. in case of a power failure. From Steven Ayre's comment I > understand that voicemail_default.db should not be put on tmpfs. Are > there any other db files that should *not* be put on tmpfs? > > Overview of files I have in db/ on my test box: > > call_limit.db > core.db > directory.db > fifo.db > sofia_reg_external.db > sofia_reg_internal.db > sofia_reg_internal-ipv6.db > voicemail_default.db > zrtp.dat > > > Regards, > Patrick > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/46fde1fd/attachment-0001.html From jeff at jefflenk.com Wed Aug 18 13:12:47 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 18 Aug 2010 13:12:47 -0700 (PDT) Subject: [Freeswitch-users] FS processing 302 In-Reply-To: References: <396D3026B6034431B93A6B3DC0803461@left> <1281542398575-5412733.post@n2.nabble.com> <55D75CA26DF8494B9FD88112BEAE8211@left> <1281554134702-5413561.post@n2.nabble.com> <1D954802FAF148468C766D9636EF95F1@left> <1282054315191-5432276.post@n2.nabble.com> Message-ID: <1282162367118-5437694.post@n2.nabble.com> Chris, can you use a debugger and place a breakpoint on this line? sofia_glue.c:2036 sip:10.11.0.20 Setting proxy route to sofia/external/99955520000 and determine what/why this is happening? Just guessing here! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-processing-302-tp5393783p5437694.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Wed Aug 18 13:51:21 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Aug 2010 15:51:21 -0500 Subject: [Freeswitch-users] Freeswitch: issue in making simultaneous call In-Reply-To: References: Message-ID: and make sure the sps is not set too low On Wed, Aug 18, 2010 at 2:13 PM, Michael Collins wrote: > Comments inline... > > On Wed, Aug 18, 2010 at 3:02 AM, Chaitanya Bhatt // Viva > wrote: >> >> Hey >> >> I am facing problem in making few outbound calls simultaneously. I am >> executing "api originate" command from perl over ESL. After calling >> originate ,same perl file listens for CHANNEL_ORIGINATE CHANNEL_ANSWER >> CHANNEL_HANGUP_COMPLETE events. When perl file receives >> CHANNEL_HANGUP_COMPLETE event, it get terminates. >> >> I have done this code of originate & hangup listen event in perl cgi. When >> i do 10-20 request (10-20 simultaneous call), at freeswitch only 2-3 calls >> get serviced. >> To find out issue when i commented the code of event listen in cgi, i was >> able to make all 10-20 calls simulateneously. > > I wonder if your script isn't able to handle all the events quickly enough? > >> >> I have already checked apache configuration & is working well. >> Now i want to know whether freeswitch has limitation on number of clients >> listening to its events or there is some other issue of ESL. > > Using Perl, ESL, and POE I've done upwards of 50-60 simultaneous calls with > originate. I used bgapi originate xxxx. It's not the most efficient way but > it did work. However, the best way to handle something like this is to have > several processes each with a specific purpose: one to send the bgapi > originate commands and one to listen for the CHANNEL_XXX events and act > accordingly. By having each process do just one thing you can scale better > and it's easier to identify trouble spots. > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From sean at obscuradigital.com Wed Aug 18 16:26:14 2010 From: sean at obscuradigital.com (Sean Holt) Date: Wed, 18 Aug 2010 16:26:14 -0700 Subject: [Freeswitch-users] Paging calls Message-ID: Hello list, I wondering if there?s anyone on the list that has had success configuring paging/intercom using an all polycom phone environment and freeswitch. I?ve been able to configure the polycom config files to auto answer on the second line but I?m lost when configuring my dialplan. Currently I?m pointing an extension to a ring group, but the only phone that auto-answers is the first phone in the ring group list. Phones: 321 polycom Thanks for any help Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/9a4d9a16/attachment.html From dujinfang at gmail.com Wed Aug 18 17:31:41 2010 From: dujinfang at gmail.com (Seven Du) Date: Thu, 19 Aug 2010 08:31:41 +0800 Subject: [Freeswitch-users] performance between bridged call and conference Message-ID: Hi, Can someone explain the performance difference between bridged calls and 2-party conference? or just in the code point of view? Since in some scenarios third party may join into a bridged call, so we need to transfer a bridged call into a conference first. Make a conference anyway event for 2-parties will make logic simpler and clear. Thanks. -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From william.suffill at gmail.com Wed Aug 18 18:59:06 2010 From: william.suffill at gmail.com (William Suffill) Date: Wed, 18 Aug 2010 21:59:06 -0400 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: Message-ID: Sample cfgs should have a cfg of something similar. Call the phones & dump them in a conference bridge. --W Sent from my phone so sorry for any issues quoting spelling etc On Aug 18, 2010 7:36 PM, "Sean Holt" wrote: Hello list, I wondering if there?s anyone on the list that has had success configuring paging/intercom using an all polycom phone environment and freeswitch. I?ve been able to configure the polycom config files to auto answer on the second line but I?m lost when configuring my dialplan. Currently I?m pointing an extension to a ring group, but the only phone that auto-answers is the first phone in the ring group list. Phones: 321 polycom Thanks for any help Sean _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100818/a8fd346a/attachment.html From brian at freeswitch.org Wed Aug 18 19:04:16 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Aug 2010 21:04:16 -0500 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: Message-ID: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> see mad boss example. /b On Aug 18, 2010, at 8:59 PM, William Suffill wrote: > Sample cfgs should have a cfg of something similar. Call the phones & dump them in a conference bridge. > > --W > From mnhassan at usa.net Wed Aug 18 20:14:55 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Thu, 19 Aug 2010 09:14:55 +0600 Subject: [Freeswitch-users] Paging calls In-Reply-To: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> Message-ID: Buy the book! Would show you the light, whenever you feel lost in the dark! :) Regards HASSAN On Thu, Aug 19, 2010 at 08:04, Brian West wrote: > see mad boss example. > > /b > > On Aug 18, 2010, at 8:59 PM, William Suffill wrote: > > > Sample cfgs should have a cfg of something similar. Call the phones & > dump them in a conference bridge. > > > > --W > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/2cabb30b/attachment.html From kward at binarysignal.com Wed Aug 18 20:35:18 2010 From: kward at binarysignal.com (Kurt Ward) Date: Wed, 18 Aug 2010 20:35:18 -0700 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> Message-ID: <0858DDC6-9DB9-4296-ACE3-37D8E259DE49@binarysignal.com> I will second that remark, it is a really good resource, and is well presented. On Aug 18, 2010, at 8:14 PM, Nyamul Hassan wrote: > Buy the book! Would show you the light, whenever you feel lost in the dark! > > :) > > Regards > HASSAN > > > > On Thu, Aug 19, 2010 at 08:04, Brian West wrote: > see mad boss example. > > /b > > On Aug 18, 2010, at 8:59 PM, William Suffill wrote: > > > Sample cfgs should have a cfg of something similar. Call the phones & dump them in a conference bridge. > > > > --W > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chaitanya at vivainfomedia.com Wed Aug 18 23:05:05 2010 From: chaitanya at vivainfomedia.com (Chaitanya Bhatt // Viva) Date: Thu, 19 Aug 2010 11:35:05 +0530 Subject: [Freeswitch-users] Freeswitch: issue in making simultaneous call In-Reply-To: References: Message-ID: Thanks for your quick response !!! I think it's better to have separate process for originate & event listen. But due to my application design issue, i have to stick with same cgi process for originate & event listen. I will check feasibility of separate process in my app design. Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Chaitanya Bhatt Software Engineer. Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, New Link Road, Opp. Oshiwara Bus Depot, Goregaon West, Mumbai 400104. Direct: +91.22.40310356 Board: +91.22.40310310 Email : chaitanya at vivainfomedia.com Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 On Thu, Aug 19, 2010 at 2:21 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > and make sure the sps is not set too low > > On Wed, Aug 18, 2010 at 2:13 PM, Michael Collins > wrote: > > Comments inline... > > > > On Wed, Aug 18, 2010 at 3:02 AM, Chaitanya Bhatt // Viva > > wrote: > >> > >> Hey > >> > >> I am facing problem in making few outbound calls simultaneously. I am > >> executing "api originate" command from perl over ESL. After calling > >> originate ,same perl file listens for CHANNEL_ORIGINATE CHANNEL_ANSWER > >> CHANNEL_HANGUP_COMPLETE events. When perl file receives > >> CHANNEL_HANGUP_COMPLETE event, it get terminates. > >> > >> I have done this code of originate & hangup listen event in perl cgi. > When > >> i do 10-20 request (10-20 simultaneous call), at freeswitch only 2-3 > calls > >> get serviced. > >> To find out issue when i commented the code of event listen in cgi, i > was > >> able to make all 10-20 calls simulateneously. > > > > I wonder if your script isn't able to handle all the events quickly > enough? > > > >> > >> I have already checked apache configuration & is working well. > >> Now i want to know whether freeswitch has limitation on number of > clients > >> listening to its events or there is some other issue of ESL. > > > > Using Perl, ESL, and POE I've done upwards of 50-60 simultaneous calls > with > > originate. I used bgapi originate xxxx. It's not the most efficient way > but > > it did work. However, the best way to handle something like this is to > have > > several processes each with a specific purpose: one to send the bgapi > > originate commands and one to listen for the CHANNEL_XXX events and act > > accordingly. By having each process do just one thing you can scale > better > > and it's easier to identify trouble spots. > > > > -MC > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/2fe5f776/attachment-0001.html From dujinfang at gmail.com Thu Aug 19 00:14:43 2010 From: dujinfang at gmail.com (Seven Du) Date: Thu, 19 Aug 2010 15:14:43 +0800 Subject: [Freeswitch-users] Some build 32 bit binary on 64bit Mac In-Reply-To: References: Message-ID: I think this is the correct way. CFLAGS=-m32 CXXFLAGS=-m32 LDFLAGS=-m32 ./configure --prefix=/opt/fs-i386 --host=i386 Thanks MikeJ and Math on irc. btw, @Math, I dont' use --build=i386-apple-darwin10, it seems throws errors. On Thu, Aug 12, 2010 at 8:20 PM, Seven Du wrote: > There are many reasons I need a 32bit lib: > > 1) 64bit app cannot load flash in webkit in QT(at least in my case). > I'm try to build a client with libfreeswitch and flash in. > > 2) I want the lib run in iPhone simulator. Eventually on ARM on iphone. > > 3) If you release prebuilt desktop binary, you'd better don't forget > 32 bit users. > > 4) perhaps more... > > > On Thu, Aug 12, 2010 at 7:33 PM, Steven Ayre wrote: >> If you're on 64bit, why do you want to compile as 32bit? >> >> -Steve >> >> >> >> On 12 August 2010 11:36, Seven Du wrote: >>> >>> I'm on Mac 10.6.4 64bit, a fresh clone of git head. I tried to build >>> FS into 32 bit. >>> >>> I use the following command when configure: >>> >>> CFLAGS="-arch i386" CXXFLAGS="-arch i386" LDFLAGS="-arch i386" >>> ./configure --prefix=/opt/fs-i386 --host=i386 --build=i386 >>> --target=i386 >>> >>> >>> However, when make, it throws some errors, I manually *change* the >>> source like from "ifdef" to "ifndef" to trick it and make further, >>> however, I still couldn't get fully built. >>> >>> some errors below. and ideas? thanks. >>> >>> >>> >>> src/switch_apr.c: In function 'switch_vasprintf': >>> src/switch_apr.c:1024: warning: implicit declaration of function >>> 'vasprintf' >>> >>> >>> >>> src/switch_core_session.c: In function 'switch_core_session_thread': >>> src/switch_core_session.c:1193: warning: format '%d' expects type >>> 'int', but argument 8 has type 'switch_size_t' >>> src/switch_core_session.c:1203: warning: format '%d' expects type >>> 'int', but argument 8 has type 'switch_size_t' >>> src/switch_core_session.c:1221: warning: format '%d' expects type >>> 'int', but argument 8 has type 'switch_size_t' >>> >>> >>> src/switch_core.c: In function 'send_heartbeat': >>> src/switch_core.c:84: warning: format '%d' expects type 'int', but >>> argument 5 has type 'long unsigned int' >>> src/switch_core.c: In function 'change_user_group': >>> src/switch_core.c:677: warning: implicit declaration of function >>> 'setgroups' >>> src/switch_core.c:704: warning: implicit declaration of function >>> 'initgroups' >>> >>> >>> >>> >>> src/switch_core.c: In function 'send_heartbeat': >>> src/switch_core.c:84: warning: format '%d' expects type 'int', but >>> argument 5 has type 'long unsigned int' >>> src/switch_core.c: In function 'change_user_group': >>> src/switch_core.c:677: warning: implicit declaration of function >>> 'setgroups' >>> src/switch_core.c:704: warning: implicit declaration of function >>> 'initgroups' >>> >>> >>> >>> >>> >>> src/switch_rtp.c: In function 'rtp_common_write': >>> src/switch_rtp.c:3437: error: 'u_long' undeclared (first use in this >>> function) >>> src/switch_rtp.c:3437: error: (Each undeclared identifier is reported only >>> once >>> src/switch_rtp.c:3437: error: for each function it appears in.) >>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>> make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>> >>> >>> >>> >>> >>> src/switch_utils.c: In function 'switch_build_uri': >>> src/switch_utils.c:1530: error: 'NI_MAXHOST' undeclared (first use in >>> this function) >>> src/switch_utils.c:1530: error: (Each undeclared identifier is >>> reported only once >>> src/switch_utils.c:1530: error: for each function it appears in.) >>> src/switch_utils.c:1530: error: 'NI_MAXSERV' undeclared (first use in >>> this function) >>> >>> >>> >>> more detailed log: >>> >>> http://pastebin.freeswitch.org/13615 >>> >>> >>> >>> -- >>> Blog: http://www.dujinfang.com >>> Proj:? http://www.freeswitch.org.cn >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Blog: http://www.dujinfang.com > Proj:? http://www.freeswitch.org.cn > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From asilva at wirelessmundi.com Thu Aug 19 01:09:50 2010 From: asilva at wirelessmundi.com (Antonio) Date: Thu, 19 Aug 2010 10:09:50 +0200 Subject: [Freeswitch-users] core.db table channels application field store wrong data when executing voicemail Message-ID: <1282205390.25391.41.camel@marces.tc.commsmundi.com> Hi, during multiple selects in the table "channels" just realized that when a call is executing the app voicemail, to check for messages, during the navigation menu of the voicemail it changes to application "sleep". Is it normal? or is a bug for fs? The call: During the auth: sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| in the navigation menu: sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| 101 at 192.168.10.25||EARLY|||| sqlite> select * from channels; 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| 101 at 192.168.10.25||EARLY|||| Thanks, Ant?nio From anthony.minessale at gmail.com Thu Aug 19 07:20:24 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Aug 2010 09:20:24 -0500 Subject: [Freeswitch-users] core.db table channels application field store wrong data when executing voicemail In-Reply-To: <1282205390.25391.41.camel@marces.tc.commsmundi.com> References: <1282205390.25391.41.camel@marces.tc.commsmundi.com> Message-ID: No, every app that is ever executed will change that field, some apps in turn execute more apps. On Thu, Aug 19, 2010 at 3:09 AM, Antonio wrote: > > Hi, > > during multiple selects in the table "channels" just realized that when > a call is executing the app voicemail, to check for messages, during the > navigation menu of the voicemail it changes to application "sleep". > > Is it normal? or is a bug for fs? > > > The call: > > During the auth: > > sqlite> select * from channels; > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > sqlite> select * from channels; > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > sqlite> select * from channels; > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > sqlite> select * from channels; > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > sqlite> select * from channels; > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > sqlite> select * from channels; > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > in the navigation menu: > > sqlite> select * from channels; > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| > 101 at 192.168.10.25||EARLY|||| > > sqlite> select * from channels; > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| > 101 at 192.168.10.25||EARLY|||| > > > > Thanks, > Ant?nio > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Thu Aug 19 07:44:13 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Aug 2010 09:44:13 -0500 Subject: [Freeswitch-users] Which db/*.db files are not suited for tmpfs? In-Reply-To: References: <4C6C23A9.5@puzzled.xs4all.nl> Message-ID: core.db and possibly call_limit.db are the only 2 that have any intense usage so really only the core one, (which is regenerated on every new run anyway) needs to be moved to ram for a visible improvement. On Wed, Aug 18, 2010 at 2:21 PM, Michael Collins wrote: > I was surprised at the lack of information on this topic in the wiki. If > anyone has information on how to put SQLite db's into a ramdisk and any > pointers on that let me know so that we can properly document it on the > wiki. Also, I'm sure there are snippets in past mail list conversations so > be sure to search them for nuggets of wisdom... > > -MC > > On Wed, Aug 18, 2010 at 11:17 AM, Patrick Lists > wrote: >> >> Hi, >> >> Earlier I read the interesting postings about the use of tmpfs. Looking >> at the files in db/ I wonder if anyone knows which ones needs to be >> persistent across FreeSWITCH restarts or which are unsuitable for use >> with tmpfs e.g. in case of a power failure. From Steven Ayre's comment I >> understand that voicemail_default.db should not be put on tmpfs. Are >> there any other db files that should *not* be put on tmpfs? >> >> Overview of files I have in db/ on my test box: >> >> call_limit.db >> core.db >> directory.db >> fifo.db >> sofia_reg_external.db >> sofia_reg_internal.db >> sofia_reg_internal-ipv6.db >> voicemail_default.db >> zrtp.dat >> >> >> Regards, >> Patrick >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From brian at freeswitch.org Thu Aug 19 07:47:12 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Aug 2010 09:47:12 -0500 Subject: [Freeswitch-users] Paging calls In-Reply-To: <0858DDC6-9DB9-4296-ACE3-37D8E259DE49@binarysignal.com> References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> <0858DDC6-9DB9-4296-ACE3-37D8E259DE49@binarysignal.com> Message-ID: <318B3FCB-3A30-4B8B-9FCC-14509938218E@freeswitch.org> I just don't want our default answer to be "READ THE BOOK *SMACK*" ... so lets all try not to get into that mindset ;) /b On Aug 18, 2010, at 10:35 PM, Kurt Ward wrote: > I will second that remark, it is a really good resource, and is well presented. > > > On Aug 18, 2010, at 8:14 PM, Nyamul Hassan wrote: > >> Buy the book! Would show you the light, whenever you feel lost in the dark! >> >> :) > From anthony.minessale at gmail.com Thu Aug 19 08:07:12 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Aug 2010 10:07:12 -0500 Subject: [Freeswitch-users] Paging calls In-Reply-To: <318B3FCB-3A30-4B8B-9FCC-14509938218E@freeswitch.org> References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> <0858DDC6-9DB9-4296-ACE3-37D8E259DE49@binarysignal.com> <318B3FCB-3A30-4B8B-9FCC-14509938218E@freeswitch.org> Message-ID: If we write the cookbook, then we can say "READ THE BOOK*S* *SMACK*" then it'll be ok. On Thu, Aug 19, 2010 at 9:47 AM, Brian West wrote: > I just don't want our default answer to be "READ THE BOOK *SMACK*" ... so lets all try not to get into that mindset ?;) > > /b > > On Aug 18, 2010, at 10:35 PM, Kurt Ward wrote: > >> I will second that remark, it is a really good resource, and is well presented. >> >> >> On Aug 18, 2010, at 8:14 PM, Nyamul Hassan wrote: >> >>> Buy the book! ?Would show you the light, whenever you feel lost in the dark! >>> >>> :) >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From 12ukwn at gmail.com Thu Aug 19 08:08:25 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Thu, 19 Aug 2010 17:08:25 +0200 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> Message-ID: <20100819170825.3031c24e@anubis.defcon1> Le Thu, 19 Aug 2010 09:14:55 +0600, Nyamul Hassan a ?crit : > Buy the book! Would show you the light, whenever you feel lost in the > dark! Yeah, and what about when you can't afford it? > :) No. -- "For an adequate time call 555-3321" From brian at freeswitch.org Thu Aug 19 08:11:04 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Aug 2010 10:11:04 -0500 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> <0858DDC6-9DB9-4296-ACE3-37D8E259DE49@binarysignal.com> <318B3FCB-3A30-4B8B-9FCC-14509938218E@freeswitch.org> Message-ID: <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> This is one of the perfect topics for the Cookbook... Now will be doing this in the style of Julia Childs? :P /b On Aug 19, 2010, at 10:07 AM, Anthony Minessale wrote: > If we write the cookbook, then we can say "READ THE BOOK*S* *SMACK*" > then it'll be ok. > > On Thu, Aug 19, 2010 at 9:47 AM, Brian West wrote: >> I just don't want our default answer to be "READ THE BOOK *SMACK*" ... so lets all try not to get into that mindset ;) >> >> /b From tgraziano at myitdepartment.net Thu Aug 19 08:18:49 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Thu, 19 Aug 2010 11:18:49 -0400 Subject: [Freeswitch-users] Paging calls In-Reply-To: <20100819170825.3031c24e@anubis.defcon1> References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> <20100819170825.3031c24e@anubis.defcon1> Message-ID: On Thu, Aug 19, 2010 at 11:08 AM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote: > Le Thu, 19 Aug 2010 09:14:55 +0600, > Nyamul Hassan a ?crit : > > > Buy the book! Would show you the light, whenever you feel lost in the > > dark! > > Yeah, and what about when you can't afford it? > > > :) > > No. > > polycom uses alert headers for extra functions (paging--which is unicast, and intercom). it can also be used for functions like special ringback tones when returning from things like a parking slot timeout, it really does depend on how you inject the functions. If there is a mad boss example, it probably bears looking at. > -- > "For an adequate time call 555-3321" > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/48df84bd/attachment.html From anthony.minessale at gmail.com Thu Aug 19 08:24:11 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Aug 2010 10:24:11 -0500 Subject: [Freeswitch-users] Paging calls In-Reply-To: <20100819170825.3031c24e@anubis.defcon1> References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> <20100819170825.3031c24e@anubis.defcon1> Message-ID: Then you can download the code for free and read that along with thousands of FREE pages of WIKI articles. But if you can't afford the book, you probably don't have a computer either. On Thu, Aug 19, 2010 at 10:08 AM, Jean-Yves F. Barbier <12ukwn at gmail.com> wrote: > Le Thu, 19 Aug 2010 09:14:55 +0600, > Nyamul Hassan a ?crit : > >> Buy the book! ?Would show you the light, whenever you feel lost in the >> dark! > > Yeah, and what about when you can't afford it? > >> :) > > No. > > -- > "For an adequate time call 555-3321" > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Thu Aug 19 08:24:52 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Aug 2010 10:24:52 -0500 Subject: [Freeswitch-users] Paging calls In-Reply-To: <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> <0858DDC6-9DB9-4296-ACE3-37D8E259DE49@binarysignal.com> <318B3FCB-3A30-4B8B-9FCC-14509938218E@freeswitch.org> <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> Message-ID: You mean with a bottle of sherry in one hand? naturally! On Thu, Aug 19, 2010 at 10:11 AM, Brian West wrote: > This is one of the perfect topics for the Cookbook... Now will be doing this in the style of Julia Childs? > > :P > > /b > > On Aug 19, 2010, at 10:07 AM, Anthony Minessale wrote: > >> If we write the cookbook, then we can say "READ THE BOOK*S* *SMACK*" >> then it'll be ok. >> >> On Thu, Aug 19, 2010 at 9:47 AM, Brian West wrote: >>> I just don't want our default answer to be "READ THE BOOK *SMACK*" ... so lets all try not to get into that mindset ?;) >>> >>> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From brian at freeswitch.org Thu Aug 19 08:28:15 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Aug 2010 10:28:15 -0500 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: <55D66276-05A5-4418-BC70-B4918E3CD416@freeswitch.org> <0858DDC6-9DB9-4296-ACE3-37D8E259DE49@binarysignal.com> <318B3FCB-3A30-4B8B-9FCC-14509938218E@freeswitch.org> <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> Message-ID: <94689933-4B95-4D08-AD7E-725445F44CB4@freeswitch.org> Sounds perfect. Goes well with truffle bubbles? /b On Aug 19, 2010, at 10:24 AM, Anthony Minessale wrote: > You mean with a bottle of sherry in one hand? naturally! > > On Thu, Aug 19, 2010 at 10:11 AM, Brian West wrote: >> This is one of the perfect topics for the Cookbook... Now will be doing this in the style of Julia Childs? >> >> :P >> >> /b From sean at obscuradigital.com Thu Aug 19 08:30:01 2010 From: sean at obscuradigital.com (Sean Holt) Date: Thu, 19 Aug 2010 08:30:01 -0700 Subject: [Freeswitch-users] Paging calls In-Reply-To: <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> Message-ID: Glad I was able start this discussion...... Anyways I did buy the book before it was released and I read through the mad boss example, but no success. Curious if anyone else has had success setting up this functionality using this particular 321 Polycom model? Thanks, Sean On 8/19/10 8:11 AM, "Brian West" wrote: > This is one of the perfect topics for the Cookbook... Now will be doing this > in the style of Julia Childs? > > :P > > /b > > On Aug 19, 2010, at 10:07 AM, Anthony Minessale wrote: > >> If we write the cookbook, then we can say "READ THE BOOK*S* *SMACK*" >> then it'll be ok. >> >> On Thu, Aug 19, 2010 at 9:47 AM, Brian West wrote: >>> I just don't want our default answer to be "READ THE BOOK *SMACK*" ... so >>> lets all try not to get into that mindset ;) >>> >>> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Sean Holt Director of IT +1 415 227 9979 STUDIO +1 510 207 9553 MOBILE Obscura Digital 729 Tennessee St. San Francisco, CA 94107 www.obscuradigital.com \ SAN FRANCISCO \ NEW YORK \ TOKYO \ From msc at freeswitch.org Thu Aug 19 09:06:52 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Aug 2010 09:06:52 -0700 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> Message-ID: On Thu, Aug 19, 2010 at 8:30 AM, Sean Holt wrote: > Glad I was able start this discussion...... > > Anyways I did buy the book before it was released and I read through the > mad > boss example, but no success. Curious if anyone else has had success > setting up this functionality using this particular 321 Polycom model? > I just tested this on a Polycom 320 (I don't have a 321) and it worked perfectly. I put a Snom 300 as x1001 and the Polycom 320 as x1004, which are both in the sales group. (See conf/directory/default.xml for group definitions.) I called "0911" and the Polycom immediately went to intercom mode. (The Snom 300 did not, which I suspect means it isn't viable for this particular intercom solution.) What happens when you put a Polycom 321 into the sales group and then dial 0911? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/c35fe94b/attachment.html From tgraziano at myitdepartment.net Thu Aug 19 09:25:50 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Thu, 19 Aug 2010 12:25:50 -0400 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> Message-ID: On Thu, Aug 19, 2010 at 12:06 PM, Michael Collins wrote: > > > On Thu, Aug 19, 2010 at 8:30 AM, Sean Holt wrote: > >> Glad I was able start this discussion...... >> >> Anyways I did buy the book before it was released and I read through the >> mad >> boss example, but no success. Curious if anyone else has had success >> setting up this functionality using this particular 321 Polycom model? >> > > I just tested this on a Polycom 320 (I don't have a 321) and it worked > perfectly. I put a Snom 300 as x1001 and the Polycom 320 as x1004, which are > both in the sales group. (See conf/directory/default.xml for group > definitions.) I called "0911" and the Polycom immediately went to intercom > mode. (The Snom 300 did not, which I suspect means it isn't viable for this > particular intercom solution.) > > What happens when you put a Polycom 321 into the sales group and then dial > 0911? > -MC > > > I don't have the book :>). If the call is using an alert header to do the paging I am pretty sire the snom's don't support that dialogue though... > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/a5a4738f/attachment.html From sean at obscuradigital.com Thu Aug 19 10:05:42 2010 From: sean at obscuradigital.com (Sean Holt) Date: Thu, 19 Aug 2010 10:05:42 -0700 Subject: [Freeswitch-users] Paging calls In-Reply-To: Message-ID: Hey Tony, Yeah I?m able to get one Polycom to answer but more then one doesn?t seem to work. Not sure what I?m missing. On 8/19/10 9:25 AM, "Tony Graziano" wrote: > On Thu, Aug 19, 2010 at 12:06 PM, Michael Collins wrote: >> >> >> On Thu, Aug 19, 2010 at 8:30 AM, Sean Holt wrote: >>> Glad I was able start this discussion...... >>> >>> Anyways I did buy the book before it was released and I read through the mad >>> boss example, but no success. ?Curious if anyone else has had success >>> setting up this functionality using this particular 321 Polycom model? >> >> I just tested this on a Polycom 320 (I don't have a 321) and it worked >> perfectly. I put a Snom 300 as x1001 and the Polycom 320 as x1004, which are >> both in the sales group. (See conf/directory/default.xml for group >> definitions.) I called "0911" and the Polycom immediately went to intercom >> mode. (The Snom 300 did not, which I suspect means it isn't viable for this >> particular intercom solution.) >> >> What happens when you put a Polycom 321 into the sales group and then dial >> 0911? >> -MC >> >> > I don't have the book :>). If the call is using an alert header to do the > paging I am pretty sire the snom's don't support that dialogue though... >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Sean Holt Director of IT +1 415 227 9979 STUDIO +1 510 207 9553 MOBILE Obscura Digital 729 Tennessee St. San Francisco, CA 94107 www.obscuradigital.com \ SAN FRANCISCO \ NEW YORK \ TOKYO \ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/f0941b9f/attachment-0001.html From alanlo at commverge.com Wed Aug 18 20:10:39 2010 From: alanlo at commverge.com (alan lo) Date: Thu, 19 Aug 2010 11:10:39 +0800 Subject: [Freeswitch-users] IVR menu failure/timeout action In-Reply-To: References: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> Message-ID: Thx for the answer, but I tried but it doesn?t work. Below is some of my code in my file ?v_IVR_demo.xml? under folder ?ivr_menus? The call dropped immediately after I press 1 in the phone Below is the log. Phone number and IP are masked with (MyPhone) and (MyIP) respectively. 2010-08-19 10:47:34.220750 [NOTICE] mod_dptools.c:720 Channel [sofia/external/(MyPhone)@ (MyIP)] has been answered 2010-08-19 10:47:44.892625 [NOTICE] switch_ivr.c:1450 Transfer sofia/external/(MyPhone)@ (MyIP) to XML[submenu at public] 2010-08-19 10:47:44.892625 [NOTICE] switch_cpp.cpp:597 Hangup sofia/external/(MyPhone)@ (MyIP) [CS_ROUTING] [NORMAL_CLEARING] 2010-08-19 10:47:44.892625 [NOTICE] switch_core_session.c:1188 Session 6 (sofia/external/(MyPhone)@ (MyIP)) Ended 2010-08-19 10:47:44.892625 [NOTICE] switch_core_session.c:1190 Close Channel sofia/external/(MyPhone)@ (MyIP) [CS_DESTROY] Alan _____ From: David Ponzone [mailto:david.ponzone at ipeva.fr] Sent: Wednesday, August 18, 2010 11:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] IVR menu failure/timeout action I guess if you use: transfer sub_menu_1 XML it will stay in the current context. And if you use: transfer sub_menu_1 XML foo it will transfer to context foo. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 18/08/2010 ? 11:39, alan lo a ?crit : Dear all, Referring to the threat below, where should the dp extension stored? In the same XML of ivr or should store it in other locations? Thx! Alan I have confirmed this behavior in the latest git as of last night. I don't know if this is by design or not. I would open JIRA as a feature request to have this behavior modified. The other alternative would be to have each IVR submenu called via transfer instead of menu-sub: Then have a dp extension to handle that: -MC On Mon, Jul 19, 2010 at 2:37 AM, Raymond Chan >wrote: > Hi all, > > > > I am experiencing an IVR problem. Except the top menu, all other sub menu > will not hang up after reached max invalid input or timeout limit. It will > go back up upper menu. I want it to load the exit-sound and then hang the > call when it reached max invalid input or timeout limit. Do you tell how to > configure? > > > > > > greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" > > > greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" > > > invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" > > exit-sound="voicemail/vm-goodbye-m.wav" > > timeout="5000" > > max-failures="2"> > > > > > > > > > > Thanks > > > > Raymond _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/9ccfd7e4/attachment-0001.html From chat2jesse at gmail.com Wed Aug 18 17:05:50 2010 From: chat2jesse at gmail.com (jesse zhao) Date: Wed, 18 Aug 2010 17:05:50 -0700 Subject: [Freeswitch-users] FS and Gtalk integration Message-ID: hi, help: I searched online pages and FS wiki , basically the doc about Gtalk and FS is pretty lousy. I still couldn't get my gtalk and FS work together. two major issues: 1) when I call gtalk(gtalk=XYZ) from SIP phone, call is established, but no audio. 2) how could I call SIP from gtalk ? when I request chat to sip, gchat in gmail requests an invitation. suppose the voice request is : conf+1000 at 172.18.115.73 here are my config files: jingle_profiles/client.xml dialplan/default.xml I tried suggestion from http://lists.freeswitch.org/pipermail/freeswitch-users/2007-February/028456.html, above two issues still not work. I heard id: intralanman is good at this, if you saw the post, please share your advice. -jesse From mike at jerris.com Thu Aug 19 10:15:35 2010 From: mike at jerris.com (Michael Jerris) Date: Thu, 19 Aug 2010 13:15:35 -0400 Subject: [Freeswitch-users] Some build 32 bit binary on 64bit Mac In-Reply-To: References: Message-ID: <5C4FF74D-36E8-48BB-BC08-50566D393131@jerris.com> we should fix this in configure so if you are on 64 bit apple and pass --host=i386 it sets those other flags for you. Can you provide a patch for this? Mike On Aug 19, 2010, at 3:14 AM, Seven Du wrote: > I think this is the correct way. > > CFLAGS=-m32 CXXFLAGS=-m32 LDFLAGS=-m32 ./configure > --prefix=/opt/fs-i386 --host=i386 > > Thanks MikeJ and Math on irc. > > btw, @Math, I dont' use --build=i386-apple-darwin10, it seems throws errors. > > On Thu, Aug 12, 2010 at 8:20 PM, Seven Du wrote: >> There are many reasons I need a 32bit lib: >> >> 1) 64bit app cannot load flash in webkit in QT(at least in my case). >> I'm try to build a client with libfreeswitch and flash in. >> >> 2) I want the lib run in iPhone simulator. Eventually on ARM on iphone. >> >> 3) If you release prebuilt desktop binary, you'd better don't forget >> 32 bit users. >> >> 4) perhaps more... >> >> >> On Thu, Aug 12, 2010 at 7:33 PM, Steven Ayre wrote: >>> If you're on 64bit, why do you want to compile as 32bit? >>> >>> -Steve >>> >>> >>> >>> On 12 August 2010 11:36, Seven Du wrote: >>>> >>>> I'm on Mac 10.6.4 64bit, a fresh clone of git head. I tried to build >>>> FS into 32 bit. >>>> >>>> I use the following command when configure: >>>> >>>> CFLAGS="-arch i386" CXXFLAGS="-arch i386" LDFLAGS="-arch i386" >>>> ./configure --prefix=/opt/fs-i386 --host=i386 --build=i386 >>>> --target=i386 >>>> >>>> >>>> However, when make, it throws some errors, I manually *change* the >>>> source like from "ifdef" to "ifndef" to trick it and make further, >>>> however, I still couldn't get fully built. >>>> >>>> some errors below. and ideas? thanks. >>>> >>>> >>>> >>>> src/switch_apr.c: In function 'switch_vasprintf': >>>> src/switch_apr.c:1024: warning: implicit declaration of function >>>> 'vasprintf' >>>> >>>> >>>> >>>> src/switch_core_session.c: In function 'switch_core_session_thread': >>>> src/switch_core_session.c:1193: warning: format '%d' expects type >>>> 'int', but argument 8 has type 'switch_size_t' >>>> src/switch_core_session.c:1203: warning: format '%d' expects type >>>> 'int', but argument 8 has type 'switch_size_t' >>>> src/switch_core_session.c:1221: warning: format '%d' expects type >>>> 'int', but argument 8 has type 'switch_size_t' >>>> >>>> >>>> src/switch_core.c: In function 'send_heartbeat': >>>> src/switch_core.c:84: warning: format '%d' expects type 'int', but >>>> argument 5 has type 'long unsigned int' >>>> src/switch_core.c: In function 'change_user_group': >>>> src/switch_core.c:677: warning: implicit declaration of function >>>> 'setgroups' >>>> src/switch_core.c:704: warning: implicit declaration of function >>>> 'initgroups' >>>> >>>> >>>> >>>> >>>> src/switch_core.c: In function 'send_heartbeat': >>>> src/switch_core.c:84: warning: format '%d' expects type 'int', but >>>> argument 5 has type 'long unsigned int' >>>> src/switch_core.c: In function 'change_user_group': >>>> src/switch_core.c:677: warning: implicit declaration of function >>>> 'setgroups' >>>> src/switch_core.c:704: warning: implicit declaration of function >>>> 'initgroups' >>>> >>>> >>>> >>>> >>>> >>>> src/switch_rtp.c: In function 'rtp_common_write': >>>> src/switch_rtp.c:3437: error: 'u_long' undeclared (first use in this >>>> function) >>>> src/switch_rtp.c:3437: error: (Each undeclared identifier is reported only >>>> once >>>> src/switch_rtp.c:3437: error: for each function it appears in.) >>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>> make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>>> >>>> >>>> >>>> >>>> >>>> src/switch_utils.c: In function 'switch_build_uri': >>>> src/switch_utils.c:1530: error: 'NI_MAXHOST' undeclared (first use in >>>> this function) >>>> src/switch_utils.c:1530: error: (Each undeclared identifier is >>>> reported only once >>>> src/switch_utils.c:1530: error: for each function it appears in.) >>>> src/switch_utils.c:1530: error: 'NI_MAXSERV' undeclared (first use in >>>> this function) >>>> >>>> >>>> >>>> more detailed log: >>>> >>>> http://pastebin.freeswitch.org/13615 >>>> >>>> >>>> >>>> -- >>>> Blog: http://www.dujinfang.com >>>> Proj: http://www.freeswitch.org.cn >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> >> -- >> Blog: http://www.dujinfang.com >> Proj: http://www.freeswitch.org.cn >> > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tgraziano at myitdepartment.net Thu Aug 19 10:25:36 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Thu, 19 Aug 2010 13:25:36 -0400 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: Message-ID: On Thu, Aug 19, 2010 at 1:05 PM, Sean Holt wrote: > Hey Tony, > > Yeah I?m able to get one Polycom to answer but more then one doesn?t seem > to work. Not sure what I?m missing. > the method polycom uses is UNICAST. This means slow packets might mean a missed target, and probably a max amount of devices in the group. the more in the group the more likely it wont work well. I have never had success with more than 10 devices on a segregated vlan (i dont use FS as anything other than media right now). If you need to cover a large area or enough volume, a sip paging system is better suited. The polycom paging 'application" as it is, is very limited in what it is able to do. > > > On 8/19/10 9:25 AM, "Tony Graziano" wrote: > > On Thu, Aug 19, 2010 at 12:06 PM, Michael Collins > wrote: > > > > On Thu, Aug 19, 2010 at 8:30 AM, Sean Holt > wrote: > > Glad I was able start this discussion...... > > Anyways I did buy the book before it was released and I read through the > mad > boss example, but no success. Curious if anyone else has had success > setting up this functionality using this particular 321 Polycom model? > > > I just tested this on a Polycom 320 (I don't have a 321) and it worked > perfectly. I put a Snom 300 as x1001 and the Polycom 320 as x1004, which are > both in the sales group. (See conf/directory/default.xml for group > definitions.) I called "0911" and the Polycom immediately went to intercom > mode. (The Snom 300 did not, which I suspect means it isn't viable for this > particular intercom solution.) > > What happens when you put a Polycom 321 into the sales group and then dial > 0911? > -MC > > > I don't have the book :>). If the call is using an alert header to do the > paging I am pretty sire the snom's don't support that dialogue though... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/7edf15f2/attachment.html From anthony.minessale at gmail.com Thu Aug 19 10:31:48 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Aug 2010 12:31:48 -0500 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> Message-ID: The book is not really part of this discusion. Some people just mentioned it in passing. The examples are in the default config and on the wiki and as Michael pointed out, they work. On Thu, Aug 19, 2010 at 11:25 AM, Tony Graziano wrote: > On Thu, Aug 19, 2010 at 12:06 PM, Michael Collins > wrote: >> >> >> On Thu, Aug 19, 2010 at 8:30 AM, Sean Holt >> wrote: >>> >>> Glad I was able start this discussion...... >>> >>> Anyways I did buy the book before it was released and I read through the >>> mad >>> boss example, but no success. ?Curious if anyone else has had success >>> setting up this functionality using this particular 321 Polycom model? >> >> I just tested this on a Polycom 320 (I don't have a 321) and it worked >> perfectly. I put a Snom 300 as x1001 and the Polycom 320 as x1004, which are >> both in the sales group. (See conf/directory/default.xml for group >> definitions.) I called "0911" and the Polycom immediately went to intercom >> mode. (The Snom 300 did not, which I suspect means it isn't viable for this >> particular intercom solution.) >> >> What happens when you put a Polycom 321 into the sales group and then dial >> 0911? >> -MC >> >> > I don't have the book :>). If the call is using an alert header to do the > paging I am pretty sire the snom's don't support that dialogue though... >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From neilp at cs.stanford.edu Thu Aug 19 10:41:33 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 19 Aug 2010 23:11:33 +0530 Subject: [Freeswitch-users] bootstrap.sh hanging on Message-ID: Hi All, bootstrap.sh is hanging for me while running on vanilla Ubuntu 9.04 at this point: automake: compiling `check_dlopen_sofia.c' with per-target flags requires `AM_PROG_CC_C_O' in `configure.ac' tests/Makefile.am:37: while processing program `check_dlopen_sofia' Just sits here until I hit enter and the script exits. I also tried running make from libs/sofia-sip/tests, but that doesn't work. I am running the latest from git. Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/e58aa537/attachment.html From msc at freeswitch.org Thu Aug 19 10:42:58 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Aug 2010 10:42:58 -0700 Subject: [Freeswitch-users] Paging calls In-Reply-To: References: <3C1CBFC9-595C-4FBB-8008-9F2B3D531B54@freeswitch.org> Message-ID: On Thu, Aug 19, 2010 at 10:31 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The book is not really part of this discusion. Some people just > mentioned it in passing. > The examples are in the default config and on the wiki and as Michael > pointed out, they work. > FYI, I just did this with two Poly 320's and they both went into intercom mode the instant I dialed 0911. I had them set as x1003 and x1004. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/7aa3c004/attachment-0001.html From msc at freeswitch.org Thu Aug 19 10:46:02 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Aug 2010 10:46:02 -0700 Subject: [Freeswitch-users] IVR menu failure/timeout action In-Reply-To: References: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> Message-ID: Alan, Please collect a debug trace of the call failing. Use "console loglevel debug" at the CLI. Also, use pastebin for the log so that this thread doesn't get too long. http://pastebin:freeswitch at pastebin.freeswitch.org/ After you have pastebin'd the logs please give us the URL of the pb entry and we'll have a look. Thanks, MC On Wed, Aug 18, 2010 at 8:10 PM, alan lo wrote: > Thx for the answer, but I tried but it doesn?t work. > > Below is some of my code in my file ?v_IVR_demo.xml? under folder > ?ivr_menus? > > ** > > * * > > * * > > * * > > ** > > > > ** > > > > The call dropped immediately after I press 1 in the phone > > Below is the log. Phone number and IP are masked with (MyPhone) and (MyIP) > respectively. > > 2010-08-19 10:47:34.220750 [NOTICE] mod_dptools.c:720 Channel > [sofia/external/(MyPhone)@ (MyIP)] has been answered > > 2010-08-19 10:47:44.892625 [NOTICE] switch_ivr.c:1450 Transfer > sofia/external/(MyPhone)@ (MyIP) to XML[submenu at public] > > 2010-08-19 10:47:44.892625 [NOTICE] switch_cpp.cpp:597 Hangup > sofia/external/(MyPhone)@ (MyIP) [CS_ROUTING] [NORMAL_CLEARING] > > 2010-08-19 10:47:44.892625 [NOTICE] switch_core_session.c:1188 Session 6 > (sofia/external/(MyPhone)@ (MyIP)) Ended > > 2010-08-19 10:47:44.892625 [NOTICE] switch_core_session.c:1190 Close > Channel sofia/external/(MyPhone)@ (MyIP) [CS_DESTROY] > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/7e65798b/attachment.html From chat2jesse at gmail.com Thu Aug 19 11:07:50 2010 From: chat2jesse at gmail.com (jesse zhao) Date: Thu, 19 Aug 2010 11:07:50 -0700 Subject: [Freeswitch-users] FS and Gtalk integration In-Reply-To: References: Message-ID: Any expert on this issue? -jesse On Wed, Aug 18, 2010 at 5:05 PM, jesse zhao wrote: > hi, help: > > ?I searched online pages and FS wiki , basically the doc about Gtalk > and FS is pretty lousy. I still couldn't get > my gtalk and FS work together. > > ? two major issues: > > ?1) when I call gtalk(gtalk=XYZ) from SIP phone, call is established, > but no audio. > ?2) how could I call SIP from gtalk ? ?when I request chat to sip, > gchat in gmail requests an invitation. > ? ? suppose the voice request is : conf+1000 at 172.18.115.73 > > ?here are my config files: > ?jingle_profiles/client.xml > > ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? ? > ? > > > > ?dialplan/default.xml > > ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? > > ? ? > ? ? ? ? ? > ? ? ? ? ? expression="^gtalk=([a-zA-z0-9.-]+)$"> > ? ? ? ? ? ? > ? ? ? ? ? ? data="effective_caller_id_number=myuser at gmail.com"/> > ? ? ? ? ? ? data="dingaling/gmail.com/$1 at gmail.com"/> > ? ? ? ? ? > > I tried suggestion from > http://lists.freeswitch.org/pipermail/freeswitch-users/2007-February/028456.html, > above two issues still not work. > > ?I heard id: intralanman is good at this, if you saw the post, please > share your advice. > > -jesse > From david.ponzone at ipeva.fr Thu Aug 19 11:13:07 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 19 Aug 2010 20:13:07 +0200 Subject: [Freeswitch-users] bootstrap.sh hanging on In-Reply-To: References: Message-ID: <410E4BB1-F8E5-402A-92D3-2AE52C106951@ipeva.fr> is that on a newly cloned git tree ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 19/08/2010 ? 19:41, Neil Patel a ?crit : > Hi All, > > bootstrap.sh is hanging for me while running on vanilla Ubuntu 9.04 > at this point: > > automake: compiling `check_dlopen_sofia.c' with per-target flags > requires `AM_PROG_CC_C_O' in `configure.ac' > tests/Makefile.am:37: while processing program `check_dlopen_sofia' > > Just sits here until I hit enter and the script exits. I also tried > running make from libs/sofia-sip/tests, but that doesn't work. > > I am running the latest from git. > > Thanks, > Neil > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/393d8dc8/attachment.html From sean at obscuradigital.com Thu Aug 19 11:19:19 2010 From: sean at obscuradigital.com (Sean Holt) Date: Thu, 19 Aug 2010 11:19:19 -0700 Subject: [Freeswitch-users] Paging calls In-Reply-To: Message-ID: Thanks everyone. Sean On 8/19/10 10:42 AM, "Michael Collins" wrote: > > > On Thu, Aug 19, 2010 at 10:31 AM, Anthony Minessale > wrote: >> The book is not really part of this discusion. ?Some people just >> mentioned it in passing. >> The examples are in the default config and on the wiki and as Michael >> pointed out, they work. > FYI, > > I just did this with two Poly 320's and they both went into intercom mode the > instant I dialed 0911. I had them set as x1003 and x1004. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Sean Holt Director of IT +1 415 227 9979 STUDIO +1 510 207 9553 MOBILE Obscura Digital 729 Tennessee St. San Francisco, CA 94107 www.obscuradigital.com \ SAN FRANCISCO \ NEW YORK \ TOKYO \ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100819/3bd14dad/attachment-0001.html From stephen at stephenjc.com Thu Aug 19 11:27:53 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Thu, 19 Aug 2010 14:27:53 -0400 Subject: [Freeswitch-users] nosmp timing issue Message-ID: Just as an FYI, on a Single Processor without hyper threading or multiple cores adding nosmp to the kernel options fixed my inaccurate time keeping. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print From brokendash at gmail.com Thu Aug 19 18:02:53 2010 From: brokendash at gmail.com (broken dash) Date: Thu, 19 Aug 2010 20:02:53 -0500 Subject: [Freeswitch-users] playback options Message-ID: I'm pulling shout cast streams into freeswitch using the playback action and I'm wondering if there is a variable like playback_delimeter that I could set within my dialplan/scripts that would essentially load up the stream and chop off a configurable amount of time before it essentially bridges up the audio to the caller? or perhaps there is a dialplan routine that could do this for me? Brian From alanlo at commverge.com Thu Aug 19 20:05:21 2010 From: alanlo at commverge.com (alan lo) Date: Fri, 20 Aug 2010 11:05:21 +0800 Subject: [Freeswitch-users] IVR menu failure/timeout action In-Reply-To: References: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> Message-ID: MC I have put the log on the following URL. http://pastebin.freeswitch.org/13682 Please help check it! Thanks. Alan _____ From: Michael Collins [mailto:msc at freeswitch.org] Sent: Friday, August 20, 2010 1:46 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] IVR menu failure/timeout action Alan, Please collect a debug trace of the call failing. Use "console loglevel debug" at the CLI. Also, use pastebin for the log so that this thread doesn't get too long. http://pastebin:freeswitch at pastebin.freeswitch.org/ After you have pastebin'd the logs please give us the URL of the pb entry and we'll have a look. Thanks, MC On Wed, Aug 18, 2010 at 8:10 PM, alan lo wrote: Thx for the answer, but I tried but it doesn???t work. Below is some of my code in my file ???v_IVR_demo.xml??? under folder ???ivr_menus??? ? ? ? ? ? ? ? The call dropped immediately after I press 1 in the phone Below is the log. Phone number and IP are masked with (MyPhone) and (MyIP) respectively. 2010-08-19 10:47:34.220750 [NOTICE] mod_dptools.c:720 Channel [sofia/external/(MyPhone)@ (MyIP)] has been answered 2010-08-19 10:47:44.892625 [NOTICE] switch_ivr.c:1450 Transfer sofia/external/(MyPhone)@ (MyIP) to XML[submenu at public] 2010-08-19 10:47:44.892625 [NOTICE] switch_cpp.cpp:597 Hangup sofia/external/(MyPhone)@ (MyIP) [CS_ROUTING] [NORMAL_CLEARING] 2010-08-19 10:47:44.892625 [NOTICE] switch_core_session.c:1188 Session 6 (sofia/external/(MyPhone)@ (MyIP)) Ended 2010-08-19 10:47:44.892625 [NOTICE] switch_core_session.c:1190 Close Channel sofia/external/(MyPhone)@ (MyIP) [CS_DESTROY] ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/18e7833d/attachment.html From dujinfang at gmail.com Thu Aug 19 20:30:15 2010 From: dujinfang at gmail.com (Seven Du) Date: Fri, 20 Aug 2010 11:30:15 +0800 Subject: [Freeswitch-users] Some build 32 bit binary on 64bit Mac In-Reply-To: <5C4FF74D-36E8-48BB-BC08-50566D393131@jerris.com> References: <5C4FF74D-36E8-48BB-BC08-50566D393131@jerris.com> Message-ID: Hi Mike, I would love to, but I couldn't figure out how to make a patch. Actually, I pulled a fresh git and run that configure against, still get all those errors. I tried my old code base again, I reverted all code changes, git pull, and can still build ok with the following diff and make mega clean && make. also I cleaned what I can think about: cd freeswitch/libs (cd esl ; make clean) (cd ilbc; make clean) (cd broadvoice; make clean) (cd apr; make clean) (cd apr-util; make clean) (cd js; make clean) (cd tiff-3.8.2; make clean) (cd srtp; make clean) (cd sqlite; make clean) (cd speex; make clean) (cd spandsp; make clean) (cd sofia-sip; make clean) (cd pcre; make clean) (cd libsndfile; make clean) (cd libedit; make clean) (cd apr-util/xml/expat; make clean) (cd js/nsprpub; make clean) (find js/ -name "*.o" -exec rm -f {} \;) (cd libg722_1; make clean) cd .. (cd src/mod/languages/mod_lua/lua; make clean) seven at seven-macpro:~/workspace/freeswitch/i386/freeswitch$ git status # On branch master # Changed but not updated: # (use "git add/rm ..." to update what will be committed) # (use "git checkout -- ..." to discard changes in working directory) # # modified: libs/esl/Makefile # deleted: libs/js/libtool.m4 # deleted: libs/js/shtool diff --git a/libs/esl/Makefile b/libs/esl/Makefile index a180406..a3c1829 100644 --- a/libs/esl/Makefile +++ b/libs/esl/Makefile @@ -4,11 +4,11 @@ LIBEDIT_DIR=../../libs/libedit DEBUG=-g -ggdb BASE_FLAGS=$(INCS) -DHAVE_EDITLINE $(DEBUG) -I$(LIBEDIT_DIR)/src/ -fPIC PICKY=-O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmis -CFLAGS=$(BASE_FLAGS) $(PICKY) -CXXFLAGS=$(BASE_FLAGS) -Wall -Werror -Wno-unused-variable +CFLAGS=$(BASE_FLAGS) $(PICKY) -m32 +CXXFLAGS=$(BASE_FLAGS) -Wall -Werror -Wno-unused-variable -m32 MYLIB=libesl.a LIBS=-lncurses -lpthread -lesl -lm -LDFLAGS=-L. +LDFLAGS=-L. -m32 OBJS=src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o src/esl_json.o SRC=src/esl.c src/esl_json.c src/esl_event.c src/esl_threadmutex.c src/esl_config.c src/esl_oop HEADERS=src/include/esl_config.h src/include/esl_event.h src/include/esl.h src/include/esl_thre I diff-ed configure and Makefile and src/Makefile, they are identical between old code base and new clean pulled. Getting lost. On Fri, Aug 20, 2010 at 1:15 AM, Michael Jerris wrote: > we should fix this in configure so if you are on 64 bit apple and pass --host=i386 it sets those other flags for you. ?Can you provide a patch for this? > > Mike > > On Aug 19, 2010, at 3:14 AM, Seven Du wrote: > >> I think this is the correct way. >> >> CFLAGS=-m32 CXXFLAGS=-m32 LDFLAGS=-m32 ./configure >> --prefix=/opt/fs-i386 --host=i386 >> >> Thanks MikeJ and Math on irc. >> >> btw, @Math, I dont' use --build=i386-apple-darwin10, it seems throws errors. >> >> On Thu, Aug 12, 2010 at 8:20 PM, Seven Du wrote: >>> There are many reasons I need a 32bit lib: >>> >>> 1) 64bit app cannot load flash in webkit in QT(at least in my case). >>> I'm try to build a client with libfreeswitch and flash in. >>> >>> 2) I want the lib run in iPhone simulator. Eventually on ARM on iphone. >>> >>> 3) If you release prebuilt desktop binary, you'd better don't forget >>> 32 bit users. >>> >>> 4) perhaps more... >>> >>> >>> On Thu, Aug 12, 2010 at 7:33 PM, Steven Ayre wrote: >>>> If you're on 64bit, why do you want to compile as 32bit? >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 12 August 2010 11:36, Seven Du wrote: >>>>> >>>>> I'm on Mac 10.6.4 64bit, a fresh clone of git head. I tried to build >>>>> FS into 32 bit. >>>>> >>>>> I use the following command when configure: >>>>> >>>>> CFLAGS="-arch i386" CXXFLAGS="-arch i386" LDFLAGS="-arch i386" >>>>> ./configure --prefix=/opt/fs-i386 --host=i386 --build=i386 >>>>> --target=i386 >>>>> >>>>> >>>>> However, when make, it throws some errors, I manually *change* the >>>>> source like from "ifdef" to "ifndef" to trick it and make further, >>>>> however, I still couldn't get fully built. >>>>> >>>>> some errors below. and ideas? thanks. >>>>> >>>>> >>>>> >>>>> src/switch_apr.c: In function 'switch_vasprintf': >>>>> src/switch_apr.c:1024: warning: implicit declaration of function >>>>> 'vasprintf' >>>>> >>>>> >>>>> >>>>> src/switch_core_session.c: In function 'switch_core_session_thread': >>>>> src/switch_core_session.c:1193: warning: format '%d' expects type >>>>> 'int', but argument 8 has type 'switch_size_t' >>>>> src/switch_core_session.c:1203: warning: format '%d' expects type >>>>> 'int', but argument 8 has type 'switch_size_t' >>>>> src/switch_core_session.c:1221: warning: format '%d' expects type >>>>> 'int', but argument 8 has type 'switch_size_t' >>>>> >>>>> >>>>> src/switch_core.c: In function 'send_heartbeat': >>>>> src/switch_core.c:84: warning: format '%d' expects type 'int', but >>>>> argument 5 has type 'long unsigned int' >>>>> src/switch_core.c: In function 'change_user_group': >>>>> src/switch_core.c:677: warning: implicit declaration of function >>>>> 'setgroups' >>>>> src/switch_core.c:704: warning: implicit declaration of function >>>>> 'initgroups' >>>>> >>>>> >>>>> >>>>> >>>>> src/switch_core.c: In function 'send_heartbeat': >>>>> src/switch_core.c:84: warning: format '%d' expects type 'int', but >>>>> argument 5 has type 'long unsigned int' >>>>> src/switch_core.c: In function 'change_user_group': >>>>> src/switch_core.c:677: warning: implicit declaration of function >>>>> 'setgroups' >>>>> src/switch_core.c:704: warning: implicit declaration of function >>>>> 'initgroups' >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> src/switch_rtp.c: In function 'rtp_common_write': >>>>> src/switch_rtp.c:3437: error: 'u_long' undeclared (first use in this >>>>> function) >>>>> src/switch_rtp.c:3437: error: (Each undeclared identifier is reported only >>>>> once >>>>> src/switch_rtp.c:3437: error: for each function it appears in.) >>>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>>> src/switch_rtp.c:3437: error: expected ')' before 'rtp_session' >>>>> make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> src/switch_utils.c: In function 'switch_build_uri': >>>>> src/switch_utils.c:1530: error: 'NI_MAXHOST' undeclared (first use in >>>>> this function) >>>>> src/switch_utils.c:1530: error: (Each undeclared identifier is >>>>> reported only once >>>>> src/switch_utils.c:1530: error: for each function it appears in.) >>>>> src/switch_utils.c:1530: error: 'NI_MAXSERV' undeclared (first use in >>>>> this function) >>>>> >>>>> >>>>> >>>>> more detailed log: >>>>> >>>>> http://pastebin.freeswitch.org/13615 >>>>> >>>>> >>>>> >>>>> -- >>>>> Blog: http://www.dujinfang.com >>>>> Proj: ?http://www.freeswitch.org.cn >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> >>> -- >>> Blog: http://www.dujinfang.com >>> Proj: ?http://www.freeswitch.org.cn >>> >> >> >> >> -- >> Blog: http://www.dujinfang.com >> Proj: ?http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From neilp at cs.stanford.edu Thu Aug 19 20:59:17 2010 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 20 Aug 2010 09:29:17 +0530 Subject: [Freeswitch-users] bootstrap.sh hanging on In-Reply-To: <410E4BB1-F8E5-402A-92D3-2AE52C106951@ipeva.fr> References: <410E4BB1-F8E5-402A-92D3-2AE52C106951@ipeva.fr> Message-ID: Yes, as per these instructions . On Thu, Aug 19, 2010 at 11:43 PM, David Ponzone wrote: > is that on a newly cloned git tree ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 19/08/2010 ? 19:41, Neil Patel a ?crit : > > Hi All, > > bootstrap.sh is hanging for me while running on vanilla Ubuntu 9.04 at this > point: > > automake: compiling `check_dlopen_sofia.c' with per-target flags requires > `AM_PROG_CC_C_O' in `configure.ac' > tests/Makefile.am:37: while processing program `check_dlopen_sofia' > > Just sits here until I hit enter and the script exits. I also tried running > make from libs/sofia-sip/tests, but that doesn't work. > > I am running the latest from git. > > Thanks, > Neil > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/c0b87cbb/attachment.html From woodydickson at gmail.com Fri Aug 20 02:29:31 2010 From: woodydickson at gmail.com (Woody Dickson) Date: Fri, 20 Aug 2010 17:29:31 +0800 Subject: [Freeswitch-users] freeswitch CPU usage Message-ID: Hi, I am doing some experiments with Freeswitch by torturing it to see how the machine's CPU response to heavy loaded situation. The test is done on a 16 core 5550 dual quad core server running fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. What I found so strange was that while CPU usage remains pretty low and distributed among all cores at 190 - 200 calls per second. Then, after added a few more calls per second, all CPU becomes fully utilized. Is this due to some wrong setting? Any idea how I can tweak the configuration and continue my test? Thanks, Woody From 12ukwn at gmail.com Fri Aug 20 03:25:42 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 12:25:42 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: <20100820122542.7e58117e@anubis.defcon1> Le Fri, 20 Aug 2010 17:29:31 +0800, Woody Dickson a ?crit : Hi Woody, > I am doing some experiments with Freeswitch by torturing it to see how > the machine's CPU response to heavy loaded situation. > The test is done on a 16 core 5550 dual quad core server running > fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. nice beast > What I found so strange was that while CPU usage remains pretty low > and distributed among all cores at 190 - 200 calls per second. Then, > after added a few more calls per second, all CPU becomes fully > utilized. > > Is this due to some wrong setting? Any idea how I can tweak the > configuration and continue my test? AFAIK this is not normal, I spend a lot of time seeking for a scalable solution that would fit either personal or large company phone system and thus read a lot of docs & tests about VoIP - most of them show a quasi-linear curve for cps, except when you're close to overloading, which isn't even close for your machine. JY -- [Babe] Ruth made a big mistake when he gave up pitching. -- Tris Speaker, 1921 From david.ponzone at ipeva.fr Fri Aug 20 03:36:46 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Fri, 20 Aug 2010 12:36:46 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: Have you followed all the recommendations for best performances ? Sorry if that sounds obvious to you but things like: ulimit -s 240 reducing core loglevel to warning increasing tx/rx buffer of the ethernet to 4096 Are you hitting only one SIP profie with your stress test ? Remember there is only one thread per SIP profile. But some people told me recently on #freeswitch that they had achieved close to 300cps on only one SIP profile. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 20/08/2010 ? 11:29, Woody Dickson a ?crit : > Hi, > > I am doing some experiments with Freeswitch by torturing it to see how > the machine's CPU response to heavy loaded situation. > The test is done on a 16 core 5550 dual quad core server running > fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. > > What I found so strange was that while CPU usage remains pretty low > and distributed among all cores at 190 - 200 calls per second. Then, > after added a few more calls per second, all CPU becomes fully > utilized. > > Is this due to some wrong setting? Any idea how I can tweak the > configuration and continue my test? > > Thanks, > Woody > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/aed74559/attachment.html From 12ukwn at gmail.com Fri Aug 20 03:55:43 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 12:55:43 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: <20100820125543.2c462087@anubis.defcon1> Le Fri, 20 Aug 2010 12:36:46 +0200, David Ponzone a ?crit : ... > But some people told me recently on #freeswitch that they had achieved > close to 300cps on only one SIP profile. Hi Dave, do you remember on what kind of CPU (& RAM qty) it was? JY -- How to Raise Your I.Q. by Eating Gifted Children -- Book title by Lewis B. Frumkes From david.ponzone at ipeva.fr Fri Aug 20 04:22:24 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Fri, 20 Aug 2010 13:22:24 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <20100820125543.2c462087@anubis.defcon1> References: <20100820125543.2c462087@anubis.defcon1> Message-ID: <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> Can't remember. I really think that the people with interesting perfs data, if they don't consider their data as confidential, should share that on the wiki. Everybody around is asking for scalability data, and there is no real source of information. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 20/08/2010 ? 12:55, Jean-Yves F. Barbier a ?crit : > Le Fri, 20 Aug 2010 12:36:46 +0200, > David Ponzone a ?crit : > > ... >> But some people told me recently on #freeswitch that they had >> achieved >> close to 300cps on only one SIP profile. > > Hi Dave, do you remember on what kind of CPU (& RAM qty) it was? > > JY > -- > How to Raise Your I.Q. by Eating Gifted Children > -- Book title by Lewis B. Frumkes > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/71505b8c/attachment.html From 12ukwn at gmail.com Fri Aug 20 04:47:46 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 13:47:46 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> Message-ID: <20100820134746.5f95e9dc@anubis.defcon1> Le Fri, 20 Aug 2010 13:22:24 +0200, David Ponzone a ?crit : > Can't remember. > > I really think that the people with interesting perfs data, if they > don't consider their data as confidential, should share that on the > wiki. > Everybody around is asking for scalability data, and there is no real > source of information. Hey good idea! - as far as modus operandi is fully exposed. -- Jesus Saves, Moses Invests, But only Buddha pays Dividends. From woodydickson at gmail.com Fri Aug 20 05:34:13 2010 From: woodydickson at gmail.com (Woody Dickson) Date: Fri, 20 Aug 2010 20:34:13 +0800 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: These are all set correctly. I just tried some older version ( 1.0.4 ) and found that I could run up to 600 cps. I am very surprised that the performance seems to be degraded with the newer version. On Fri, Aug 20, 2010 at 6:36 PM, David Ponzone wrote: > Have you followed all the recommendations for best performances ? > Sorry if that sounds obvious to you but things like: > ulimit -s 240 > reducing core loglevel to warning > increasing tx/rx buffer of the ethernet to 4096 > Are you hitting only one SIP profie with your stress test ? > Remember there is only one thread per SIP profile. > But some people told me recently on #freeswitch that they had achieved close > to 300cps on only one SIP profile. > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 20/08/2010 ? 11:29, Woody Dickson a ?crit : > > Hi, > > I am doing some experiments with Freeswitch by torturing it to see how > the machine's CPU response to heavy loaded situation. > The test is done on a 16 core 5550 dual quad core server running > fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. > > What I found so strange was that while CPU usage remains pretty low > and distributed among all cores at 190 - 200 calls per second. ?Then, > after added a few more calls per second, all CPU becomes fully > utilized. > > Is this due to some wrong setting? ?Any idea how I can tweak the > configuration and continue my test? > > Thanks, > Woody > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From david.ponzone at ipeva.fr Fri Aug 20 06:37:22 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Fri, 20 Aug 2010 15:37:22 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <20100820134746.5f95e9dc@anubis.defcon1> References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <20100820134746.5f95e9dc@anubis.defcon1> Message-ID: I think it should be doable as a table. I created a template table here: http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#Some_random_results If you think something is missing, please feel free to tell me or update the table yourself. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 20/08/2010 ? 13:47, Jean-Yves F. Barbier a ?crit : > Le Fri, 20 Aug 2010 13:22:24 +0200, > David Ponzone a ?crit : > >> Can't remember. >> >> I really think that the people with interesting perfs data, if they >> don't consider their data as confidential, should share that on the >> wiki. >> Everybody around is asking for scalability data, and there is no real >> source of information. > > Hey good idea! - as far as modus operandi is fully exposed. > > -- > Jesus Saves, > Moses Invests, > But only Buddha pays Dividends. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/a73ca303/attachment-0001.html From doddlephone at gmail.com Fri Aug 20 06:44:11 2010 From: doddlephone at gmail.com (Doddle WebPhone) Date: Fri, 20 Aug 2010 10:44:11 -0300 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: You can also use SIP and RTP: http://widget.doddlephone.com/ (web driven phone) Sergio On Thu, Aug 12, 2010 at 5:01 AM, Dennis wrote: > it sounds as if there is a way to stream audio from fs over red5 to a > flash-player. > > could someone explain how it is done? we do not want any telephony or > any bi-directional things. we just need a simple one-way solution to > listen to calls. > > in the moment we stream like this: fs -> icecast2 -> flash > audio-player (which makes it a http stream) > > we would like to go another way: fs -> red5 -> flash audio-player > (where the stream to the flash player ist rtmp) > > it would be great, if someone could explain how it works! > > thanks > dennis > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/f21fdc03/attachment.html From 12ukwn at gmail.com Fri Aug 20 07:34:57 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 16:34:57 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <20100820134746.5f95e9dc@anubis.defcon1> Message-ID: <20100820163457.6b6f1b88@anubis.defcon1> Le Fri, 20 Aug 2010 15:37:22 +0200, David Ponzone a ?crit : > I think it should be doable as a table. > I created a template table here: > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#Some_random_results Hehe not so bad after all for such a venerable CPU :) May be the sipp command line used would help to standardize tests. One thing about the DB part though: you should also mention using another file system for the DB's files on HD or even SSD, AND warn people about total data loss in case of power failure w/ the tmpfs possibility. JY -- Iowa State -- the high school after high school! -- Crow T. Robot From david.ponzone at ipeva.fr Fri Aug 20 07:48:50 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Fri, 20 Aug 2010 16:48:50 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <20100820163457.6b6f1b88@anubis.defcon1> References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <20100820134746.5f95e9dc@anubis.defcon1> <20100820163457.6b6f1b88@anubis.defcon1> Message-ID: <0DDC0B63-AD58-44A7-9282-899B4810B35B@ipeva.fr> JY, There is a misunderstanding. I only added the table. The DB part was written before by someone else. Check the history of the page to see who wrote that, or correct it yourself. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 20/08/2010 ? 16:34, Jean-Yves F. Barbier a ?crit : > Le Fri, 20 Aug 2010 15:37:22 +0200, > David Ponzone a ?crit : > >> I think it should be doable as a table. >> I created a template table here: >> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#Some_random_results > > Hehe not so bad after all for such a venerable CPU :) > May be the sipp command line used would help to standardize tests. > > One thing about the DB part though: you should also mention using > another > file system for the DB's files on HD or even SSD, AND warn people > about total data loss in case of power failure w/ the tmpfs > possibility. > > JY > -- > Iowa State -- the high school after high school! > -- Crow T. Robot > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/a9025cdc/attachment.html From anthony.minessale at gmail.com Fri Aug 20 07:52:21 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 09:52:21 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: Yet another load test thread......................... This is why we don't like them. Too bad you missed my presentation at cluecon..... Do you hear yourself btw? Quibbling about completely free software *only* doing 200cps in your fake test? 200cps is 4 times the industry standard FYI and if you were really doing 200cps in real life you would be so rich at our expense that you could afford more boxes. you clearly ignored all of our repeated recommendations. Use the following or you are on your own....... This OS Centos 5.x x86_64 All of these ulimits in your shell ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 240 ulimit -l unlimited ulimit -a run with freeswitch -heavy-timer if you insist on sipp use our scenario file that fixes several errors in the default one. http://www.freeswitch.org/eg/load_test/dft_cap.xml On Fri, Aug 20, 2010 at 4:29 AM, Woody Dickson wrote: > Hi, > > I am doing some experiments with Freeswitch by torturing it to see how > the machine's CPU response to heavy loaded situation. > The test is done on a 16 core 5550 dual quad core server running > fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. > > What I found so strange was that while CPU usage remains pretty low > and distributed among all cores at 190 - 200 calls per second. ?Then, > after added a few more calls per second, all CPU becomes fully > utilized. > > Is this due to some wrong setting? ?Any idea how I can tweak the > configuration and continue my test? > > Thanks, > Woody > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From dswardstrom at remotelink.com Fri Aug 20 07:53:51 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Fri, 20 Aug 2010 07:53:51 -0700 (PDT) Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> Message-ID: <1282316031103-5444786.post@n2.nabble.com> I agree that it would be a good idea to share performance data on the Wiki. If this is done, I have some suggestions (after trying to walk the Wiki many times): - Create a "main" page for performance data. - Allow people to build their own pages relating to performance. - Link these pages to this main page. - Link the main performance page on the Main Page of the Wiki (http://wiki.freeswitch.org/wiki/Main_Page). Perhaps as a new bullet item in: Feature Documentation Then find any already provided performance data in the Wiki pages and get them linked in. We (RemoteLink) have some data regarding number of Conference Users that we could share. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-CPU-usage-tp5443765p5444786.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Fri Aug 20 08:10:32 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 10:10:32 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <1282316031103-5444786.post@n2.nabble.com> References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <1282316031103-5444786.post@n2.nabble.com> Message-ID: The problem is that performance testing is very fluid and relevant to each machine and OS and test performed and you can't get concrete expectations without some advanced skills in deploying FS. On Fri, Aug 20, 2010 at 9:53 AM, David Swardstrom wrote: > > I agree that it would be a good idea to share performance data on the Wiki. > If this is done, I have some suggestions (after trying to walk the Wiki many > times): > - Create a "main" page for performance data. > - Allow people to build their own pages relating to performance. > - Link these pages to this main page. > - Link the main performance page on the Main Page of the Wiki > (http://wiki.freeswitch.org/wiki/Main_Page). > ?Perhaps as a new bullet item in: Feature Documentation > > Then find any already provided performance data in the Wiki pages and get > them linked in. > > We (RemoteLink) have some data regarding number of Conference Users that > we could share. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-CPU-usage-tp5443765p5444786.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From 12ukwn at gmail.com Fri Aug 20 08:13:22 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 17:13:22 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: <20100820171322.6a30cccc@anubis.defcon1> Le Fri, 20 Aug 2010 09:52:21 -0500, Anthony Minessale a ?crit : > Yet another load test thread......................... > > This is why we don't like them. Yep, but people are like that: we looove sooo much tests! (any volunteer to write mod_test for FS and mod_test_test to test mod_test?:) ... > if you insist on sipp use our scenario file that fixes several errors > in the default one. > > http://www.freeswitch.org/eg/load_test/dft_cap.xml Hum, reminds me about quite the same values I saw in an OpenSER serie of tests, thanks for a clear standardization Anthony. JY -- Trust no one -- The X-Files From vipkilla at gmail.com Fri Aug 20 08:17:31 2010 From: vipkilla at gmail.com (vip killa) Date: Fri, 20 Aug 2010 11:17:31 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: can we have the code to doddlephone? On Fri, Aug 20, 2010 at 9:44 AM, Doddle WebPhone wrote: > You can also use SIP and RTP: http://widget.doddlephone.com/ (web driven > phone) > > Sergio > > > On Thu, Aug 12, 2010 at 5:01 AM, Dennis wrote: > >> it sounds as if there is a way to stream audio from fs over red5 to a >> flash-player. >> >> could someone explain how it is done? we do not want any telephony or >> any bi-directional things. we just need a simple one-way solution to >> listen to calls. >> >> in the moment we stream like this: fs -> icecast2 -> flash >> audio-player (which makes it a http stream) >> >> we would like to go another way: fs -> red5 -> flash audio-player >> (where the stream to the flash player ist rtmp) >> >> it would be great, if someone could explain how it works! >> >> thanks >> dennis >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/3dda6526/attachment.html From anthony.minessale at gmail.com Fri Aug 20 08:19:45 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 10:19:45 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: also don't forget... Turn off manage-presence param in sofia. Put your test extension first in your context so it does not hit the demo dialplan which is not designed for high cps. Send me a box like yours cos I only have an 8 core box. On Fri, Aug 20, 2010 at 9:52 AM, Anthony Minessale wrote: > Yet another load test thread......................... > > This is why we don't like them. > > > Too bad you missed my presentation at cluecon..... > > Do you hear yourself btw? Quibbling about completely free software > *only* doing 200cps in your fake test? > 200cps is 4 times the industry standard FYI and if you were really > doing 200cps in real life you would be so rich at our expense that you > could afford more boxes. > > you clearly ignored all of our repeated recommendations. > Use the following or you are on your own....... > > This OS > Centos 5.x x86_64 > > All of these ulimits in your shell > ulimit -c unlimited > ulimit -d unlimited > ulimit -f unlimited > ulimit -i unlimited > ulimit -n 999999 > ulimit -q unlimited > ulimit -u unlimited > ulimit -v unlimited > ulimit -x unlimited > ulimit -s 240 > ulimit -l unlimited > ulimit -a > > > run with freeswitch -heavy-timer > > if you insist on sipp use our scenario file that fixes several errors > in the default one. > > http://www.freeswitch.org/eg/load_test/dft_cap.xml > > > > On Fri, Aug 20, 2010 at 4:29 AM, Woody Dickson wrote: >> Hi, >> >> I am doing some experiments with Freeswitch by torturing it to see how >> the machine's CPU response to heavy loaded situation. >> The test is done on a 16 core 5550 dual quad core server running >> fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. >> >> What I found so strange was that while CPU usage remains pretty low >> and distributed among all cores at 190 - 200 calls per second. ?Then, >> after added a few more calls per second, all CPU becomes fully >> utilized. >> >> Is this due to some wrong setting? ?Any idea how I can tweak the >> configuration and continue my test? >> >> Thanks, >> Woody >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From ktngl at yahoo.co.uk Fri Aug 20 08:32:58 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Fri, 20 Aug 2010 15:32:58 +0000 (GMT) Subject: [Freeswitch-users] dtmf events outbound in 'freeswitcher' ruby library Message-ID: <653943.85681.qm@web29212.mail.ird.yahoo.com> Does anyone know how to get dtmf events in outbound with the freeswitcher ruby library. I see that in inbound 'FSL::Inbound.add_event_hook(:DTMF) {|event| dtmf_handler(event) ' can be used but how can it be done in outbound -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/28e8d111/attachment.html From david.ponzone at ipeva.fr Fri Aug 20 08:38:54 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Fri, 20 Aug 2010 17:38:54 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <1282316031103-5444786.post@n2.nabble.com> Message-ID: <59015441-4FD5-4B76-9C09-EF9218DF7D92@ipeva.fr> Anthony, You're right, the perf data collection I proposed should focus on real life data, because any stress test results would be of no use for other people in the community. We're doing business with real life data, not with theory. I propose to remove the stress test/real life column from the table. I think the point is really for one to know if the perf issues he is experiencing are abnormal, and possibly to contact someone who is having better performances wih a similar config, so he can understand the issue. Also, that would be helpful to scale a new system for someone without any historical data. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 20/08/2010 ? 17:10, Anthony Minessale a ?crit : > The problem is that performance testing is very fluid and relevant to > each machine and OS and test performed and you can't get concrete > expectations without some advanced skills in deploying FS. > > > On Fri, Aug 20, 2010 at 9:53 AM, David Swardstrom > wrote: >> >> I agree that it would be a good idea to share performance data on >> the Wiki. >> If this is done, I have some suggestions (after trying to walk the >> Wiki many >> times): >> - Create a "main" page for performance data. >> - Allow people to build their own pages relating to performance. >> - Link these pages to this main page. >> - Link the main performance page on the Main Page of the Wiki >> (http://wiki.freeswitch.org/wiki/Main_Page). >> Perhaps as a new bullet item in: Feature Documentation >> >> Then find any already provided performance data in the Wiki pages >> and get >> them linked in. >> >> We (RemoteLink) have some data regarding number of Conference Users >> that >> we could share. >> >> -- >> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-CPU-usage-tp5443765p5444786.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/c2e2ab7d/attachment-0001.html From 12ukwn at gmail.com Fri Aug 20 08:52:24 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 17:52:24 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <1282316031103-5444786.post@n2.nabble.com> Message-ID: <20100820175224.5d472a21@anubis.defcon1> Le Fri, 20 Aug 2010 10:10:32 -0500, Anthony Minessale a ?crit : > The problem is that performance testing is very fluid and relevant to > each machine and OS and test performed and you can't get concrete > expectations without some advanced skills in deploying FS. Well, it could have a real interest by adding the tweaked parameters, may be there are some, yet unexplored, that would raise the response. -- Why don't elephants eat penguins ? Because they can't get the wrappers off ... From anthony.minessale at gmail.com Fri Aug 20 09:01:18 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 11:01:18 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <20100820175224.5d472a21@anubis.defcon1> References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <1282316031103-5444786.post@n2.nabble.com> <20100820175224.5d472a21@anubis.defcon1> Message-ID: Its an ok thing as long as it comes with a disclaimer that "results may vary" and "you have to know what you are doing" And those who work on it need to volunteer to help with the emails it may generate asking why they can't match the results =D On Fri, Aug 20, 2010 at 10:52 AM, Jean-Yves F. Barbier <12ukwn at gmail.com> wrote: > Le Fri, 20 Aug 2010 10:10:32 -0500, > Anthony Minessale a ?crit : > >> The problem is that performance testing is very fluid and relevant to >> each machine and OS and test performed and you can't get concrete >> expectations without some advanced skills in deploying FS. > > Well, it could have a real interest by adding the tweaked parameters, > may be there are some, yet unexplored, that would raise the response. > > -- > Why don't elephants eat penguins ? > Because they can't get the wrappers off ... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From vhatz at kinetix.gr Fri Aug 20 09:19:43 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou) Date: Fri, 20 Aug 2010 19:19:43 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: <4C6EAB1F.9060904@kinetix.gr> On 20/8/2010 5:52 ??, Anthony Minessale wrote: > Yet another load test thread......................... > > This is why we don't like them. > > > Too bad you missed my presentation at cluecon..... > > Do you hear yourself btw? Quibbling about completely free software > *only* doing 200cps in your fake test? I don't know if I missed any message in this thread, but why do you say that the tester did a fake test? > 200cps is 4 times the industry standard FYI and if you were really > doing 200cps in real life you would be so rich at our expense that you > could afford more boxes. > > you clearly ignored all of our repeated recommendations. > Use the following or you are on your own....... > > This OS > Centos 5.x x86_64 If someone is on his own for any other OS than Centos 5.x x86_64 then why is FS offered for so many different platforms? It's only natural for people to want to test it on their favorite platform... Perhaps, as you recommend, Centos 5.x x86_64 gives the best results over all other platforms, but last time I checked, a user is still allowed to ask performance related questions for his favorite platform in this mailing list, right? Woody's post was not expressing any negative opinion about FS's performance. He didn't comment on whether 200cps was too little or too much. He just wrote that the CPU usage increased in an unexpected manner and wonders how/if he can solve this. He wrote that he was doing a load test scenario, he didn't write that he was making money at 200cps "at your expense". And he wrote all this in a manner which was neither insulting nor abusive. The poster just wanted to learn something, see if others have gotten better results with other setups. He was trying to make a valid discussion to solve a problem, and other users were actually replying to him. This is what mailing lists are for, right? I really don't understand why red flags have to be raised every time the word "performance" is mentioned in this mailing list. -- Best regards, Vlasis Hatzistavrou. From 12ukwn at gmail.com Fri Aug 20 09:21:05 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 18:21:05 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <1282316031103-5444786.post@n2.nabble.com> <20100820175224.5d472a21@anubis.defcon1> Message-ID: <20100820182105.40f5f58b@anubis.defcon1> Le Fri, 20 Aug 2010 11:01:18 -0500, Anthony Minessale a ?crit : > Its an ok thing as long as it comes with a disclaimer that "results > may vary" and "you have to know what you are doing" > And those who work on it need to volunteer to help with the emails it > may generate asking why they can't match the results =D It can be resumed: YMMV, RTFM & FYI... -- From gmaruzz at celliax.org Fri Aug 20 09:27:14 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 20 Aug 2010 18:27:14 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6EAB1F.9060904@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> Message-ID: On Fri, Aug 20, 2010 at 6:19 PM, Vlasis Hatzistavrou wrote: > > I really don't understand why red flags have to be raised every time the > word "performance" is mentioned in this mailing list. > Don't tell THAT word again! :)))! -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From anthony.minessale at gmail.com Fri Aug 20 09:34:05 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 11:34:05 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6EAB1F.9060904@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> Message-ID: That's easy for you to say because you are using the mailing list not running it or taking a personal vested interest in maintaining the quality of it's content. We are trying to make sure we help everyone all day long every day constantly. If you personally had to deal with every time we get this recurring "load testing" threads, you might understand. The alternative would be for us to lower our standards and let the list run away with itself and not care if we actually keep up on it. What mailing lists are for is anything the owner of the list wants it to be for. He is welcome to continue to say anything he wants and we are obliged to reply however we want. It's a 2 way street. You seem to suggest that we don't have to right to have policies on our own community that many people are very satisfied with only because we do exactly what we want to run our own community. You as well are welcome to your opinion which is duly noted. We still however, do not like load testing threads. The individuals here who have volunteered to set up some wiki info with the proper disclaimers may produce a resource we can send people to and be done with it. That is what WIKIs and FAQs are for..... On Fri, Aug 20, 2010 at 11:19 AM, Vlasis Hatzistavrou wrote: > ?On 20/8/2010 5:52 ??, Anthony Minessale wrote: >> Yet another load test thread......................... >> >> This is why we don't like them. >> >> >> Too bad you missed my presentation at cluecon..... >> >> Do you hear yourself btw? Quibbling about completely free software >> *only* doing 200cps in your fake test? > I don't know if I missed any message in this thread, but why do you say > that the tester did a fake test? > >> 200cps is 4 times the industry standard FYI and if you were really >> doing 200cps in real life you would be so rich at our expense that you >> could afford more boxes. >> >> you clearly ignored all of our repeated recommendations. >> Use the following or you are on your own....... >> >> This OS >> Centos 5.x x86_64 > > If someone is on his own for any other OS than Centos 5.x x86_64 then > why is FS offered for so many different platforms? > > It's only natural for people to want to test it on their favorite > platform... Perhaps, as you recommend, Centos 5.x x86_64 gives the best > results over all other platforms, but last time I checked, a user is > still allowed to ask performance related questions for his favorite > platform in this mailing list, right? > > Woody's post was not expressing any negative opinion about FS's > performance. He didn't comment on whether 200cps was too little or too > much. He just wrote that the CPU usage increased in an unexpected manner > and wonders how/if he can solve this. He wrote that he was doing a load > test scenario, he didn't write that he was making money at 200cps "at > your expense". And he wrote all this in a manner which was neither > insulting nor abusive. > > The poster just wanted to learn something, see if others have gotten > better results with other setups. He was trying to make a valid > discussion to solve a problem, and other users were actually replying to > him. > > This is what mailing lists are for, right? > > I really don't understand why red flags have to be raised every time the > word "performance" is mentioned in this mailing list. > > -- > Best regards, > Vlasis Hatzistavrou. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From tculjaga at gmail.com Fri Aug 20 10:30:52 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 20 Aug 2010 19:30:52 +0200 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> Message-ID: can we get the wireshark sniff taken on FS (both sides) without being filtered ? T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/449e4a8d/attachment.html From vetali100 at gmail.com Fri Aug 20 10:56:28 2010 From: vetali100 at gmail.com (Vitalii Colosov) Date: Fri, 20 Aug 2010 20:56:28 +0300 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> Message-ID: Hi Dennis, Regarding change in RTP data.. 1. Which codecs are being used? Do you perform any transcoding in FS? 2. If no transcoding is done, maybe you could enable "proxy-media" for a moment http://wiki.freeswitch.org/wiki/Proxy_Media and check again whether the RTP is changed or not by FS, at this time. Regards, Vitalie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/dfa119b4/attachment.html From msc at freeswitch.org Fri Aug 20 12:42:37 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Aug 2010 12:42:37 -0700 Subject: [Freeswitch-users] IVR menu failure/timeout action In-Reply-To: References: <576CA05DD2404C1891913B70753E94E2@COMMVERGEHK.LOCAL> Message-ID: Alan, I it looks like the transfer is going to the public context which I'm assuming is not what you want. Try changing this: ** To this: ** Let us know if that helps. -MC On Thu, Aug 19, 2010 at 8:05 PM, alan lo wrote: > MC > > I have put the log on the following URL. > > http://pastebin.freeswitch.org/13682 > > > > Please help check it! Thanks. > > > > Alan > > > ------------------------------ > > *From:* Michael Collins [mailto:msc at freeswitch.org] > *Sent:* Friday, August 20, 2010 1:46 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] IVR menu failure/timeout action > > > > Alan, > > Please collect a debug trace of the call failing. Use "console loglevel > debug" at the CLI. Also, use pastebin for the log so that this thread > doesn't get too long. http://pastebin:freeswitch at pastebin.freeswitch.org/ > > After you have pastebin'd the logs please give us the URL of the pb entry > and we'll have a look. > Thanks, > MC > > On Wed, Aug 18, 2010 at 8:10 PM, alan lo wrote: > > Thx for the answer, but I tried but it doesn???t work. > > Below is some of my code in my file ?? v_IVR_demo.xml?? under folder ??? > ivr_menus?? > > ** > > *? ** * > > *? ? ? ** * > > *? ** * > > ** > > ? > > ** > > ? > > The call dropped immediately after I press 1 in the phone > > Below is the log. Phone number and IP are masked with (MyPhone) and (MyIP) > respectively. > > 2010-08-19 10:47:34.220750 [NOTICE] mod_dptools.c:720 Channel > [sofia/external/(MyPhone)@ (MyIP)] has been answered > > 2010-08-19 10:47:44.892625 [NOTICE] switch_ivr.c:1450 Transfer > sofia/external/(MyPhone)@ (MyIP) to XML[submenu at public] > > 2010-08-19 10:47:44.892625 [NOTICE] switch_cpp.cpp:597 Hangup > sofia/external/(MyPhone)@ (MyIP) [CS_ROUTING] [NORMAL_CLEARING] > > 2010-08-19 10:47:44.892625 [NOTICE] switch_core_session.c:1188 Session 6 > (sofia/external/(MyPhone)@ (MyIP)) Ended > > 2010-08-19 10:47:44.892625 [NOTICE] switch_core_session.c:1190 Close > Channel sofia/external/(MyPhone)@ (MyIP) [CS_DESTROY] > > ? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/2ba4bf54/attachment-0001.html From ssa1357 at yahoo.com Fri Aug 20 13:14:32 2010 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Fri, 20 Aug 2010 13:14:32 -0700 (PDT) Subject: [Freeswitch-users] Mod_h323 fail to load Message-ID: <40241.74408.qm@web53405.mail.re2.yahoo.com> Dear All, I try to install mod_h323, I compile it as it comes in wiki (http://wiki.freeswitch.org/wiki/Mod_h323) but when I try to load mod_h323 in fs_cli I faces this error: > >2010-08-20 20:06:07.781508 [CRIT] switch_loadable_module.c:882 Error Loading >module /usr/local/freeswitch/mod/mod_h323.so >**libh323_linux_x86_.so.1.22.0: cannot open shared object file: No such file or >directory** > >I cannot find what the problem is. and also I cannot find libh323_linux file. >Would somebody please help me with it? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/57df1db5/attachment.html From 12ukwn at gmail.com Fri Aug 20 13:19:43 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 22:19:43 +0200 Subject: [Freeswitch-users] radius Message-ID: <20100820221943.478f70f9@anubis.defcon1> Hi list, what can bring a RADIUS server in the equation? (I never used one, so my knowledge is basic: centralized authentication using PAP or CHAP protocols) JY -- The human race never solves any of its problems. It merely outlives them. -- David Gerrold From 12ukwn at gmail.com Fri Aug 20 13:27:22 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 22:27:22 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <40241.74408.qm@web53405.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> Message-ID: <20100820222722.46e7ccb9@anubis.defcon1> Le Fri, 20 Aug 2010 13:14:32 -0700 (PDT), Sadjad Seyed-Ahmadian a ?crit : > I try to install mod_h323, I compile it as it comes in wiki > (http://wiki.freeswitch.org/wiki/Mod_h323) > but when I try to load mod_h323 in fs_cli I faces this error: Can you find a 'mod_h323.so/.la' in /usr/local/freeswitch/lib? -- The best way to make a fire with two sticks is to make sure one of them is a match. -- Will Rogers From david.ponzone at ipeva.fr Fri Aug 20 13:29:06 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Fri, 20 Aug 2010 22:29:06 +0200 Subject: [Freeswitch-users] radius In-Reply-To: <20100820221943.478f70f9@anubis.defcon1> References: <20100820221943.478f70f9@anubis.defcon1> Message-ID: <72993EF1-0B74-443E-AADF-B7E6CAA58DCF@ipeva.fr> Hmm well yes, but PAP/CHAP is what you would used to auth a PPP user. In the case of FS, Radius would just authenticate. With the right client on the FS side, you could send all the parameters you want to FS, like where to route the call, etc... Also, Radius does accounting, so billing. I am not sure why Radius would be a better choice than a native SQL custom client, but there are probably pros and cons. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 20/08/2010 ? 22:19, Jean-Yves F. Barbier a ?crit : > Hi list, > > what can bring a RADIUS server in the equation? > (I never used one, so my knowledge is basic: centralized > authentication > using PAP or CHAP protocols) > > JY > -- > The human race never solves any of its problems. It merely outlives > them. > -- David Gerrold > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/25416bff/attachment.html From ssa1357 at yahoo.com Fri Aug 20 13:43:19 2010 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Fri, 20 Aug 2010 13:43:19 -0700 (PDT) Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <20100820222722.46e7ccb9@anubis.defcon1> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> Message-ID: <31192.48276.qm@web53406.mail.re2.yahoo.com> No I cannot find it! ________________________________ From: Jean-Yves F. Barbier <12ukwn at gmail.com> To: freeswitch-users at lists.freeswitch.org Sent: Sat, August 21, 2010 12:57:22 AM Subject: Re: [Freeswitch-users] Mod_h323 fail to load Le Fri, 20 Aug 2010 13:14:32 -0700 (PDT), Sadjad Seyed-Ahmadian a ?crit : > I try to install mod_h323, I compile it as it comes in wiki > (http://wiki.freeswitch.org/wiki/Mod_h323) > but when I try to load mod_h323 in fs_cli I faces this error: Can you find a 'mod_h323.so/.la' in /usr/local/freeswitch/lib? -- The best way to make a fire with two sticks is to make sure one of them is a match. -- Will Rogers _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/20784fe9/attachment.html From phone.bytes at gmail.com Fri Aug 20 13:46:31 2010 From: phone.bytes at gmail.com (Phone) Date: Fri, 20 Aug 2010 14:46:31 -0600 Subject: [Freeswitch-users] Webapi Examples location Message-ID: <4C6EE9A7.4050108@gmail.com> I am exploring the Webapi. The sample applications "Voicemail" and "Telecast" are working as they should, but where do I find the source for them?? Thanks From 12ukwn at gmail.com Fri Aug 20 13:52:05 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 22:52:05 +0200 Subject: [Freeswitch-users] radius In-Reply-To: <72993EF1-0B74-443E-AADF-B7E6CAA58DCF@ipeva.fr> References: <20100820221943.478f70f9@anubis.defcon1> <72993EF1-0B74-443E-AADF-B7E6CAA58DCF@ipeva.fr> Message-ID: <20100820225205.5592733a@anubis.defcon1> Le Fri, 20 Aug 2010 22:29:06 +0200, David Ponzone a ?crit : > Hmm well yes, but PAP/CHAP is what you would used to auth a PPP user. > In the case of FS, Radius would just authenticate. > With the right client on the FS side, you could send all the > parameters you want to FS, like where to route the call, etc... Ok, until this point quite similar to LDAP (without the headaches though:) > Also, Radius does accounting, so billing. Ha ok, so the data exchange is bidir. > I am not sure why Radius would be a better choice than a native SQL > custom client, but there are probably pros and cons. From what I read (wikipedia), RADIUS precisely avoid multiple DB replications in favor of only one DB for multiple RADIUS svrs (as for ISPs modems, emailing logins, etc) by securing all communications conveying the authentication data. Thanks David. -- Hear about... the new rule at the girls' school? Lights out by ten, candles by eleven. From 12ukwn at gmail.com Fri Aug 20 14:06:15 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 20 Aug 2010 23:06:15 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <31192.48276.qm@web53406.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> Message-ID: <20100820230615.126250f8@anubis.defcon1> Le Fri, 20 Aug 2010 13:43:19 -0700 (PDT), Sadjad Seyed-Ahmadian a ?crit : > No I cannot find it! So your compilation likely failed and you don't have either mod_h323 nor the library you mentioned available. Try to completely redirect the compilation output into a file to be able to review carefully what's happened: make mod_h323 >MYLOG.LOG 2>&1 -- From steveayre at gmail.com Fri Aug 20 14:07:07 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 20 Aug 2010 22:07:07 +0100 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <31192.48276.qm@web53406.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> Message-ID: mod_h323.so will be in /usr/local/freeswitch/mod (or /opt/freeswitch/mod). That's fine, it's loading that file fine, the issue is the h323plus library dependancy isn't being loaded. It's trying to load the libh323_linux_x86_.so.1.22.0.so file and can't find it. Can you find that file anywhere on your system? Can you confirm that h323plus is installed, or have you copied the freeswitch files over from another system? If you can, can you see which files are in that package? On my system the equivalent file is at /usr/lib/libh323.so.1.22.0 The file's name is different, but that could just be something about the OS (I'm on debian lenny). Your filename seems to be missing something though... not sure if that's causing an issue. It has a _ followed by a dot, which makes it seem like something was meant to be in there. -Steve On 20 August 2010 21:43, Sadjad Seyed-Ahmadian wrote: > No I cannot find it! > > ------------------------------ > *From:* Jean-Yves F. Barbier <12ukwn at gmail.com> > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Sat, August 21, 2010 12:57:22 AM > *Subject:* Re: [Freeswitch-users] Mod_h323 fail to load > > Le Fri, 20 Aug 2010 13:14:32 -0700 (PDT), > Sadjad Seyed-Ahmadian a ?crit : > > > I try to install mod_h323, I compile it as it comes in wiki > > (http://wiki.freeswitch.org/wiki/Mod_h323) > > but when I try to load mod_h323 in fs_cli I faces this error: > > Can you find a 'mod_h323.so/.la' in /usr/local/freeswitch/lib? > > -- > The best way to make a fire with two sticks is to make sure one of them > is a match. > -- Will Rogers > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/ef9324d3/attachment.html From ssa1357 at yahoo.com Fri Aug 20 14:10:01 2010 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Fri, 20 Aug 2010 14:10:01 -0700 (PDT) Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <31192.48276.qm@web53406.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> Message-ID: <739604.57080.qm@web53408.mail.re2.yahoo.com> I got the pint files are in /usr/local/lib and I make a link in /isr/local/freeswitch/lib to them, but when I try to load mod_h323 freeswitch crashed!! ________________________________ From: Sadjad Seyed-Ahmadian To: FreeSWITCH Users Help Sent: Sat, August 21, 2010 1:13:19 AM Subject: Re: [Freeswitch-users] Mod_h323 fail to load No I cannot find it! ________________________________ From: Jean-Yves F. Barbier <12ukwn at gmail.com> To: freeswitch-users at lists.freeswitch.org Sent: Sat, August 21, 2010 12:57:22 AM Subject: Re: [Freeswitch-users] Mod_h323 fail to load Le Fri, 20 Aug 2010 13:14:32 -0700 (PDT), Sadjad Seyed-Ahmadian a ?crit : > I try to install mod_h323, I compile it as it comes in wiki > (http://wiki.freeswitch.org/wiki/Mod_h323) > but when I try to load mod_h323 in fs_cli I faces this error: Can you find a 'mod_h323.so/.la' in /usr/local/freeswitch/lib? -- The best way to make a fire with two sticks is to make sure one of them is a match. -- Will Rogers _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/8c137bed/attachment.html From anthony.minessale at gmail.com Fri Aug 20 14:24:27 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 16:24:27 -0500 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <739604.57080.qm@web53408.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> Message-ID: probably missing ldconfig or LD_LIBRARY_PATH On Fri, Aug 20, 2010 at 4:10 PM, Sadjad Seyed-Ahmadian wrote: > I got the pint files are in /usr/local/lib and I make a link in > /isr/local/freeswitch/lib to them, but when I try to load mod_h323 > freeswitch crashed!! > > ________________________________ > From: Sadjad Seyed-Ahmadian > To: FreeSWITCH Users Help > Sent: Sat, August 21, 2010 1:13:19 AM > Subject: Re: [Freeswitch-users] Mod_h323 fail to load > > No I cannot find it! > ________________________________ > From: Jean-Yves F. Barbier <12ukwn at gmail.com> > To: freeswitch-users at lists.freeswitch.org > Sent: Sat, August 21, 2010 12:57:22 AM > Subject: Re: [Freeswitch-users] Mod_h323 fail to load > > Le Fri, 20 Aug 2010 13:14:32 -0700 (PDT), > Sadjad Seyed-Ahmadian a ?crit : > >> I try to install mod_h323, I compile it as it comes in wiki >> (http://wiki.freeswitch.org/wiki/Mod_h323) >> but when I try to load mod_h323 in fs_cli I faces this error: > > Can you find a 'mod_h323.so/.la' in /usr/local/freeswitch/lib? > > -- > The best way to make a fire with two sticks is to make sure one of them > is a match. > ??? ??? -- Will Rogers > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From kris at kriskinc.com Fri Aug 20 14:38:49 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 20 Aug 2010 14:38:49 -0700 Subject: [Freeswitch-users] Selecting codec by sample rate Message-ID: Hello everyone, Does FreeSWITCH support the ability to accept/reject/filter/modify codec by sample rate? I've defined SILK at 8000h as the only supported codec in my profile yet clients calling in that prefer SILK at 12000h over SILK at 8000h are allowed to bring up the call at SILK at 12000h. It seems that SILK at any sample rate is allowed as long as SILK at any sample rate is defined in the profile. My profile is set to scrooge and I've tried various combinations of late-negotiation, absolute_codec_string, etc to no avail. Should I be able to accept/reject specific sample rates? If not I'm willing to create a bounty for it... Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Fri Aug 20 14:45:45 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Aug 2010 16:45:45 -0500 Subject: [Freeswitch-users] Selecting codec by sample rate In-Reply-To: References: Message-ID: <477D0E04-D327-40BF-92B9-07C54B6FFF20@freeswitch.org> Sounds like you need to define the inbound and outbound codec preferences to make sure that happens. Are you doing that? /b On Aug 20, 2010, at 4:38 PM, Kristian Kielhofner wrote: > Hello everyone, > > Does FreeSWITCH support the ability to accept/reject/filter/modify > codec by sample rate? > > I've defined SILK at 8000h as the only supported codec in my profile > yet clients calling in that prefer SILK at 12000h over SILK at 8000h are > allowed to bring up the call at SILK at 12000h. It seems that SILK at > any sample rate is allowed as long as SILK at any sample rate is > defined in the profile. > > My profile is set to scrooge and I've tried various combinations of > late-negotiation, absolute_codec_string, etc to no avail. > > Should I be able to accept/reject specific sample rates? If not I'm > willing to create a bounty for it... > > Thanks! From anthony.minessale at gmail.com Fri Aug 20 14:50:30 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 16:50:30 -0500 Subject: [Freeswitch-users] Selecting codec by sample rate In-Reply-To: References: Message-ID: There is no current way to do this because it will attempt to do the best with what it has to work with. Currently its considered a preference not a requirement which one to use. It probably is possible to invent a way but it's not there now. On Fri, Aug 20, 2010 at 4:38 PM, Kristian Kielhofner wrote: > Hello everyone, > > ?Does FreeSWITCH support the ability to accept/reject/filter/modify > codec by sample rate? > > ?I've defined SILK at 8000h as the only supported codec in my profile > yet clients calling in that prefer SILK at 12000h over SILK at 8000h are > allowed to bring up the call at SILK at 12000h. ?It seems that SILK at > any sample rate is allowed as long as SILK at any sample rate is > defined in the profile. > > ?My profile is set to scrooge and I've tried various combinations of > late-negotiation, absolute_codec_string, etc to no avail. > > ?Should I be able to accept/reject specific sample rates? ?If not I'm > willing to create a bounty for it... > > Thanks! > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From kris at kriskinc.com Fri Aug 20 15:08:51 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 20 Aug 2010 15:08:51 -0700 Subject: [Freeswitch-users] Selecting codec by sample rate In-Reply-To: References: Message-ID: Hi Tony, Thanks for getting back to me. What would it take (time/money) to enforce this down to the ptime? I've created a bounty: http://wiki.freeswitch.org/wiki/Bounty#Enforce_codec_selection_by_sample_rate_and_ptime http://jira.freeswitch.org/browse/BOUNTY-21 On Fri, Aug 20, 2010 at 2:50 PM, Anthony Minessale wrote: > There is no current way to do this because it will attempt to do the > best with what it has to work with. > Currently its considered a preference not a requirement which one to use. > It probably is possible to invent a way but it's not there now. > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From vhatz at kinetix.gr Fri Aug 20 15:17:26 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Sat, 21 Aug 2010 01:17:26 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <4C6EAB1F.9060904@kinetix.gr> Message-ID: <4C6EFEF6.2000106@kinetix.gr> Anthony Minessale wrote: > That's easy for you to say because you are using the mailing list not > running it or taking a personal vested interest in maintaining the > quality of it's content. > > We are trying to make sure we help everyone all day long every day constantly. > If you personally had to deal with every time we get this recurring > "load testing" threads, you might understand. > > I understand that you may feel frustrated on a topic that you have already covered many times in the past in the list. The fact, however, that the list still receives performance-related questions shows that the users are still interested in the subject, that they still have questions and that they need to compare practices, information, set ups, anything one can think of, despite the information already available in the wiki and in the mailing list. The fact that some users still ask and wonder about performance does not mean that they dispute FS's performance or that the users are too lazy to test thoroughly or that they don't know what they are doing (OK, granted some of us don't, but may be willing to learn). Performance related questions pop up all the time in the mailing lists of probably every software that I have watched. IMHO, this is a good thing. Higher performance is of interest in all kinds of software, this being especially true for VoIP software. The question "ok, great, we get 200 cps. Now, how can we get the software to do more" does not show stupidity, greed or ingratitude. It is a valid question that can move forward the development of any software. > The alternative would be for us to lower our standards and let the > list run away with itself and not care if we actually keep up on it. > > What mailing lists are for is anything the owner of the list wants it to be for. > He is welcome to continue to say anything he wants and we are obliged > to reply however we want. > It's a 2 way street. You seem to suggest that we don't have to right > to have policies on our own community that many people are very > satisfied with only because we do exactly what we want to run our own > community. > I never suggested that you don't have the right to have policies in the list that you have created. I'm saying something different: although I didn't see anywhere a policy that the list users are forbidden to post performance related questions, this is exactly how such questions are treated in practice. Of course, I can't even begin to imagine any software mailing list with a policy of forbidding the discussion of "taboo" subjects. And this is exactly the reason why I wrote my post. Specifically in this case, Woody wrote a valid question about a subject not forbidden by any policy. He deserved an answer by anyone willing to reply as much as the next user on the list. I know that you have a busy schedule, and that you cannot possibly participate in all threads, especially in subjects already covered, even more so for subjects that you don't like. But there are already experienced users in the list who are willing to reply. Some users, who were willing to help, actually did reply and in fact a whole discussion about performance and test data was generated, showing beyond doubt that this is a hot topic for FS users. And IMHO this is a good thing for FS. The original question in this thread did not attack your work or FreeSWITCH in general in any way. So, no need to accuse the user of making money on your expense or that he performs fake tests. It was just a question, after all. Best regards, Vlasis Hatzistavrou. From anthony.minessale at gmail.com Fri Aug 20 15:43:04 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 17:43:04 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6EFEF6.2000106@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> Message-ID: I am not accusing the user of anything I am making factual statements. They are indeed fake tests because it's sipp and 600cps. I said *if* he was doing many hundreds of cps *in real life* he would indeed be very rich at our expense, (meaning using our software not meaning he is chaining us to a wall in his basement) and thus, he would be able to afford more boxes. Do you feel I was not willing to help when I provided a slew of configuration options that will increase performance as well as proper sipp files to use? I also endorsed the idea of the wiki page as long as they maintain it. Why are you arguing on someone else's behalf on a matter that needs no arguing. You are continuing to argue that I am not allowed to say what I want on my own mailing list which is somewhat troubling. If you decide to reply again, please limit the length of your post. On Fri, Aug 20, 2010 at 5:17 PM, Vlasis Hatzistavrou (KTI) wrote: > Anthony Minessale wrote: >> That's easy for you to say because you are using the mailing list not >> running it or taking a personal vested interest in maintaining ?the >> quality of it's content. >> >> We are trying to make sure we help everyone all day long every day constantly. >> If you personally had to deal with every time we get this recurring >> "load testing" threads, you might understand. >> >> > I understand that you may feel frustrated on a topic that you have > already covered many times in the past in the list. > > The fact, however, that the list still receives performance-related > questions shows that the users are still interested in the subject, that > they still have questions and that they need to compare practices, > information, set ups, anything one can think of, despite the information > already available in the wiki and in the mailing list. > > The fact that some users still ask and wonder about performance does not > mean that they dispute FS's performance or that the users are too lazy > to test thoroughly or that they don't know what they are doing (OK, > granted some of us don't, but may be willing to learn). > > Performance related questions pop up all the time in the mailing lists > of probably every software that I have watched. IMHO, this is a good > thing. Higher performance is of interest in all kinds of software, this > being especially true for VoIP software. > > The question > > "ok, great, we get 200 cps. Now, how can we get the software to do more" > > does not show stupidity, greed or ingratitude. It is a valid question > that can move forward the development of any software. > >> The alternative would be for us to lower our standards and let the >> list run away with itself and not care if we actually keep up on it. >> >> What mailing lists are for is anything the owner of the list wants it to be for. >> He is welcome to continue to say anything he wants and we are obliged >> to reply however we want. >> It's a 2 way street. ?You seem to suggest that we don't have to right >> to have policies on our own community that many people are very >> satisfied with only because we do exactly what we want to run our own >> community. >> > I never suggested that you don't have the right to have policies in the > list that you have created. I'm saying something different: although I > didn't see anywhere a policy that the list users are forbidden to post > performance related questions, this is exactly how such questions are > treated in practice. > > Of course, I can't even begin to imagine any software mailing list with > a policy of forbidding the discussion of "taboo" subjects. And this is > exactly the reason why I wrote my post. > > Specifically in this case, Woody wrote a valid question about a subject > not forbidden by any policy. He deserved an answer by anyone willing to > reply as much as the next user on the list. > > I know that you have a busy schedule, and that you cannot possibly > participate in all threads, especially in subjects already covered, even > more so for subjects that you don't like. > > But there are already experienced users in the list who are willing to > reply. Some users, who were willing to help, actually did reply and in > fact a whole discussion about performance and test data was generated, > showing beyond doubt that this is a hot topic for FS users. And IMHO > this is a good thing for FS. > > The original question in this thread did not attack your work or > FreeSWITCH in general in any way. So, no need to accuse the user of > making money on your expense or that he performs fake tests. It was just > a question, after all. > > Best regards, > Vlasis Hatzistavrou. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Fri Aug 20 15:48:51 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Aug 2010 17:48:51 -0500 Subject: [Freeswitch-users] Selecting codec by sample rate In-Reply-To: References: Message-ID: The first part is straightforward but the part after the "likewise" with the same codec in many rates may be problematic. I'll have to think about it but it's not at all trivial of a change and will have to be carefully tested to avoid regressions. On Fri, Aug 20, 2010 at 5:08 PM, Kristian Kielhofner wrote: > Hi Tony, > > ?Thanks for getting back to me. ?What would it take (time/money) to > enforce this down to the ptime? > > ?I've created a bounty: > > http://wiki.freeswitch.org/wiki/Bounty#Enforce_codec_selection_by_sample_rate_and_ptime > http://jira.freeswitch.org/browse/BOUNTY-21 > > On Fri, Aug 20, 2010 at 2:50 PM, Anthony Minessale > wrote: >> There is no current way to do this because it will attempt to do the >> best with what it has to work with. >> Currently its considered a preference not a requirement which one to use. >> It probably is possible to invent a way but it's not there now. >> >> > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From chat2jesse at gmail.com Fri Aug 20 15:49:44 2010 From: chat2jesse at gmail.com (jesse zhao) Date: Fri, 20 Aug 2010 15:49:44 -0700 Subject: [Freeswitch-users] Webapi Examples location In-Reply-To: <4C6EE9A7.4050108@gmail.com> References: <4C6EE9A7.4050108@gmail.com> Message-ID: why my message post to the mail list gets blocked? testing by reply . -jesse On Fri, Aug 20, 2010 at 1:46 PM, Phone wrote: > I am exploring the Webapi. > > The sample applications "Voicemail" and "Telecast" are working as they > should, but where do I find the source for them?? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chat2jesse at gmail.com Fri Aug 20 15:58:40 2010 From: chat2jesse at gmail.com (jesse zhao) Date: Fri, 20 Aug 2010 15:58:40 -0700 Subject: [Freeswitch-users] why dingaling is using G722 codec? Message-ID: I am dialing from SIP to Jingle. in mod_jingle.conf.xml I only configured PCMU and GSM codec. How come FS sends G722 codec and gtalk side immedially rejects the call. 2010-08-20 14:23:56.777993 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2010-08-20 14:23:57.000832 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- From kris at kriskinc.com Fri Aug 20 16:27:22 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 20 Aug 2010 16:27:22 -0700 Subject: [Freeswitch-users] Selecting codec by sample rate In-Reply-To: References: Message-ID: Tony, Thanks, that's what I was trying to asses (trivial vs. non-trivial). I'm spoiled because most of my requests seem to be trivial and get fixed in an hour or less :). Well the bounty is still up and open. This is something I'm very interested in and if you get any more details let me know. Thanks again! On Fri, Aug 20, 2010 at 3:48 PM, Anthony Minessale wrote: > The first part is straightforward but the part after the "likewise" > with the same codec in many rates may be problematic. > I'll have to think about it but it's not at all trivial of a change > and will have to be carefully tested to avoid regressions. > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From vhatz at kinetix.gr Fri Aug 20 16:39:36 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Sat, 21 Aug 2010 02:39:36 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> Message-ID: <4C6F1238.6000109@kinetix.gr> Anthony Minessale wrote: > I am not accusing the user of anything I am making factual statements. > > They are indeed fake tests because it's sipp and 600cps. > Can you elaborate? I read 200 cps in the original post instead of 600 cps and perhaps you can spot a fake test from miles away but please enlighten me. > I said *if* he was doing many hundreds of cps *in real life* he would > indeed be very rich at our expense, (meaning using our software not > meaning he is chaining us to a wall in his basement) First of all, he would be rich using a million other things needed to set up and run a business, FS being only one of them. He would not be getting rich at YOUR expense, there is a huge difference. That was a wrong thing to say. > Do you feel I was not willing to help when I provided a slew of > configuration options that will increase performance as well as proper > sipp files to use? > > You helped, but only after making a point that you really don't like performance related questions, accusing the poster of fake tests (whatever that means) and how he was (or could be) making money at your expense. You've reacted in a similar manner in other performance reacted questions in the past. If you are not going to help by answering, who could ever accuse you? We all know that you are busy. But in the end, if you bother to reply and help, do you really need to bash first? > I also endorsed the idea of the wiki page as long as they maintain it. > > Why are you arguing on someone else's behalf on a matter that needs no arguing. > I argue on his behalf because I have been at his place in the past. I hated it. It's that simple. Getting bashed for asking a simple question is NOT nice, to say the least. > You are continuing to argue that I am not allowed to say what I want > on my own mailing list which is somewhat troubling. > Where did I say that exactly? All I'm saying is: "if you don't like a question on the list, then please spare us the bashing". You don't even have to answer if you don't want to. You wrote in your reply to Woody: "Do you hear yourself btw?" I can ask you the same question: Do you hear yourself? You act as if some subjects are taboo in this list. > If you decide to reply again, please limit the length of your post. > > Now it is the size of my posts that troubles you? I've seen longer posts in the list and you didn't complain. Perhaps you should add a size restriction directive in the mailing list policy, along with a "no performance questions allowed" directive. That should clear things up significantly. Cheers, Vlasis. From msc at freeswitch.org Fri Aug 20 16:57:00 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Aug 2010 16:57:00 -0700 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6F1238.6000109@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> Message-ID: > You wrote in your reply to Woody: "Do you hear yourself btw?" I can ask > you the same question: Do you hear yourself? You act as if some subjects > are taboo in this list. > Taboos aren't really the issue. Spending valuable time and energy in unfruitful debates is an issue. > > > If you decide to reply again, please limit the length of your post. > > > > > Now it is the size of my posts that troubles you? I've seen longer posts > in the list and you didn't complain. > Brevity is the essence of communication - I highly recommend it. > > Perhaps you should add a size restriction directive in the mailing list > policy, along with a "no performance questions allowed" directive. That > should clear things up significantly. > Would you be willing just to let the subject go? I think everyone is tired of this thread. -MC P.S. - Just a clarification: Using SIPp is a "fake" test to do 200/400/600/xxx calls per second. Getting thousands of actual humans making actual calls is a "real" test. In other words, since SIPp can only do so much its test results are of limited value in the real world. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/3f89ddf2/attachment.html From msc at freeswitch.org Fri Aug 20 16:59:17 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Aug 2010 16:59:17 -0700 Subject: [Freeswitch-users] Webapi Examples location In-Reply-To: References: <4C6EE9A7.4050108@gmail.com> Message-ID: On Fri, Aug 20, 2010 at 3:49 PM, jesse zhao wrote: > why my message post to the mail list gets blocked? testing by reply . > Not blocked, just moderated until you join the mailing list. Once you're verified you should be good to go. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100820/617ba6d8/attachment.html From chat2jesse at gmail.com Fri Aug 20 17:30:04 2010 From: chat2jesse at gmail.com (jesse zhao) Date: Fri, 20 Aug 2010 17:30:04 -0700 Subject: [Freeswitch-users] scalability of FS vs yate Message-ID: I am wondering whether someone did testing of FreeSwitch vs Yate under the same hardware system and testing profiles. Which one is more scalable? Any usage case of FS in large service providers or organization in terms of 100K subscribers? It is nice to list customers of FS in the wiki page. thanks! -jesse From fdelawarde at wirelessmundi.com Fri Aug 20 09:21:21 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Fri, 20 Aug 2010 18:21:21 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <1282316031103-5444786.post@n2.nabble.com> <20100820175224.5d472a21@anubis.defcon1> Message-ID: <1282321281.16719.163.camel@luna.tc.commsmundi.com> Stress test results are only useful to make the headlines or for PhD students, those tests should only be made by experts (unbiased if possible) that and results should be shown in a special "cool-pure-raw performance testing" page. For the rest of us, real life scenarios with a small description of the modules involved would be useful to help with server dimensioning. If people contribute, with enough data it could give a pretty nice idea of what to expect. Of course those results can vary a lot so a huge disclaimer must be there to avoid having special people (mainly US citizens) suing the wiki owner. ;-) Fran?ois. On Fri, 2010-08-20 at 11:01 -0500, Anthony Minessale wrote: > Its an ok thing as long as it comes with a disclaimer that "results > may vary" and "you have to know what you are doing" > And those who work on it need to volunteer to help with the emails it > may generate asking why they can't match the results =D > > > On Fri, Aug 20, 2010 at 10:52 AM, Jean-Yves F. Barbier <12ukwn at gmail.com> wrote: > > Le Fri, 20 Aug 2010 10:10:32 -0500, > > Anthony Minessale a ?crit : > > > >> The problem is that performance testing is very fluid and relevant to > >> each machine and OS and test performed and you can't get concrete > >> expectations without some advanced skills in deploying FS. > > > > Well, it could have a real interest by adding the tweaked parameters, > > may be there are some, yet unexplored, that would raise the response. > > > > -- > > Why don't elephants eat penguins ? > > Because they can't get the wrappers off ... > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > From mnhassan at usa.net Fri Aug 20 20:02:44 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sat, 21 Aug 2010 09:02:44 +0600 Subject: [Freeswitch-users] scalability of FS vs yate In-Reply-To: References: Message-ID: Hi, Depends on what you use it for, FS / Yate can be used in so many ways, so many scenarios, that each implementation may not use very essential features of the next one. 6 months ago, I was looking for a platform to build our new service launch, and evaluated Asterisk, Yate, Kamailio and Freeswitch. While each can be put to work for us, what made FreeSwitch the final choice for me was the passion of the core devs (particularly Anthony Minnesale), and the community that has been built around it. And, most importantly, they fix "any bug" within a day at the worst, mostly coming within hours! That is what I can bank on as our core switching platform, open-source or not. Regards HASSAN On Sat, Aug 21, 2010 at 06:30, jesse zhao wrote: > I am wondering whether someone did testing of FreeSwitch vs Yate under > the same hardware system and testing profiles. Which one is more > scalable? > > Any usage case of FS in large service providers or organization in > terms of 100K subscribers? It is nice to list customers of FS in the > wiki page. > > > thanks! > > -jesse > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/435b4287/attachment.html From brokendash at gmail.com Fri Aug 20 20:40:17 2010 From: brokendash at gmail.com (broken dash) Date: Fri, 20 Aug 2010 22:40:17 -0500 Subject: [Freeswitch-users] start_dtmf on BLeg Channel from the dialplan In-Reply-To: References: Message-ID: hey i had some issues with this before and wanted to ask if FS is actually seeing the DTMF events... are you seeing it in the logs at all? --example-- 2010-08-20 22:18:23.524714 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000 --/example-- thx, B On Wed, Aug 18, 2010 at 8:23 AM, Phillip Jones wrote: > Thanks. > > That didn't do it. Neither did export nolocal:execute_on_answer=start_dtmf > > umm... > > > On Tue, Aug 17, 2010 at 7:17 PM, Michael Collins wrote: >> >> Try this and see if it helps: >> >> http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_app >> -MC >> >> On Tue, Aug 17, 2010 at 3:11 PM, Phillip Jones >> wrote: >>> >>> Hi there, >>> >>> Is there a way to execute?start_dtmf on the B Leg, from the >>> dialplan,?before you bridge call? >>> >>> >>> I am using group_confirm, so that a called user is bridged when they >>> answer and press 1. >>> >>> With one of my carriers, DTMF is not working - pressing 1 does nothing. >>> >>> If I use group_confirm=exec and send the B leg to a script on answer, and >>> execute "start_dtmf" on the b leg session in the script, all works as >>> expected and pressing 1 bridges the call. >>> >>> I would perfer not to use a script if there is a way of achieving this >>> via the dialplan. >>> >>> Any help appreciated. >>> >>> Thanks >>> >>> Pj >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From chat2jesse at gmail.com Sat Aug 21 00:08:47 2010 From: chat2jesse at gmail.com (jesse) Date: Sat, 21 Aug 2010 00:08:47 -0700 Subject: [Freeswitch-users] simple echo failed Message-ID: What the heck is wrong with my system even simple "originate user/1001 & echo" doesn't work???? freeswitch at xyz-host> originate user/1001 & echo 2010-08-21 00:00:29.647778 [NOTICE] switch_channel.c:776 New Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [21277012-acc5-464c-b415-595636e5ad73] 2010-08-21 00:00:29.892892 [INFO] sofia.c:662 sofia/internal/sip:1001 at 192.168.2.111:16176 Update Callee ID to "1001" <1001> 2010-08-21 00:00:29.895316 [NOTICE] sofia.c:4389 Ring-Ready sofia/internal/sip:1001 at 192.168.2.111:16176! 2010-08-21 00:00:31.727349 [WARNING] sofia_reg.c:387 google.com Failed Registration, setting retry to 30 seconds. 2010-08-21 00:00:33.830453 [NOTICE] sofia.c:4875 Channel [sofia/internal/sip:1001 at 192.168.2.111:16176] has been answered 2010-08-21 00:00:33.830453 [NOTICE] switch_ivr.c:1475 Transfer sofia/internal/sip:1001 at 192.168.2.111:16176 to echo[&@default] +OK 21277012-acc5-464c-b415-595636e5ad73 2010-08-21 00:00:33.831470 [NOTICE] switch_core_state_machine.c:131 Hangup sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_ROUTING] [NO_ROUTE_DESTINATION] freeswitch at xyz-host> 2010-08-21 00:00:33.837480 [NOTICE] switch_core_session.c:1193 Session 31 (sofia/internal/sip:1001 at 192.168.2.111:16176) Ended 2010-08-21 00:00:33.837480 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_DESTROY] thanks! jesse From gmaruzz at celliax.org Sat Aug 21 00:22:55 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 21 Aug 2010 09:22:55 +0200 Subject: [Freeswitch-users] simple echo failed In-Reply-To: References: Message-ID: originate user/1001 &echo() note the parenthesis On Sat, Aug 21, 2010 at 9:08 AM, jesse wrote: > What the heck is wrong with my system even simple "originate user/1001 > & echo" ?doesn't work???? > > > > freeswitch at xyz-host> originate user/1001 & echo > 2010-08-21 00:00:29.647778 [NOTICE] switch_channel.c:776 New Channel > sofia/internal/sip:1001 at 192.168.2.111:16176 > [21277012-acc5-464c-b415-595636e5ad73] > 2010-08-21 00:00:29.892892 [INFO] sofia.c:662 > sofia/internal/sip:1001 at 192.168.2.111:16176 Update Callee ID to "1001" > <1001> > 2010-08-21 00:00:29.895316 [NOTICE] sofia.c:4389 Ring-Ready > sofia/internal/sip:1001 at 192.168.2.111:16176! > 2010-08-21 00:00:31.727349 [WARNING] sofia_reg.c:387 google.com Failed > Registration, setting retry to 30 seconds. > 2010-08-21 00:00:33.830453 [NOTICE] sofia.c:4875 Channel > [sofia/internal/sip:1001 at 192.168.2.111:16176] has been answered > 2010-08-21 00:00:33.830453 [NOTICE] switch_ivr.c:1475 Transfer > sofia/internal/sip:1001 at 192.168.2.111:16176 to echo[&@default] > > +OK 21277012-acc5-464c-b415-595636e5ad73 > > 2010-08-21 00:00:33.831470 [NOTICE] switch_core_state_machine.c:131 > Hangup sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_ROUTING] > [NO_ROUTE_DESTINATION] > freeswitch at xyz-host> 2010-08-21 00:00:33.837480 [NOTICE] > switch_core_session.c:1193 Session 31 > (sofia/internal/sip:1001 at 192.168.2.111:16176) Ended > 2010-08-21 00:00:33.837480 [NOTICE] switch_core_session.c:1195 Close > Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_DESTROY] > > thanks! > > jesse > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From chat2jesse at gmail.com Sat Aug 21 00:26:04 2010 From: chat2jesse at gmail.com (jesse) Date: Sat, 21 Aug 2010 00:26:04 -0700 Subject: [Freeswitch-users] simple echo failed In-Reply-To: References: Message-ID: the same error for park or hold originate user/1001 & park() 2010-08-21 00:24:25.773265 [NOTICE] switch_channel.c:776 New Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [02ec9688-14af-4c89-af0b-80789fdd99c5] 2010-08-21 00:24:26.011268 [INFO] sofia.c:662 sofia/internal/sip:1001 at 192.168.2.111:16176 Update Callee ID to "1001" <1001> 2010-08-21 00:24:26.013562 [NOTICE] sofia.c:4389 Ring-Ready sofia/internal/sip:1001 at 192.168.2.111:16176! 2010-08-21 00:24:27.643359 [NOTICE] sofia.c:4875 Channel [sofia/internal/sip:1001 at 192.168.2.111:16176] has been answered +OK 02ec9688-14af-4c89-af0b-80789fdd99c5 2010-08-21 00:24:27.644395 [NOTICE] switch_ivr.c:1475 Transfer sofia/internal/sip:1001 at 192.168.2.111:16176 to park()[&@default] 2010-08-21 00:24:27.644395 [NOTICE] switch_core_state_machine.c:131 Hangup sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_ROUTING] [NO_ROUTE_DESTINATION] freeswitch at xyz-host> 2010-08-21 00:24:27.651750 [NOTICE] switch_core_session.c:1193 Session 34 (sofia/internal/sip:1001 at 192.168.2.111:16176) Ended 2010-08-21 00:24:27.651750 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_DESTROY] 2010-08-21 00:24:29.133269 [WARNING] sofia_reg.c:387 google.com Failed Registration, setting retry to 30 seconds. On Sat, Aug 21, 2010 at 12:08 AM, jesse wrote: > What the heck is wrong with my system even simple "originate user/1001 > & echo" ?doesn't work???? > > > > freeswitch at xyz-host> originate user/1001 & echo > 2010-08-21 00:00:29.647778 [NOTICE] switch_channel.c:776 New Channel > sofia/internal/sip:1001 at 192.168.2.111:16176 > [21277012-acc5-464c-b415-595636e5ad73] > 2010-08-21 00:00:29.892892 [INFO] sofia.c:662 > sofia/internal/sip:1001 at 192.168.2.111:16176 Update Callee ID to "1001" > <1001> > 2010-08-21 00:00:29.895316 [NOTICE] sofia.c:4389 Ring-Ready > sofia/internal/sip:1001 at 192.168.2.111:16176! > 2010-08-21 00:00:31.727349 [WARNING] sofia_reg.c:387 google.com Failed > Registration, setting retry to 30 seconds. > 2010-08-21 00:00:33.830453 [NOTICE] sofia.c:4875 Channel > [sofia/internal/sip:1001 at 192.168.2.111:16176] has been answered > 2010-08-21 00:00:33.830453 [NOTICE] switch_ivr.c:1475 Transfer > sofia/internal/sip:1001 at 192.168.2.111:16176 to echo[&@default] > > +OK 21277012-acc5-464c-b415-595636e5ad73 > > 2010-08-21 00:00:33.831470 [NOTICE] switch_core_state_machine.c:131 > Hangup sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_ROUTING] > [NO_ROUTE_DESTINATION] > freeswitch at xyz-host> 2010-08-21 00:00:33.837480 [NOTICE] > switch_core_session.c:1193 Session 31 > (sofia/internal/sip:1001 at 192.168.2.111:16176) Ended > 2010-08-21 00:00:33.837480 [NOTICE] switch_core_session.c:1195 Close > Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_DESTROY] > > thanks! > > jesse > From chat2jesse at gmail.com Sat Aug 21 00:33:19 2010 From: chat2jesse at gmail.com (jesse) Date: Sat, 21 Aug 2010 00:33:19 -0700 Subject: [Freeswitch-users] simple echo failed In-Reply-To: References: Message-ID: I tried . the same error. why it is : NO_ROUTE_DESTINATION??? freeswitch at xyz-host> originate user/1001 & echo() 2010-08-21 00:31:54.475606 [NOTICE] switch_channel.c:776 New Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [92d6ba77-9093-412c-a94b-6424aa6e6033] 2010-08-21 00:31:54.700019 [INFO] sofia.c:662 sofia/internal/sip:1001 at 192.168.2.111:16176 Update Callee ID to "1001" <1001> 2010-08-21 00:31:54.703318 [NOTICE] sofia.c:4389 Ring-Ready sofia/internal/sip:1001 at 192.168.2.111:16176! 2010-08-21 00:31:56.513024 [NOTICE] sofia.c:4875 Channel [sofia/internal/sip:1001 at 192.168.2.111:16176] has been answered +OK 92d6ba77-9093-412c-a94b-6424aa6e6033 2010-08-21 00:31:56.514060 [NOTICE] switch_ivr.c:1475 Transfer sofia/internal/sip:1001 at 192.168.2.111:16176 to echo()[&@default] 2010-08-21 00:31:56.514060 [NOTICE] switch_core_state_machine.c:131 Hangup sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_ROUTING] [NO_ROUTE_DESTINATION] freeswitch at xyz-host> 2010-08-21 00:31:56.521549 [NOTICE] switch_core_session.c:1193 Session 35 (sofia/internal/sip:1001 at 192.168.2.111:16176) Ended 2010-08-21 00:31:56.521549 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_DESTROY] On Sat, Aug 21, 2010 at 12:22 AM, Giovanni Maruzzelli wrote: > originate user/1001 &echo() > > note the parenthesis > > On Sat, Aug 21, 2010 at 9:08 AM, jesse wrote: >> What the heck is wrong with my system even simple "originate user/1001 >> & echo" ?doesn't work???? >> >> >> >> freeswitch at xyz-host> originate user/1001 & echo >> 2010-08-21 00:00:29.647778 [NOTICE] switch_channel.c:776 New Channel >> sofia/internal/sip:1001 at 192.168.2.111:16176 >> [21277012-acc5-464c-b415-595636e5ad73] >> 2010-08-21 00:00:29.892892 [INFO] sofia.c:662 >> sofia/internal/sip:1001 at 192.168.2.111:16176 Update Callee ID to "1001" >> <1001> >> 2010-08-21 00:00:29.895316 [NOTICE] sofia.c:4389 Ring-Ready >> sofia/internal/sip:1001 at 192.168.2.111:16176! >> 2010-08-21 00:00:31.727349 [WARNING] sofia_reg.c:387 google.com Failed >> Registration, setting retry to 30 seconds. >> 2010-08-21 00:00:33.830453 [NOTICE] sofia.c:4875 Channel >> [sofia/internal/sip:1001 at 192.168.2.111:16176] has been answered >> 2010-08-21 00:00:33.830453 [NOTICE] switch_ivr.c:1475 Transfer >> sofia/internal/sip:1001 at 192.168.2.111:16176 to echo[&@default] >> >> +OK 21277012-acc5-464c-b415-595636e5ad73 >> >> 2010-08-21 00:00:33.831470 [NOTICE] switch_core_state_machine.c:131 >> Hangup sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_ROUTING] >> [NO_ROUTE_DESTINATION] >> freeswitch at xyz-host> 2010-08-21 00:00:33.837480 [NOTICE] >> switch_core_session.c:1193 Session 31 >> (sofia/internal/sip:1001 at 192.168.2.111:16176) Ended >> 2010-08-21 00:00:33.837480 [NOTICE] switch_core_session.c:1195 Close >> Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_DESTROY] >> >> thanks! >> >> jesse >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chat2jesse at gmail.com Sat Aug 21 00:37:24 2010 From: chat2jesse at gmail.com (jesse) Date: Sat, 21 Aug 2010 00:37:24 -0700 Subject: [Freeswitch-users] simple echo failed In-Reply-To: References: Message-ID: freaking stupid, it because the extra space between & e. why it is designed like this? so error prone and misleading -jesse On Sat, Aug 21, 2010 at 12:33 AM, jesse wrote: > I tried . the same error. why it is : NO_ROUTE_DESTINATION??? > > > freeswitch at xyz-host> originate user/1001 & echo() > 2010-08-21 00:31:54.475606 [NOTICE] switch_channel.c:776 New Channel > sofia/internal/sip:1001 at 192.168.2.111:16176 > [92d6ba77-9093-412c-a94b-6424aa6e6033] > 2010-08-21 00:31:54.700019 [INFO] sofia.c:662 > sofia/internal/sip:1001 at 192.168.2.111:16176 Update Callee ID to "1001" > <1001> > 2010-08-21 00:31:54.703318 [NOTICE] sofia.c:4389 Ring-Ready > sofia/internal/sip:1001 at 192.168.2.111:16176! > 2010-08-21 00:31:56.513024 [NOTICE] sofia.c:4875 Channel > [sofia/internal/sip:1001 at 192.168.2.111:16176] has been answered > > +OK 92d6ba77-9093-412c-a94b-6424aa6e6033 > > 2010-08-21 00:31:56.514060 [NOTICE] switch_ivr.c:1475 Transfer > sofia/internal/sip:1001 at 192.168.2.111:16176 to echo()[&@default] > 2010-08-21 00:31:56.514060 [NOTICE] switch_core_state_machine.c:131 > Hangup sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_ROUTING] > [NO_ROUTE_DESTINATION] > freeswitch at xyz-host> 2010-08-21 00:31:56.521549 [NOTICE] > switch_core_session.c:1193 Session 35 > (sofia/internal/sip:1001 at 192.168.2.111:16176) Ended > 2010-08-21 00:31:56.521549 [NOTICE] switch_core_session.c:1195 Close > Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_DESTROY] > > > > > On Sat, Aug 21, 2010 at 12:22 AM, Giovanni Maruzzelli > wrote: >> originate user/1001 &echo() >> >> note the parenthesis >> >> On Sat, Aug 21, 2010 at 9:08 AM, jesse wrote: >>> What the heck is wrong with my system even simple "originate user/1001 >>> & echo" ?doesn't work???? >>> >>> >>> >>> freeswitch at xyz-host> originate user/1001 & echo >>> 2010-08-21 00:00:29.647778 [NOTICE] switch_channel.c:776 New Channel >>> sofia/internal/sip:1001 at 192.168.2.111:16176 >>> [21277012-acc5-464c-b415-595636e5ad73] >>> 2010-08-21 00:00:29.892892 [INFO] sofia.c:662 >>> sofia/internal/sip:1001 at 192.168.2.111:16176 Update Callee ID to "1001" >>> <1001> >>> 2010-08-21 00:00:29.895316 [NOTICE] sofia.c:4389 Ring-Ready >>> sofia/internal/sip:1001 at 192.168.2.111:16176! >>> 2010-08-21 00:00:31.727349 [WARNING] sofia_reg.c:387 google.com Failed >>> Registration, setting retry to 30 seconds. >>> 2010-08-21 00:00:33.830453 [NOTICE] sofia.c:4875 Channel >>> [sofia/internal/sip:1001 at 192.168.2.111:16176] has been answered >>> 2010-08-21 00:00:33.830453 [NOTICE] switch_ivr.c:1475 Transfer >>> sofia/internal/sip:1001 at 192.168.2.111:16176 to echo[&@default] >>> >>> +OK 21277012-acc5-464c-b415-595636e5ad73 >>> >>> 2010-08-21 00:00:33.831470 [NOTICE] switch_core_state_machine.c:131 >>> Hangup sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_ROUTING] >>> [NO_ROUTE_DESTINATION] >>> freeswitch at xyz-host> 2010-08-21 00:00:33.837480 [NOTICE] >>> switch_core_session.c:1193 Session 31 >>> (sofia/internal/sip:1001 at 192.168.2.111:16176) Ended >>> 2010-08-21 00:00:33.837480 [NOTICE] switch_core_session.c:1195 Close >>> Channel sofia/internal/sip:1001 at 192.168.2.111:16176 [CS_DESTROY] >>> >>> thanks! >>> >>> jesse >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From tculjaga at gmail.com Sat Aug 21 01:24:43 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 21 Aug 2010 10:24:43 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> Message-ID: On Fri, Aug 20, 2010 at 11:24 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > probably missing ldconfig > or LD_LIBRARY_PATH > > On Fri, Aug 20, 2010 at 4:10 PM, Sadjad Seyed-Ahmadian > wrote: > > I got the pint files are in /usr/local/lib and I make a link in > > /isr/local/freeswitch/lib to them, but when I try to load mod_h323 > > freeswitch crashed!! > > > Are u sure you are following the procedure on http://wiki.freeswitch.org/wiki/Mod_h323 ? it just depends what your PATH is to find the relevant libraries. Also you can create a link as you did to within ./freeswitch/lib/ directory. anyhow: libh323_linux_x86_.so.1.22.0 libpt.so.2.9-beta0 this is what i use and no crash... now, it may be you are missing some relevant information in the h323.conf.xml ... please check the file is there (./freeswitch/conf/autoload_configs/) and that its well configured. this is what i have here and it works nice: tculjaga at nemesis:~/h323/ptlib$ cat /usr/local/freeswitch/conf/autoload_configs/h323.conf.xml tculjaga at nemesis:~/h323/ptlib$ here is the module loading freeswitch at nemesis> +OK console log level set to DEBUG freeswitch at nemesis> freeswitch at nemesis> freeswitch at nemesis> freeswitch at nemesis> load mod_h323 2010-08-21 10:22:06.706085 [CONSOLE] mod_h323.cpp:147 Starting loading mod_h323 2010-08-21 10:22:06.720299 [DEBUG] mod_h323.cpp:356 ======>FSProcess::Initialise [0xb7068668] 2010-08-21 10:22:06.722553 [DEBUG] mod_h323.cpp:591 ======>FSH323EndPoint::FSH323EndPoint [0x8652b38] 2010-08-21 10:22:06.722553 [DEBUG] mod_h323.cpp:364 ======>FSManager::Initialise [0x8652b38] 2010-08-21 10:22:06.722553 [DEBUG] mod_h323.cpp:446 ======>FSH323EndPoint::ReadConfig [0x8652b38] 2010-08-21 10:22:06.722553 [DEBUG] mod_h323.cpp:568 Created Listener 'default' 2010-08-21 10:22:06.722553 [DEBUG] mod_h323.cpp:378 Config capability PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.722553 [DEBUG] mod_h323.cpp:383 Find capability PCMA to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.723537 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-ALaw-64k{sw}" 2010-08-21 10:22:06.723537 [DEBUG] h323caps.cxx:3201 Added capability: G.711-ALaw-64k <1> 2010-08-21 10:22:06.723537 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-ALaw-64k{sw}" 2010-08-21 10:22:06.723537 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-ALaw-64k{sw}" 2010-08-21 10:22:06.723537 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-ALaw-64k{sw}" 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-ALaw-64k{sw}" 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:2014 No Extended Capabilities found to load 2010-08-21 10:22:06.724528 [DEBUG] mod_h323.cpp:399 H.323 added 1 capabilities 'G.711-ALaw-64k*{sw}' 2010-08-21 10:22:06.724528 [DEBUG] mod_h323.cpp:383 Find capability PCMU to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-uLaw-64k{sw}" 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3201 Added capability: G.711-uLaw-64k <2> 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-uLaw-64k{sw}" 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-uLaw-64k{sw}" 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-uLaw-64k{sw}" 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3289 FindCapability: "G.711-uLaw-64k{sw}" 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:2014 No Extended Capabilities found to load 2010-08-21 10:22:06.724528 [DEBUG] mod_h323.cpp:399 H.323 added 1 capabilities 'G.711-uLaw-64k*{sw}' 2010-08-21 10:22:06.724528 [DEBUG] mod_h323.cpp:383 Find capability GSM to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3289 FindCapability: "GSM-06.10{sw}" 2010-08-21 10:22:06.724528 [DEBUG] h323caps.cxx:3201 Added capability: GSM-06.10{sw} <3> 2010-08-21 10:22:06.725872 [DEBUG] h323caps.cxx:3289 FindCapability: "GSM-06.10{sw}" 2010-08-21 10:22:06.725872 [DEBUG] h323caps.cxx:3298 Found capability: GSM-06.10{sw} <3> 2010-08-21 10:22:06.725872 [DEBUG] h323caps.cxx:3289 FindCapability: "GSM-06.10{sw}" 2010-08-21 10:22:06.725872 [DEBUG] h323caps.cxx:3298 Found capability: GSM-06.10{sw} <3> 2010-08-21 10:22:06.725872 [DEBUG] h323caps.cxx:3289 FindCapability: "GSM-06.10{sw}" 2010-08-21 10:22:06.725872 [DEBUG] h323caps.cxx:3298 Found capability: GSM-06.10{sw} <3> 2010-08-21 10:22:06.726883 [DEBUG] h323caps.cxx:3289 FindCapability: "GSM-06.10{sw}" 2010-08-21 10:22:06.726883 [DEBUG] h323caps.cxx:3298 Found capability: GSM-06.10{sw} <3> 2010-08-21 10:22:06.726883 [DEBUG] h323caps.cxx:2014 No Extended Capabilities found to load 2010-08-21 10:22:06.726883 [DEBUG] mod_h323.cpp:399 H.323 added 1 capabilities 'GSM-06.10*{sw}' 2010-08-21 10:22:06.726883 [DEBUG] mod_h323.cpp:383 Find capability G723 to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.726883 [DEBUG] mod_h323.cpp:383 Find capability G729b to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.726883 [DEBUG] mod_h323.cpp:383 Find capability G729 to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.726883 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A/B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3201 Added capability: G.729A/B{sw} <4> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3201 Added capability: G.729A{sw} <5> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3201 Added capability: G.729B{sw} <6> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3201 Added capability: G.729{sw} <7> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A/B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729A/B{sw} <4> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729A{sw} <5> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729B{sw} <6> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729{sw} <7> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A/B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729A/B{sw} <4> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729A{sw} <5> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729B{sw} <6> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729{sw} <7> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A/B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729A/B{sw} <4> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729A{sw} <5> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729B{sw} <6> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729{sw} <7> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A/B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729A/B{sw} <4> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729A{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729A{sw} <5> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729B{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729B{sw} <6> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3289 FindCapability: "G.729{sw}" 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:3298 Found capability: G.729{sw} <7> 2010-08-21 10:22:06.727891 [DEBUG] h323caps.cxx:2014 No Extended Capabilities found to load 2010-08-21 10:22:06.727891 [DEBUG] mod_h323.cpp:399 H.323 added 4 capabilities 'G.729*{sw}' 2010-08-21 10:22:06.727891 [DEBUG] mod_h323.cpp:383 Find capability G729a to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.727891 [DEBUG] mod_h323.cpp:383 Find capability G729ab to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.727891 [DEBUG] mod_h323.cpp:383 Find capability G723.1 to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.735813 [DEBUG] mod_h323.cpp:383 Find capability G723.1-5k3 to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.735816 [DEBUG] mod_h323.cpp:383 Find capability G723.1a-5k3 to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.735816 [DEBUG] mod_h323.cpp:383 Find capability G723.1a-6k3 to PCMA,PCMU,GSM,G729 2010-08-21 10:22:06.735816 [DEBUG] mod_h323.cpp:412 --->fax_asn 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: T.38-IFP-COR <8> 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: UserInput/hookflash <9> 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: UserInput/basicString <10> 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: UserInput/dtmf <11> 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: UserInput/RFC2833 <12> 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: UserInput/Navigation <13> 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: UserInput/Softkey <14> 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: UserInput/PointDevice <15> 2010-08-21 10:22:06.735816 [DEBUG] h323caps.cxx:3201 Added capability: UserInput/Modal <16> 2010-08-21 10:22:06.735816 [DEBUG] osutil.cxx:188 File handle high water mark set: 53 PTCPSocket 2010-08-21 10:22:06.735816 [DEBUG] h323ep.cxx:2033 Started listener Listener[ip$192.168.1.231:1720] 2010-08-21 10:22:06.735816 [DEBUG] tlibthrd.cxx:519 Thread high water mark set: 3 2010-08-21 10:22:06.735816 [CONSOLE] mod_h323.cpp:164 H323 mod initialized and running 2010-08-21 10:22:06.735816 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_h323] 2010-08-21 10:22:06.735816 [NOTICE] switch_loadable_module.c:145 Adding Endpoint 'h323' 2010-08-21 10:22:06.735816 [DEBUG] tlibthrd.cxx:431 Started thread 0xb7068ee8 H323 Listener:b6c58b70 2010-08-21 10:22:06.735816 [DEBUG] transports.cxx:1521 Awaiting TCP connections on port 1720 2010-08-21 10:22:06.735816 [DEBUG] transports.cxx:1475 Waiting on socket accept on ip$192.168.1.231:1720 2010-08-21 10:22:06.735816 [DEBUG] tlibthrd.cxx:940 PThread::PXBlockOnIO(53,2) +OK freeswitch at nemesis> and this is the vrsion im running: freeswitch at nemesis> freeswitch at nemesis> freeswitch at nemesis> version FreeSWITCH Version 1.0.head (git-018f4d6 2010-08-10 19-52-08 +0200) freeswitch at nemesis> let me know if you still have issues. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/9346b775/attachment-0001.html From ssa1357 at yahoo.com Sat Aug 21 01:37:35 2010 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Sat, 21 Aug 2010 01:37:35 -0700 (PDT) Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> Message-ID: <305309.80502.qm@web53403.mail.re2.yahoo.com> You were right I installed from the start and file are in /usr/local/freeswitch/lib now. but freeswitch crash on loading mod_h323. here is the log: freeswitch> load mod_h323 *** glibc detected *** /usr/local/freeswitch/bin/freeswitch: malloc(): memory corruption: 0x0906b488 *** ======= Backtrace: ========= /lib/libc.so.6[0xb0a2dd] /lib/libc.so.6(__libc_malloc+0x67)[0xb0be97] /usr/lib/libstdc++.so.6(_Znwj+0x27)[0x798ab7] /usr/lib/libstdc++.so.6(_ZNSs4_Rep9_S_createEjjRKSaIcE+0x6b)[0x7740fb] /usr/lib/libstdc++.so.6[0x774ef5] /usr/lib/libstdc++.so.6(_ZNSsC1EPKcRKSaIcE+0x47)[0x775107] /usr/local/freeswitch/lib/libpt.so.2.9-beta0(_ZN8PFactoryI10PURLSchemeSsE11GetInstanceEv+0x2d)[0x68efb59] /usr/local/freeswitch/lib/libpt.so.2.9-beta0(_ZN8PFactoryI10PURLSchemeSsE8RegisterERKSsPNS1_10WorkerBaseE+0x17)[0x68efc1d] /usr/local/freeswitch/lib/libpt.so.2.9-beta0(_ZN8PFactoryI10PURLSchemeSsE6WorkerI21PURLLegacyScheme_fileEC1ERKSsb+0x36)[0x68f0168] /usr/local/freeswitch/lib/libpt.so.2.9-beta0[0x68ec0e4] /usr/local/freeswitch/lib/libpt.so.2.9-beta0[0x69807a6] /usr/local/freeswitch/lib/libpt.so.2.9-beta0[0x686d1ad] /lib/ld-linux.so.2[0xa8a223] /lib/ld-linux.so.2[0xa8a333] /lib/ld-linux.so.2[0xa8ddaa] /lib/ld-linux.so.2[0xa89e66] /lib/ld-linux.so.2[0xa8d4b2] /lib/libdl.so.2[0xbfac6d] /lib/ld-linux.so.2[0xa89e66] /lib/libdl.so.2[0xbfb2ec] /lib/libdl.so.2(dlopen+0x44)[0xbfaba4] /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_dso_open+0x60)[0x5561a0] /usr/local/freeswitch/lib/libfreeswitch.so.1[0x55b92b] /usr/local/freeswitch/mod/mod_commands.so[0x18add4] /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_api_execute+0xbd)[0x55717d] /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_console_execute+0x1e0)[0x532520] /usr/local/freeswitch/lib/libfreeswitch.so.1[0x53265b] /usr/local/freeswitch/lib/libfreeswitch.so.1[0x532fb8] /usr/local/freeswitch/lib/libfreeswitch.so.1[0x5c8906] /lib/libpthread.so.0[0xc06832] /lib/libc.so.6(clone+0x5e)[0xb71e0e] ======= Memory map: ======== 00101000-00141000 r-xp 00000000 fd:00 80584251 /usr/lib/libncurses.so.5.5 00141000-00149000 rw-p 00040000 fd:00 80584251 /usr/lib/libncurses.so.5.5 00149000-0014a000 rw-p 00149000 00:00 0 0014a000-00177000 r-xp 00000000 fd:00 80586115 /usr/lib/libgssapi_krb5.so.2.2 00177000-00178000 rw-p 0002d000 fd:00 80586115 /usr/lib/libgssapi_krb5.so.2.2 00178000-0017d000 r-xp 00000000 fd:00 81363254 /usr/local/freeswitch/mod/mod_loopback.so 0017d000-0017e000 rw-p 00004000 fd:00 81363254 /usr/local/freeswitch/mod/mod_loopback.so 0017e000-00191000 r-xp 00000000 fd:00 81363155 /usr/local/freeswitch/mod/mod_commands.so 00191000-00192000 rw-p 00013000 fd:00 81363155 /usr/local/freeswitch/mod/mod_commands.so 00192000-001ad000 r-xp 00000000 fd:00 81363156 /usr/local/freeswitch/mod/mod_conference.so 001ad000-001ae000 rw-p 0001a000 fd:00 81363156 /usr/local/freeswitch/mod/mod_conference.so 001ae000-001c0000 r-xp 00000000 fd:00 81363238 /usr/local/freeswitch/mod/mod_dptools.so 001c0000-001c1000 rw-p 00011000 fd:00 81363238 /usr/local/freeswitch/mod/mod_dptools.so 001c1000-001ce000 r-xp 00000000 fd:00 81363179 /usr/local/freeswitch/mod/mod_fifo.so 001ce000-001cf000 rw-p 0000c000 fd:00 81363179 /usr/local/freeswitch/mod/mod_fifo.so 001cf000-001e6000 r-xp 00000000 fd:00 81363268 /usr/local/freeswitch/mod/mod_voicemail.so 001e6000-001e7000 rw-p 00017000 fd:00 81363268 /usr/local/freeswitch/mod/mod_voicemail.so 001e7000-001ed000 r-xp 00000000 fd:00 81363251 /usr/local/freeswitch/mod/mod_limit.so 001ed000-001ee000 rw-p 00006000 fd:00 81363251 /usr/local/freeswitch/mod/mod_limit.so 001ee000-001f0000 r-xp 00000000 fd:00 81363174 /usr/local/freeswitch/mod/mod_esf.so 001f0000-001f1000 rw-p 00001000 fd:00 81363174 /usr/local/freeswitch/mod/mod_esf.so 001f1000-001f4000 r-xp 00000000 fd:00 81363246 /usr/local/freeswitch/mod/mod_fsv.so 001f4000-001f5000 rw-p 00002000 fd:00 81363246 /usr/local/freeswitch/mod/mod_fsv.so 001f5000-001f8000 r-xp 00000000 fd:00 81363154 /usr/local/freeswitch/mod/mod_cluechoo.so 001f8000-001f9000 rw-p 00002000 fd:00 81363154 /usr/local/freeswitch/mod/mod_cluechoo.so 001f9000-001fd000 rw-p 001f9000 00:00 0 001fd000-00200000 r-xp 00000000 fd:00 81363267 /usr/local/freeswitch/mod/mod_valet_parking.so 00200000-00201000 rw-p 00002000 fd:00 81363267 /usr/local/freeswitch/mod/mod_valet_parking.so 00201000-00205000 r-xp 00000000 fd:00 81363227 /usr/local/freeswitch/mod/mod_dialplan_xml.so 00205000-00206000 rw-p 00003000 fd:00 81363227 /usr/local/freeswitch/mod/mod_dialplan_xml.so 00206000-00209000 r-xp 00000000 fd:00 81363166 /usr/local/freeswitch/mod/mod_dialplan_asterisk.so 00209000-0020a000 rw-p 00002000 fd:00 81363166 /usr/local/freeswitch/mod/mod_dialplan_asterisk.so 0020a000-0021b000 r-xp 00000000 fd:00 81363269 /usr/local/freeswitch/mod/mod_voipcodecs.so 0021b000-0021c000 rw-p 00010000 fd:00 81363269 /usr/local/freeswitch/mod/mod_voipcodecs.so 0021c000-0021d000 r-xp 00000000 fd:00 81363247 /usr/local/freeswitch/mod/mod_g723_1.so 0021d000-0021e000 rw-p 00000000 fd:00 81363247 /usr/local/freeswitch/mod/mod_g723_1.so 0021e000-0021f000 r-xp 00000000 fd:00 81363248 /usr/local/freeswitch/mod/mod_g729.so 0021f000-00220000 rw-p 00000000 fd:00 81363248 /usr/local/freeswitch/mod/mod_g729.so 00220000-0022d000 r-xp 00000000 fd:00 81363250 /usr/local/freeswitch/mod/mod_ilbc.so 0022d000-0022e000 rw-p 0000d000 fd:00 81363250 /usr/local/freeswitch/mod/mod_ilbc.so 0022e000-0022f000 r-xp 00000000 fd:00 81363249 /usr/local/freeswitch/mod/mod_h26x.so 0022f000-00230000 rw-p 00001000 fd:00 81363249 /usr/local/freeswitch/mod/mod_h26x.so 00230000-00231000 r-xp 00000000 fd:00 81363194 /usr/local/freeswitch/mod/mod_native_file.so 00231000-00232000 rw-p 00001000 fd:00 81363194 /usr/local/freeswitch/mod/mod_native_file.so 00232000-00234000 r-xp 00000000 fd:00 81363219 /usr/local/freeswitch/mod/mod_tone_stream.so 00234000-00235000 rw-p 00001000 fd:00 81363219 /usr/local/freeswitch/mod/mod_tone_stream.so 00235000-00236000 r-xp 00000000 fd:00 81363245 /usr/local/freeswitch/mod/mod_file_string.so 00236000-00237000 rw-p 00001000 fd:00 81363245 /usr/local/freeswitch/mod/mod_file_string.so 00238000-0023b000 r-xp 00000000 fd:00 81363153 /usr/local/freeswitch/mod/mod_cdr_csv.so 0023b000-0023c000 rw-p 00002000 fd:00 81363153 /usr/local/freeswitch/mod/mod_cdr_csv.so 0023c000-0024a000 r-xp 00000000 fd:00 81363258 /usr/local/freeswitch/mod/mod_siren.so 0024a000-0024b000 rw-p 0000e000 fd:00 81363258 /usr/local/freeswitch/mod/mod_siren.so 0024b000-00288000 r-xp 00000000 fd:00 81363260 /usr/local/freeswitch/mod/mod_spidermonkey.so 00288000-0028a000 rw-p 0003c000 fd:00 81363260 /usr/local/freeswitch/mod/mod_spidermonkey.so 0028a000-0028c000 r-xp 00000000 fd:00 81363208 /usr/local/freeswitch/mod/mod_spidermonkey_socket.so 0028c000-0028d000 rw-p 00002000 fd:00 81363208 /usr/local/freeswitch/mod/mod_spidermonkey_socket.so 0028d000-00295000 r-xp 00000000 fd:00 81363197 /usr/local/freeswitch/mod/mod_say_ru.so 00295000-00298000 rw-p 00008000 fd:00 81363197 /usr/local/freeswitch/mod/mod_say_ru.so 00298000-002a1000 r-xp 00000000 fd:00 80589604 /usr/lib/libesd.so.0.2.36 002a1000-002a2000 rw-p 00009000 fd:00 80589604 /usr/lib/libesd.so.0.2.36 002ab000-002bb000 r-xp 00000000 fd:00 43812182 /lib/libresolv-2.5.so 002bb000-002bc000 r--p 0000f000 fd:00 43812182 /lib/libresolv-2.5.so 002bc000-002bd000 rw-p 00010000 fd:00 43812182 /lib/libresolv-2.5.so 002bd000-002bf000 rw-p 002bd000 00:00 0 002c1000-002c3000 r-xp 00000000 fd:00 43812179 /lib/libcom_err.so.2.1 002c3000-002c4000 rw-p 00001000 fd:00 43812179 /lib/libcom_err.so.2.1 002c6000-003f0000 r-xp 00000000 fd:00 43811027 /lib/libcrypto.so.0.9.8e 003f0000-00403000 rw-p 00129000 fd:00 43811027 /lib/libcrypto.so.0.9.8e 00403000-00407000 rw-p 00403000 00:00 0 00409000-0042e000 r-xp 00000000 fd:00 80586112 /usr/lib/libk5crypto.so.3.1 0042e000-0042f000 rw-p 00025000 fd:00 80586112 /usr/lib/libk5crypto.so.3.1 00431000-00439000 r-xp 00000000 fd:00 80586111 /usr/lib/libkrb5support.so.0.1 00439000-0043a000 rw-p 00007000 fd:00 80586111 /usr/lib/libkrb5support.so.0.1 0043c000-004cf000 r-xp 00000000 fd:00 80586113 /usr/lib/libkrb5.so.3.3 004cf000-004d2000 rw-p 00092000 fd:00 80586113 /usr/lib/libkrb5.so.3.3 004d2000-004ea000 r-xp 00000000 fd:00 80584851 /usr/lib/libsasl2.so.2.0.22 004ea000-004eb000 rw-p 00017000 fd:00 80584851 /usr/lib/libsasl2.so.2.0.22 004f4000-00674000 r-xp 00000000 fd:00 81330474 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0 00674000-0067b000 rw-p 00180000 fd:00 81330474 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0 0067b000-00691000 rw-p 0067b000 00:00 0 00691000-006b6000 r-xp 00000000 fd:00 81330455 /usr/local/freeswitch/lib/libnspr4.so 006b6000-006b8000 rw-p 00024000 fd:00 81330455 /usr/local/freeswitch/lib/libnspr4.so 006b8000-006b9000 rw-p 006b8000 00:00 0 006b9000-006e1000 r-xp 00000000 fd:00 80588895 /usr/lib/libaudiofile.so.0.0.2 006e1000-006e4000 rw-p 00027000 fd:00 80588895 /usr/lib/libaudiofile.so.0.0.2 006e4000-007c4000 r-xp 00000000 fd:00 80588852 /usr/lib/libstdc++.so.6.0.8 007c4000-007c8000 r--p 000df000 fd:00 80588852 /usr/lib/libstdc++.so.6.0.8 007c8000-007c9000 rw-p 000e3000 fd:00 80588852 /usr/lib/libstdc++.so.6.0.8 007c9000-007cf000 rw-p 007c9000 00:00 0 007d1000-007dc000 r-xp 00000000 fd:00 43812190 /lib/libgcc_s-4.1.2-20080825.so.1 007dc000-007dd000 rw-p 0000a000 fd:00 43812190 /lib/libgcc_s-4.1.2-20080825.so.1 007dd000-00824000 r-xp 00000000 fd:00 81363255 /usr/local/freeswitch/mod/mod_lua.so 00824000-00826000 rw-p 00046000 fd:00 81363255 /usr/local/freeswitch/mod/mod_lua.so 0082b000-00832000 r-xp 00000000 fd:00 81363243 /usr/local/freeswitch/mod/mod_expr.so 00832000-00833000 rw-p 00007000 fd:00 81363243 /usr/local/freeswitch/mod/mod_expr.so 00833000-0088f000 r-xp 00000000 fd:00 81363259 /usr/local/freeswitch/mod/mod_sndfile.so 0088f000-00891000 rw-p 0005b000 fd:00 81363259 /usr/local/freeswitch/mod/mod_sndfile.so 00891000-00895000 rw-p 00891000 00:00 0 00895000-008c0000 r-xp 00000000 fd:00 81363033 /usr/local/freeswitch/mod/mod_h323.so 008c0000-008c3000 rw-p 0002a000 fd:00 81363033 /usr/local/freeswitch/mod/mod_h323.so 008de000-008e1000 r-xp 00000000 fd:00 81363213 /usr/local/freeswitch/mod/mod_spidermonkey_core_db.so 008e1000-008e2000 rw-p 00002000 fd:00 81363213 /usr/local/freeswitch/mod/mod_spidermonkey_core_db.so 008e6000-008f3000 r-xp 00000000 fd:00 81363242 /usr/local/freeswitch/mod/mod_event_socket.so 008f3000-008f4000 rw-p 0000c000 fd:00 81363242 /usr/local/freeswitch/mod/mod_event_socket.so 008f4000-0092d000 r-xp 00000000 fd:00 80587794 /usr/lib/libldap-2.3.so.0.2.31 0092d000-0092e000 rw-p 00039000 fd:00 80587794 /usr/lib/libldap-2.3.so.0.2.31 00966000-0096a000 r-xp 00000000 fd:00 81363257 /usr/local/freeswitch/mod/mod_say_en.so 0096a000-0096b000 rw-p 00003000 fd:00 81363257 /usr/local/freeswitch/mod/mod_say_en.so 0097c000-00981000 r-xp 00000000 fd:00 81363252 /usr/local/freeswitch/mod/mod_local_stream.so 00981000-00982000 rw-p 00004000 fd:00 81363252 /usr/local/freeswitch/mod/mod_local_stream.so 009b9000-009bb000 r-xp 00000000 fd:00 81363158 /usr/local/freeswitch/mod/mod_console.so 009bb000-009bc000 rw-p 00002000 fd:00 81363158 /usr/local/freeswitch/mod/mod_console.so 009bc000-009fa000 r-xp 00000000 fd:00 80584801 /usr/lib/libldap_r-2.3.so.0.2.31 009fa000-009fc000 rw-p 0003d000 fd:00 80584801 /usr/lib/libldap_r-2.3.so.0.2.31 009fc000-00a01000 rw-p 009fc000 00:00 0 00a1f000-00a33000 r-xp 00000000 fd:00 81363203 /usr/local/freeswitch/mod/mod_speex.so 00a33000-00a34000 rw-p 00014000 fd:00 81363203 /usr/local/freeswitch/mod/mod_speex.so 00a7c000-00a97000 r-xp 00000000 fd:00 43810819 /lib/ld-2.5.so 00a97000-00a98000 r--p 0001a000 fd:00 43810819 /lib/ld-2.5.so 00a98000-00a99000 rw-p 0001b000 fd:00 43810819 /lib/ld-2.5.so 00aa0000-00bf2000 r-xp 00000000 fd:00 43810833 /lib/libc-2.5.so 00bf2000-00bf4000 r--p 00152000 fd:00 43810833 /lib/libc-2.5.so 00bf4000-00bf5000 rw-p 00154000 fd:00 43810833 /lib/libc-2.5.so 00bf5000-00bf8000 rw-p 00bf5000 00:00 0 00bfa000-00bfd000 r-xp 00000000 fd:00 43810854 /lib/libdl-2.5.so 00bfd000-00bfe000 r--p 00002000 fd:00 43810854 /lib/libdl-2.5.so 00bfe000-00bff000 rw-p 00003000 fd:00 43810854 /lib/libdl-2.5.so 00c01000-00c16000 r-xp 00000000 fd:00 43810894 /lib/libpthread-2.5.so 00c16000-00c17000 r--p 00015000 fd:00 43810894 /lib/libpthread-2.5.so 00c17000-00c18000 rw-p 00016000 fd:00 43810894 /lib/libpthread-2.5.so 00c18000-00c1a000 rw-p 00c18000 00:00 0 00c1c000-00c2e000 r-xp 00000000 fd:00 80577691 /usr/lib/libz.so.1.2.3 00c2e000-00c2f000 rw-p 00011000 fd:00 80577691 /usr/lib/libz.so.1.2.3 00c31000-00c58000 r-xp 00000000 fd:00 43810848 /lib/libm-2.5.so 00c58000-00c59000 r--p 00026000 fd:00 43810848 /lib/libm-2.5.so 00c59000-00c5a000 rw-p 00027000 fd:00 43810848 /lib/libm-2.5.so 00c5c000-00c97000 r-xp 00000000 fd:00 43810872 /lib/libsepol.so.1 00c97000-00c98000 rw-p 0003b000 fd:00 43810872 /lib/libsepol.so.1 00c98000-00ca2000 rw-p 00c98000 00:00 0 00ca4000-00cba000 r-xp 00000000 fd:00 43810882 /lib/libselinux.so.1 00cba000-00cbc000 rw-p 00015000 fd:00 43810882 /lib/libselinux.so.1 00cbe000-00cc5000 r-xp 00000000 fd:00 43810897 /lib/librt-2.5.so 00cc5000-00cc6000 r--p 00007000 fd:00 43810897 /lib/librt-2.5.so 00cc6000-00cc7000 rw-p 00008000 fd:00 43810897 /lib/librt-2.5.so 00cc9000-00d0d000 r-xp 00000000 fd:00 43811690 /lib/libssl.so.0.9.8e 00d0d000-00d11000 rw-p 00043000 fd:00 43811690 /lib/libssl.so.0.9.8e 00d13000-00d20000 r-xp 00000000 fd:00 80584327 /usr/lib/liblber-2.3.so.0.2.31 00d20000-00d21000 rw-p 0000c000 fd:00 80584327 /usr/lib/liblber-2.3.so.0.2.31 00d24000-00d25000 r-xp 00000000 fd:00 81363151 /usr/local/freeswitch/mod/mod_amr.so 00d25000-00d26000 rw-p 00000000 fd:00 81363151 /usr/local/freeswitch/mod/mod_amr.so 00d4d000-00d4e000 r-xp 00d4d000 00:00 0 [vdso] 00d69000-00d72000 r-xp 00000000 fd:00 43811031 /lib/libcrypt-2.5.so 00d72000-00d73000 r--p 00008000 fd:00 43811031 /lib/libcrypt-2.5.so 00d73000-00d74000 rw-p 00009000 fd:00 43811031 /lib/libcrypt-2.5.so 00d74000-00d9b000 rw-p 00d74000 00:00 0 00d9c000-00d9f000 r-xp 00000000 fd:00 81363253 /usr/local/freeswitch/mod/mod_logfile.so 00d9f000-00da0000 rw-p 00002000 fd:00 81363253 /usr/local/freeswitch/mod/mod_logfile.so 00df9000-00dfb000 r-xp 00000000 fd:00 43811004 /lib/libkeyutils-1.2.so 00dfb000-00dfc000 rw-p 00001000 fd:00 43811004 /lib/libkeyutils-1.2.so 00dfc000-00ea3000 r-xp 00000000 fd:00 81330449 /usr/local/freeswitch/lib/libjs.so.1.0.6 00ea3000-00ea8000 rw-p 000a7000 fd:00 81330449 /usr/local/freeswitch/lib/libjs.so.1.0.6 00ec1000-00ec4000 r-xp 00000000 fd:00 81363216 /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so 00ec4000-00ec5000 rw-p 00002000 fd:00 81363216 /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so 00f06000-00f12000 r-xp 00000000 fd:00 81363240 /usr/local/freeswitch/mod/mod_enum.so 00f12000-00f13000 rw-p 0000c000 fd:00 81363240 /usr/local/freeswitch/mod/mod_enum.so 00f13000-01089000 r-xp 00000000 fd:00 81363261 /usr/local/freeswitch/mod/mod_sofia.so 01089000-01096000 rw-p 00175000 fd:00 81363261 /usr/local/freeswitch/mod/mod_sofia.so 02ab5000-02b31000 r-xp 00000000 fd:00 80589691 /usr/lib/libSDL-1.2.so.0.7.3 02b31000-02b33000 rw-p 0007c000 fd:00 80589691 /usr/lib/libSDL-1.2.so.0.7.3 02b33000-02b5e000 rw-p 02b33000 00:00 0 03474000-03aa4000 r-xp 00000000 fd:00 81330468 /usr/local/freeswitch/lib/libh323_linux_x86_.so.1.22.0 03aa4000-03aec000 rw-p 00630000 fd:00 81330468 /usr/local/freeswitch/lib/libh323_linux_x86_.so.1.22.0 03aec000-03aef000 rw-p 03aec000 00:00 0 04aad000-04b87000 r-xp 00000000 fd:00 43810856 /lib/libasound.so.2.0.0 04b87000-04b8c000 rw-p 000d9000 fd:00 43810856 /lib/libasound.so.2.0.0 0675e000-06993000 r-xp 00000000 fd:00 81330450 /usr/local/freeswitch/lib/libpt.so.2.9-beta0 06993000-069ba000 rw-p 00234000 fd:00 81330450 /usr/local/freeswitch/lib/libpt.so.2.9-beta0 069ba000-069bd000 rw-p 069ba000 00:00 0 08048000-0804c000 r-xp 00000000 fd:00 81363034 /usr/local/freeswitch/bin/freeswitch 0804c000-0804d000 rw-p 00003000 fd:00 81363034 /usr/local/freeswitch/bin/freeswitch 08144000-0831e000 r-xp 00000000 fd:00 80589813 /usr/lib/libpt_linux_x86_r.so.1.10.1 0831e000-08334000 rw-p 001d9000 fd:00 80589813 /usr/lib/libpt_linux_x86_r.so.1.10.1 08334000-08339000 rw-p 08334000 00:00 0 08f47000-090a5000 rw-p 08f47000 00:00 0 [heap] b6ef8000-b6ef9000 ---p b6ef8000 00:00 0 b6ef9000-b6f34000 rw-p b6ef9000 00:00 0 b6f34000-b6f35000 ---p b6f34000 00:00 0 b6f35000-b6f70000 rw-p b6f35000 00:00 0 b6f70000-b6f71000 ---p b6f70000 00:00 0 b6f71000-b6fac000 rw-p b6f71000 00:00 0 b6fac000-b6fad000 ---p b6fac000 00:00 0 b6fad000-b6fe8000 rw-p b6fad000 00:00 0 b6fe8000-b6fe9000 ---p b6fe8000 00:00 0 b6fe9000-b7402000 rw-p b6fe9000 00:00 0 b7402000-b7403000 ---p b7402000 00:00 0 b7403000-b7470000 rw-p b7403000 00:00 0 b7470000-b7471000 ---p b7470000 00:00 0 b7471000-b74de000 rw-p b7471000 00:00 0 b74de000-b74df000 ---p b74de000 00:00 0 b74df000-b751a000 rw-p b74df000 00:00 0 b751a000-b751b00Aborted (core dumped) ________________________________ From: Anthony Minessale To: FreeSWITCH Users Help Sent: Sat, August 21, 2010 1:54:27 AM Subject: Re: [Freeswitch-users] Mod_h323 fail to load probably missing ldconfig or LD_LIBRARY_PATH On Fri, Aug 20, 2010 at 4:10 PM, Sadjad Seyed-Ahmadian wrote: > I got the pint files are in /usr/local/lib and I make a link in > /isr/local/freeswitch/lib to them, but when I try to load mod_h323 > freeswitch crashed!! > > ________________________________ > From: Sadjad Seyed-Ahmadian > To: FreeSWITCH Users Help > Sent: Sat, August 21, 2010 1:13:19 AM > Subject: Re: [Freeswitch-users] Mod_h323 fail to load > > No I cannot find it! > ________________________________ > From: Jean-Yves F. Barbier <12ukwn at gmail.com> > To: freeswitch-users at lists.freeswitch.org > Sent: Sat, August 21, 2010 12:57:22 AM > Subject: Re: [Freeswitch-users] Mod_h323 fail to load > > Le Fri, 20 Aug 2010 13:14:32 -0700 (PDT), > Sadjad Seyed-Ahmadian a ?crit : > >> I try to install mod_h323, I compile it as it comes in wiki >> (http://wiki.freeswitch.org/wiki/Mod_h323) >> but when I try to load mod_h323 in fs_cli I faces this error: > > Can you find a 'mod_h323.so/.la' in /usr/local/freeswitch/lib? > > -- > The best way to make a fire with two sticks is to make sure one of them > is a match. > -- Will Rogers > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/93fd5a26/attachment-0001.html From tculjaga at gmail.com Sat Aug 21 02:11:50 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 21 Aug 2010 11:11:50 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <305309.80502.qm@web53403.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> Message-ID: what version of ptlib are u using ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/7131adfa/attachment.html From david.ponzone at ipeva.fr Sat Aug 21 02:22:45 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Sat, 21 Aug 2010 11:22:45 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <1282321281.16719.163.camel@luna.tc.commsmundi.com> References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <1282316031103-5444786.post@n2.nabble.com> <20100820175224.5d472a21@anubis.defcon1> <1282321281.16719.163.camel@luna.tc.commsmundi.com> Message-ID: <3900B205-9416-4888-8352-87DED5A35465@ipeva.fr> I already removed the stress tests part from the table. I will try to add other information that was given here. For the disclaimer part, as english is not my native language, I would appreciate if someone could recommend one with the right content, thus avoiding our dear FreeSWITCH devs to go to jail in the future :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 20/08/2010 ? 18:21, Fran?ois Delawarde a ?crit : > Stress test results are only useful to make the headlines or for PhD > students, those tests should only be made by experts (unbiased if > possible) that and results should be shown in a special "cool-pure-raw > performance testing" page. > > For the rest of us, real life scenarios with a small description of > the > modules involved would be useful to help with server dimensioning. If > people contribute, with enough data it could give a pretty nice idea > of > what to expect. > > Of course those results can vary a lot so a huge disclaimer must be > there to avoid having special people (mainly US citizens) suing the > wiki > owner. ;-) > > Fran?ois. > > > On Fri, 2010-08-20 at 11:01 -0500, Anthony Minessale wrote: >> Its an ok thing as long as it comes with a disclaimer that "results >> may vary" and "you have to know what you are doing" >> And those who work on it need to volunteer to help with the emails it >> may generate asking why they can't match the results =D >> >> >> On Fri, Aug 20, 2010 at 10:52 AM, Jean-Yves F. Barbier <12ukwn at gmail.com >> > wrote: >>> Le Fri, 20 Aug 2010 10:10:32 -0500, >>> Anthony Minessale a ?crit : >>> >>>> The problem is that performance testing is very fluid and >>>> relevant to >>>> each machine and OS and test performed and you can't get concrete >>>> expectations without some advanced skills in deploying FS. >>> >>> Well, it could have a real interest by adding the tweaked >>> parameters, >>> may be there are some, yet unexplored, that would raise the >>> response. >>> >>> -- >>> Why don't elephants eat penguins ? >>> Because they can't get the wrappers off ... >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/80385160/attachment.html From ssa1357 at yahoo.com Sat Aug 21 02:39:16 2010 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Sat, 21 Aug 2010 02:39:16 -0700 (PDT) Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> Message-ID: <585587.29855.qm@web53407.mail.re2.yahoo.com> I use latest subversion. ________________________________ From: Tihomir Culjaga To: FreeSWITCH Users Help Sent: Sat, August 21, 2010 1:41:50 PM Subject: Re: [Freeswitch-users] Mod_h323 fail to load what version of ptlib are u using ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/a9ffb263/attachment.html From david.ponzone at ipeva.fr Sat Aug 21 03:13:38 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Sat, 21 Aug 2010 12:13:38 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <3900B205-9416-4888-8352-87DED5A35465@ipeva.fr> References: <20100820125543.2c462087@anubis.defcon1> <42C8541B-6A28-4633-8FD8-BEC8EAAB044D@ipeva.fr> <1282316031103-5444786.post@n2.nabble.com> <20100820175224.5d472a21@anubis.defcon1> <1282321281.16719.163.camel@luna.tc.commsmundi.com> <3900B205-9416-4888-8352-87DED5A35465@ipeva.fr> Message-ID: <82391E30-A4BC-4C4B-9D60-550DC3906E99@ipeva.fr> I created a dedicated page for the results, with a link from the main page. If that's not ok or if something needs to be changed/reformulated, please tell me. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 21/08/2010 ? 11:22, David Ponzone a ?crit : > I already removed the stress tests part from the table. > I will try to add other information that was given here. > > For the disclaimer part, as english is not my native language, I > would appreciate if someone could recommend one with the right > content, thus avoiding our dear FreeSWITCH devs to go to jail in the > future :) > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, > d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur. > > > > > Le 20/08/2010 ? 18:21, Fran?ois Delawarde a ?crit : > >> Stress test results are only useful to make the headlines or for PhD >> students, those tests should only be made by experts (unbiased if >> possible) that and results should be shown in a special "cool-pure- >> raw >> performance testing" page. >> >> For the rest of us, real life scenarios with a small description of >> the >> modules involved would be useful to help with server dimensioning. If >> people contribute, with enough data it could give a pretty nice >> idea of >> what to expect. >> >> Of course those results can vary a lot so a huge disclaimer must be >> there to avoid having special people (mainly US citizens) suing the >> wiki >> owner. ;-) >> >> Fran?ois. >> >> >> On Fri, 2010-08-20 at 11:01 -0500, Anthony Minessale wrote: >>> Its an ok thing as long as it comes with a disclaimer that "results >>> may vary" and "you have to know what you are doing" >>> And those who work on it need to volunteer to help with the emails >>> it >>> may generate asking why they can't match the results =D >>> >>> >>> On Fri, Aug 20, 2010 at 10:52 AM, Jean-Yves F. Barbier <12ukwn at gmail.com >>> > wrote: >>>> Le Fri, 20 Aug 2010 10:10:32 -0500, >>>> Anthony Minessale a ?crit : >>>> >>>>> The problem is that performance testing is very fluid and >>>>> relevant to >>>>> each machine and OS and test performed and you can't get concrete >>>>> expectations without some advanced skills in deploying FS. >>>> >>>> Well, it could have a real interest by adding the tweaked >>>> parameters, >>>> may be there are some, yet unexplored, that would raise the >>>> response. >>>> >>>> -- >>>> Why don't elephants eat penguins ? >>>> Because they can't get the wrappers off ... >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/56455420/attachment-0001.html From tculjaga at gmail.com Sat Aug 21 04:00:40 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 21 Aug 2010 13:00:40 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <585587.29855.qm@web53407.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com> Message-ID: On Sat, Aug 21, 2010 at 11:39 AM, Sadjad Seyed-Ahmadian wrote: > I use latest subversion. > > ya, let me pack you the version of ptlib im using.... its definitly this ... T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/bb9f538a/attachment.html From tculjaga at gmail.com Sat Aug 21 04:02:54 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 21 Aug 2010 13:02:54 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com> Message-ID: On Sat, Aug 21, 2010 at 1:00 PM, Tihomir Culjaga wrote: > > > On Sat, Aug 21, 2010 at 11:39 AM, Sadjad Seyed-Ahmadian > wrote: > >> I use latest subversion. >> >> > ya, let me pack you the version of ptlib im using.... its definitly this > ... > > T. > also, check if you are using this version libpt.so -> libpt.so.2.9-beta0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/85f9765f/attachment.html From peter.olsson at visionutveckling.se Sat Aug 21 04:05:36 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 21 Aug 2010 13:05:36 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <585587.29855.qm@web53407.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> , <585587.29855.qm@web53407.mail.re2.yahoo.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC058151@cooper> Try changing to versions proposed here: http://www.gnugk.org/compiling-gnugk.html PTLib 2.8.1 and H323Plus CVS as of 2010-05-25, I use these versions and it works perfectly. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Sadjad Seyed-Ahmadian [ssa1357 at yahoo.com] Skickat: den 21 augusti 2010 11:39 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Mod_h323 fail to load I use latest subversion. ________________________________ From: Tihomir Culjaga To: FreeSWITCH Users Help Sent: Sat, August 21, 2010 1:41:50 PM Subject: Re: [Freeswitch-users] Mod_h323 fail to load what version of ptlib are u using ? !DSPAM:4c6fa06d32932165174240! From vhatz at kinetix.gr Sat Aug 21 05:41:13 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Sat, 21 Aug 2010 15:41:13 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> Message-ID: <4C6FC969.6020801@kinetix.gr> Michael Collins wrote: > > > You wrote in your reply to Woody: "Do you hear yourself btw?" I > can ask > you the same question: Do you hear yourself? You act as if some > subjects > are taboo in this list. > > > Taboos aren't really the issue. Spending valuable time and energy in > unfruitful debates is an issue. People at times have been accused of various different things in this list just because they dared to ask something! And all you have to say is that this is an unfruitful debate? For me the whole debate is VERY fruitful if the next user who dares to mention the dreaded word "performance" is spared the irony and the bashing. > > If you decide to reply again, please limit the length of your post. > > > > > Now it is the size of my posts that troubles you? I've seen longer > posts > in the list and you didn't complain. > > Brevity is the essence of communication - I highly recommend it. Then, please recommend it to all cases and not just in my case. Tony's reply was not so brief either, so, please go ahead and say the same to him. That is, only if you care to appear impartial. Me on the other hand, I'd prefer an actual reply to my arguments, instead of dismissing the entire discussion for the sake of brevity. Up to now I have seen more evasion tactics than arguments against the points I've made. > > -MC > > P.S. - Just a clarification: Using SIPp is a "fake" test to do > 200/400/600/xxx calls per second. Getting thousands of actual humans > making actual calls is a "real" test. In other words, since SIPp can > only do so much its test results are of limited value in the real world. Really, is this what a "fake" test is? Because I thought that it was a test that never actually happened. English is not my native language, but at least I know that much. What you wrote in your P.S. is so wrong in so many ways. For the "sake of brevity" I will only say that it is a bad attempt to cover Tony's rude reply to a list user and I'll leave it at that for the moment. Otherwise we will have to get in a discussion of what a test is, what is a "real" as opposed to a "fake" test and whether sipp or test tools in general have limited value in the real world. I don't have a problem with long discussions, but is seems they are not favored in this context so I'll pass for the moment, until the circumstances require it. > > > > Perhaps you should add a size restriction directive in the mailing > list > policy, along with a "no performance questions allowed" directive. > That > should clear things up significantly. > > Would you be willing just to let the subject go? I think everyone is > tired of this thread. Tired? Now you can speak for "everyone"? I know quite a few people who are actually reading this thread with lots of interest. Sure, I'm willing to let go... Are you? Have a great weekend, Vlasis. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/c73264c7/attachment.html From gmaruzz at celliax.org Sat Aug 21 06:08:43 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 21 Aug 2010 15:08:43 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6FC969.6020801@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> Message-ID: Vlasis, please calm down and try to understand the context you're interacting with. This is not a matter of civil rights or politically correctness or different sensibilities or respect for the freedom of speech or whatever thing of this kind. This is a matter about a consistent, practical and every time asserted displease from the core team developers (and most of the not-so-core team developers) toward stress tests and performance measurements. It has been stated many many times, and the reason is: because fake test (sipp) does not gives you results that reflects in real world case, because sipp is black magic, because there are many variable involved (OS, hardware, kind of load, etc), because this is likely to attract undue interest on naked figures [we can do 1 million cps!!!!!], because to get it right involves a tremendous lot of experience, because the only useful test for your usage is to put in production a reasonable dimensioned machine and then see of it goes, because we read all the message in the list and particularly on performance themes we are pushed to intervene before too many wrong things are asserted, we don't like threads debating about performance testing. But, because we're human and nice, and helpful and caring, on the side of the bashing we also offer pearls of wisdom that can be used for the purpose, instead of simply bashing or ignore the poster. So, take it as a mysterious ways of the Zen Masters to visit black art on performance test posters. -giovanni On Sat, Aug 21, 2010 at 2:41 PM, Vlasis Hatzistavrou (KTI) wrote: > > Michael Collins wrote: > > >> >> You wrote in your reply to Woody: "Do you hear yourself btw?" I can ask >> you the same question: Do you hear yourself? You act as if some subjects >> are taboo in this list. > > Taboos aren't really the issue. Spending valuable time and energy in > unfruitful debates is an issue. > > People at times have been accused of various different things in this list > just because they dared to ask something! And all you have to say is that > this is an unfruitful debate? > > For me the whole debate is VERY fruitful if the next user who dares to > mention the dreaded word "performance" is spared the irony and the bashing. > >> > If you decide to reply again, please limit the length of your post. >> > >> > >> Now it is the size of my posts that troubles you? I've seen longer posts >> in the list and you didn't complain. > > Brevity is the essence of communication - I highly recommend it. > > Then, please recommend it to all cases and not just in my case. > > Tony's reply was not so brief either, so, please go ahead and say the same > to him. That is, only if you care to appear impartial. > > Me on the other hand, I'd prefer an actual reply to my arguments, instead of > dismissing the entire discussion for the sake of brevity. Up to now I have > seen more evasion tactics than arguments against the points I've made. > > > -MC > > P.S. - Just a clarification: Using SIPp is a "fake" test to do > 200/400/600/xxx calls per second. Getting thousands of actual humans making > actual calls is a "real" test. In other words, since SIPp can only do so > much its test results are of limited value in the real world. > > Really, is this what a "fake" test is? Because I thought that it was a test > that never actually happened. English is not my native language, but at > least I know that much. > > What you wrote in your P.S. is so wrong in so many ways. For the "sake of > brevity" I will only say that it is a bad attempt to cover Tony's rude reply > to a list user and I'll leave it at that for the moment. > > Otherwise we will have to get in a discussion of what a test is, what is a > "real" as opposed to a "fake" test and whether sipp or test tools in general > have limited value in the real world. I don't have a problem with long > discussions, but is seems they are not favored in this context so I'll pass > for the moment, until the circumstances require it. > > >> >> Perhaps you should add a size restriction directive in the mailing list >> policy, along with a "no performance questions allowed" directive. That >> should clear things up significantly. > > Would you be willing just to let the subject go? I think everyone is tired > of this thread. > > Tired? Now you can speak for "everyone"? I know quite a few people who are > actually reading this thread with lots of interest. > > Sure, I'm willing to let go... Are you? > > Have a great weekend, > Vlasis. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From 12ukwn at gmail.com Sat Aug 21 06:40:07 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 21 Aug 2010 15:40:07 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6FC969.6020801@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> Message-ID: <20100821154007.7a45cd51@anubis.defcon1> Le Sat, 21 Aug 2010 15:41:13 +0300, "Vlasis Hatzistavrou (KTI)" a ?crit : PLS Vlasis, calm down, I think eveyone is loosing temper riht now. Let's say you took a test option, whether it is a good one or not could be debated for a long time, but if YOU think it is the right way to go for YOUR needs then go with it! So my advice would be to eventually adjust your method only if you think it would help in your particular case of operation; and stick to it from now in order to always use the same test environment, to have a constant relevance of results and thus constant comparisons possibilities too in time :) (nooo, not in the head!!!:) JY -- From phone.bytes at gmail.com Sat Aug 21 07:53:47 2010 From: phone.bytes at gmail.com (D Al) Date: Sat, 21 Aug 2010 08:53:47 -0600 Subject: [Freeswitch-users] Webapi Examples location In-Reply-To: References: <4C6EE9A7.4050108@gmail.com> Message-ID: Now that we have the "list blocking" issue resolved....can anyone help with the location of the source for these included examples? Thanks On Fri, Aug 20, 2010 at 5:59 PM, Michael Collins wrote: > > > On Fri, Aug 20, 2010 at 3:49 PM, jesse zhao wrote: > >> why my message post to the mail list gets blocked? testing by reply . >> > Not blocked, just moderated until you join the mailing list. Once you're > verified you should be good to go. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/2c8161fa/attachment.html From vhatz at kinetix.gr Sat Aug 21 07:54:48 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Sat, 21 Aug 2010 17:54:48 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> Message-ID: <4C6FE8B8.4030809@kinetix.gr> Giovanni Maruzzelli wrote: > Vlasis, > > please calm down and try to understand the context you're interacting with. > > This is not a matter of civil rights or politically correctness or > different sensibilities or respect for the freedom of speech or > whatever thing of this kind. > > This is a matter about a consistent, practical and every time asserted > displease from the core team developers (and most of the not-so-core > team developers) toward stress tests and performance measurements. > > It has been stated many many times, and the reason is: because fake > test (sipp) does not gives you results that reflects in real world > case, because sipp is black magic, because there are many variable > involved (OS, hardware, kind of load, etc), because this is likely to > attract undue interest on naked figures [we can do 1 million > cps!!!!!], because to get it right involves a tremendous lot of > experience, because the only useful test for your usage is to put in > production a reasonable dimensioned machine and then see of it goes, > because we read all the message in the list and particularly on > performance themes we are pushed to intervene before too many wrong > things are asserted, we don't like threads debating about performance > testing. > > But, because we're human and nice, and helpful and caring, on the side > of the bashing we also offer pearls of wisdom that can be used for the > purpose, instead of simply bashing or ignore the poster. > > So, take it as a mysterious ways of the Zen Masters to visit black art > on performance test posters. > > -giovanni Hello Giovanni, First of all, what makes you think I am not calm? :):) No matter what the context is, most of us expect basic courtesy from our peers in all our interactions. When such courtesy becomes rare, some people may react. It is not a matter of civil rights. There is a story behind this. I've been treated badly when I raised questions in the past. We've argued with Tony about this in the past. He said pretty much the same thing to me as to Woody, that I was making money on his expense. A colleague of his even suggested to us that VoIP testing is tricky giving us the impression that it should probably not be done by us because we were idiots. People in this list are still treated badly for asking the simplest of questions without being insulting or abusive in the slightest degree. This is just crazy! So, do you think that performance questions should not be brought up in this list? Or that they should be brought up only by some sort of gurus and not by any guy who just installed FS and tries to learn it? Don't you wonder why the devs feel that FS performance is so touchy a subject? Also, watch out because your post was probably longer than what the list maintainers prefer... They may tell you off. If they don't, then they just have double standards. Rgds, Vlasis. From vhatz at kinetix.gr Sat Aug 21 07:58:27 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Sat, 21 Aug 2010 17:58:27 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <20100821154007.7a45cd51@anubis.defcon1> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> Message-ID: <4C6FE993.3090102@kinetix.gr> Hello Jean-Yves, Thanks for the effort to calm me down, but I am calm and fine. :) Just for the record I wasn't the one doing the tests, I was just defending the guy who asked the original question because I've been in his place and felt for him. ;) Rgds, Vlasis. Jean-Yves F. Barbier wrote: > Le Sat, 21 Aug 2010 15:41:13 +0300, > "Vlasis Hatzistavrou (KTI)" a ?crit : > > PLS Vlasis, calm down, I think eveyone is loosing temper riht now. > > Let's say you took a test option, whether it is a good one or not could be > debated for a long time, but if YOU think it is the right way to go for > YOUR needs then go with it! > So my advice would be to eventually adjust your method only if you think > it would help in your particular case of operation; and stick to it from > now in order to always use the same test environment, to have a constant > relevance of results and thus constant comparisons possibilities too in > time :) > > (nooo, not in the head!!!:) > > JY > From 12ukwn at gmail.com Sat Aug 21 08:41:56 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 21 Aug 2010 17:41:56 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6FE993.3090102@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> Message-ID: <20100821174156.13b667ae@anubis.defcon1> Le Sat, 21 Aug 2010 17:58:27 +0300, "Vlasis Hatzistavrou (KTI)" a ?crit : > Hello Jean-Yves, > > Thanks for the effort to calm me down, but I am calm and fine. :) That I understand, Vlasis; just a matter of speaking ;) > Just for the record I wasn't the one doing the tests, I was just > defending the guy who asked the original question because I've been in > his place and felt for him. My purpose was just to extinguish this sterile discussion. Each part have real good arguments to counter the other part, but as nobody wants to leave its part of the cake I'm trying to guess a solution that would be the best for everybody tout le monde. This is why I think even a "bad" test could be good BUT ONLY if test conditions are even every time it is performed. The *real* question behind all of this is: why always test and compare? Wouldn't it be because large and greedy companies (also ear governments, especially corrupted ones by these same large companies) put that in our mind to sell every day more (and "better") to people that don't need it... Think about that. When I'm working for other I try to make it for the best and do it like if it was for myself - so I don't ask less from open-source devs that make tremendous programs, which if fortunately (almost) the case. So I can make a proposition to the dev team to _*definitely*_ fire down this kinda non-discussion: * Make a realistic serie of recursive test (who's more skilled than you to build that?:), * Always use the same machine with the same environment (possibly a "weak" one, such as a simple P4 2.0GHz w/ 2GB RAM: think about third world countries for which this kind of project is very important), * Publish the test serie results each time it is needed. (nooo, not the head AND the kidneys!!!) JY PS: I didn't choosed the fortune signature, it is a cron:) -- There appears to be irrefutable evidence that the mere fact of overcrowding induces violence. -- Harvey Wheeler From brian at freeswitch.org Sat Aug 21 09:20:49 2010 From: brian at freeswitch.org (Brian West) Date: Sat, 21 Aug 2010 11:20:49 -0500 Subject: [Freeswitch-users] simple echo failed In-Reply-To: References: Message-ID: <146EFC92-F165-4931-99C1-DAAEE2267233@freeswitch.org> I don't recall any examples or documentation showing that.... How about slowing down a little and paying closer attention to things instead of rushing thru them and missing the little things... We are all guilty of this from time to time. /b On Aug 21, 2010, at 2:37 AM, jesse wrote: > freaking stupid, it because the extra space between & e. > why it is designed like this? so error prone and misleading > > -jesse From mnhassan at usa.net Sat Aug 21 09:23:16 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sat, 21 Aug 2010 22:23:16 +0600 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <20100821174156.13b667ae@anubis.defcon1> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <20100821174156.13b667ae@anubis.defcon1> Message-ID: One thing that you are totaly "ignoring" Vlasis, is that the performance tests are irritating to the core-devs because FreeSwitch's CPS is already so high, that their effort / time are better used at improving / adding other things. Like the new "concurrent call recovery" feature, which is just simply crazy, if you ask me. Just think about it, at 200 CPS, for 100% ASR (IVR) and 1 min ALOC / ACD, you're talking about 12,000 concurrent sessions. Or in a "real-life" call routing scenario, at 50% ASR and 5 mins ALOC, it is equivalent to 30,000 concurrent sessions, or 1,000 E1s! And all that, using regular, run-of-the-mill, processor, motherboard, etc. The server Woody mentions would probably cost around $ 6,000 - 7,000/- from HP or IBM. Who can beat that? And, to top that off, Woody was comparing to 600 CPS that he got on v1.04 (36,000 or 90,000 conc. sess.)! The core-devs are very passionate about their work. And, they have created a beast! So, get the beast to work "in real life", and then complain about problems. I've seen them instantly jump into helping with the situation. Bug reports are fixed within hours. Even the "latest GIT" is claimed to be the most stable versions! Not many software projects (open source or not) can boast that, if at all. Regards HASSAN On Sat, Aug 21, 2010 at 21:41, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote: > Le Sat, 21 Aug 2010 17:58:27 +0300, > "Vlasis Hatzistavrou (KTI)" a ?crit : > > > Hello Jean-Yves, > > > > Thanks for the effort to calm me down, but I am calm and fine. :) > > That I understand, Vlasis; just a matter of speaking ;) > > > Just for the record I wasn't the one doing the tests, I was just > > defending the guy who asked the original question because I've been in > > his place and felt for him. > > My purpose was just to extinguish this sterile discussion. > > Each part have real good arguments to counter the other part, but as > nobody wants to leave its part of the cake I'm trying to guess a > solution that would be the best for everybody tout le monde. > > This is why I think even a "bad" test could be good BUT ONLY if test > conditions are even every time it is performed. > > The *real* question behind all of this is: why always test and compare? > Wouldn't it be because large and greedy companies (also ear governments, > especially corrupted ones by these same large companies) put that in our > mind to sell every day more (and "better") to people that don't need it... > Think about that. > > When I'm working for other I try to make it for the best and do it like > if it was for myself - so I don't ask less from open-source devs that > make tremendous programs, which if fortunately (almost) the case. > > So I can make a proposition to the dev team to _*definitely*_ fire down > this > kinda non-discussion: > * Make a realistic serie of recursive test (who's more skilled than > you to build that?:), > * Always use the same machine with the same environment (possibly a "weak" > one, such as a simple P4 2.0GHz w/ 2GB RAM: think about third world > countries for which this kind of project is very important), > * Publish the test serie results each time it is needed. > > (nooo, not the head AND the kidneys!!!) > > JY > > PS: I didn't choosed the fortune signature, it is a cron:) > -- > There appears to be irrefutable evidence that the mere fact of overcrowding > induces violence. > -- Harvey Wheeler > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/22e1b282/attachment-0001.html From vhatz at kinetix.gr Sat Aug 21 09:56:31 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Sat, 21 Aug 2010 19:56:31 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <20100821174156.13b667ae@anubis.defcon1> Message-ID: <4C70053F.5000701@kinetix.gr> Nyamul Hassan wrote: > One thing that you are totaly "ignoring" Vlasis, is that the > performance tests are irritating to the core-devs because FreeSwitch's > CPS is already so high, that their effort / time are better used at > improving / adding other things. Like the new "concurrent call > recovery" feature, which is just simply crazy, if you ask me. > Hassan, (no irony intended, please believe me) but one thing you are totally ignoring too is that we were not debating about the validity of the tests. We were talking about attitude towards posts with questions on certain subjects. If users are still talking about performance this is because they are interested in it and are drawn to FS because of it. It's only natural to talk about it. And users of all software are talking about performance. > Just think about it, at 200 CPS, for 100% ASR (IVR) and 1 min ALOC / > ACD, you're talking about 12,000 concurrent sessions. Or in a > "real-life" call routing scenario, at 50% ASR and 5 mins ALOC, it is > equivalent to 30,000 concurrent sessions, or 1,000 E1s! > > And all that, using regular, run-of-the-mill, processor, motherboard, > etc. The server Woody mentions would probably cost around $ 6,000 - > 7,000/- from HP or IBM. Who can beat that? And, to top that off, > Woody was comparing to 600 CPS that he got on v1.04 (36,000 or 90,000 > conc. sess.)! > > The core-devs are very passionate about their work. And, they have > created a beast! So, get the beast to work "in real life", and then > complain about problems. Hassan, you didn't understand Woody's post neither mine. Did you read me or Woody complaining about problems with FS? I don't think so. Woody asked a very simple question. He didn't complain. Yet he was bashed. There was no complain in his message to the list, unless he sent another one which I missed completely. On the other hand, I have 2 questions regarding your comments: 1) Where exactly did you read 600 cps? Perhaps I missed a post but I only read 1 post from Woody with a scenario of 190 to 200 cps. Still a very high score, but this is not the point. 2) Sorry if it sounds ironic (again, no irony intended) but why are do you suggest that we complain about problems only after we use FS in real life? Because many of us do. On the other hand, why are lab tests that much frowned upon? Some people know how to test and reproduce production scenarios in a lab. Really. > I've seen them instantly jump into helping with the situation. Bug > reports are fixed within hours. Even the "latest GIT" is claimed to > be the most stable versions! Not many software projects (open source > or not) can boast that, if at all. Hassan, no-one questioned the hard work put in by the developers. We were having a different discussion here. As a sidenote, when the latest git/svn which, by nature, is work under development, is more stable than the latest stable release then yes, not many projects can boast that. On the contrary, most projects avoid this situation, because the users will have to put in production source code which is in a "fluid" state. Again, this is just a sidenote, (a comment on a comment) no intention to start a new flame here. :) Rgds, Vlasis. From david.ponzone at ipeva.fr Sat Aug 21 09:55:50 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Sat, 21 Aug 2010 18:55:50 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6FE993.3090102@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> Message-ID: <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> Guys, As it seems that this debate is going to last forever without an acceptable conclusion, let's try to refocus on something positive. What is interesting for most of us are the real-world performances. As I said previously, the performance reports page is ready: http://wiki.freeswitch.org/wiki/Real-world_results It would be really nice if the people who have interesting figures could start filling it, as it would probably take them 5 to 10 minutes! David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 21/08/2010 ? 16:58, Vlasis Hatzistavrou (KTI) a ?crit : > Hello Jean-Yves, > > Thanks for the effort to calm me down, but I am calm and fine. :) > > Just for the record I wasn't the one doing the tests, I was just > defending the guy who asked the original question because I've been in > his place and felt for him. > > ;) > > Rgds, > Vlasis. > > Jean-Yves F. Barbier wrote: >> Le Sat, 21 Aug 2010 15:41:13 +0300, >> "Vlasis Hatzistavrou (KTI)" a ?crit : >> >> PLS Vlasis, calm down, I think eveyone is loosing temper riht now. >> >> Let's say you took a test option, whether it is a good one or not >> could be >> debated for a long time, but if YOU think it is the right way to go >> for >> YOUR needs then go with it! >> So my advice would be to eventually adjust your method only if you >> think >> it would help in your particular case of operation; and stick to it >> from >> now in order to always use the same test environment, to have a >> constant >> relevance of results and thus constant comparisons possibilities >> too in >> time :) >> >> (nooo, not in the head!!!:) >> >> JY >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/0783689c/attachment.html From mnhassan at usa.net Sat Aug 21 10:11:01 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sat, 21 Aug 2010 23:11:01 +0600 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> Message-ID: Vlasis, Woody's 2nd email talked about 600 CPS on v1.04. And, perhaps you can share your usage scenario, and how much pressure you could put on FreeSwitch before it "crashed"? No hard feelings, and request to you is to also not take any from Tony's comments. The core team, as I've seen, is devoted to make FS better for "production use". Point out any flaw in "production", and you'll see it first hand. Will like to see your implementation case, btw. Regards HASSAN On Sat, Aug 21, 2010 at 22:55, David Ponzone wrote: > Guys, > > As it seems that this debate is going to last forever without an acceptable > conclusion, let's try to refocus on something positive. > > What is interesting for most of us are the real-world performances. > As I said previously, the performance reports page is ready: > http://wiki.freeswitch.org/wiki/Real-world_results > > It would be really nice if the people who have interesting figures could > start filling it, as it would probably take them 5 to 10 minutes! > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 21/08/2010 ? 16:58, Vlasis Hatzistavrou (KTI) a ?crit : > > Hello Jean-Yves, > > Thanks for the effort to calm me down, but I am calm and fine. :) > > Just for the record I wasn't the one doing the tests, I was just > defending the guy who asked the original question because I've been in > his place and felt for him. > > ;) > > Rgds, > Vlasis. > > Jean-Yves F. Barbier wrote: > > Le Sat, 21 Aug 2010 15:41:13 +0300, > > "Vlasis Hatzistavrou (KTI)" a ?crit : > > > PLS Vlasis, calm down, I think eveyone is loosing temper riht now. > > > Let's say you took a test option, whether it is a good one or not could be > > debated for a long time, but if YOU think it is the right way to go for > > YOUR needs then go with it! > > So my advice would be to eventually adjust your method only if you think > > it would help in your particular case of operation; and stick to it from > > now in order to always use the same test environment, to have a constant > > relevance of results and thus constant comparisons possibilities too in > > time :) > > > (nooo, not in the head!!!:) > > > JY > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/a0d07263/attachment-0001.html From mnhassan at usa.net Sat Aug 21 10:14:32 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sat, 21 Aug 2010 23:14:32 +0600 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> Message-ID: Aah... the last line means, just the stats on peak performance, modules in action, CPU / RAM / Disk IO / Network IO. Regards HASSAN On Sat, Aug 21, 2010 at 23:11, Nyamul Hassan wrote: > Vlasis, Woody's 2nd email talked about 600 CPS on v1.04. And, perhaps you > can share your usage scenario, and how much pressure you could put on > FreeSwitch before it "crashed"? > > No hard feelings, and request to you is to also not take any from Tony's > comments. The core team, as I've seen, is devoted to make FS better for > "production use". Point out any flaw in "production", and you'll see it > first hand. > > Will like to see your implementation case, btw. > > Regards > HASSAN > > > > On Sat, Aug 21, 2010 at 22:55, David Ponzone wrote: > >> Guys, >> >> As it seems that this debate is going to last forever without an >> acceptable conclusion, let's try to refocus on something positive. >> >> What is interesting for most of us are the real-world performances. >> As I said previously, the performance reports page is ready: >> http://wiki.freeswitch.org/wiki/Real-world_results >> >> It would be really nice if the people who have interesting figures could >> start filling it, as it would probably take them 5 to 10 minutes! >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 21/08/2010 ? 16:58, Vlasis Hatzistavrou (KTI) a ?crit : >> >> Hello Jean-Yves, >> >> Thanks for the effort to calm me down, but I am calm and fine. :) >> >> Just for the record I wasn't the one doing the tests, I was just >> defending the guy who asked the original question because I've been in >> his place and felt for him. >> >> ;) >> >> Rgds, >> Vlasis. >> >> Jean-Yves F. Barbier wrote: >> >> Le Sat, 21 Aug 2010 15:41:13 +0300, >> >> "Vlasis Hatzistavrou (KTI)" a ?crit : >> >> >> PLS Vlasis, calm down, I think eveyone is loosing temper riht now. >> >> >> Let's say you took a test option, whether it is a good one or not could >> be >> >> debated for a long time, but if YOU think it is the right way to go for >> >> YOUR needs then go with it! >> >> So my advice would be to eventually adjust your method only if you think >> >> it would help in your particular case of operation; and stick to it from >> >> now in order to always use the same test environment, to have a constant >> >> relevance of results and thus constant comparisons possibilities too in >> >> time :) >> >> >> (nooo, not in the head!!!:) >> >> >> JY >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/88a2dbff/attachment.html From curriegrad2004 at gmail.com Fri Aug 20 22:14:41 2010 From: curriegrad2004 at gmail.com (Jeffrey Leung) Date: Fri, 20 Aug 2010 22:14:41 -0700 Subject: [Freeswitch-users] mod_shout with vorbis support on Windows Message-ID: Is it possible to compile mod_shout on Windows to include vorbis support? I seem not to be getting information on how I enable vorbis support on the Windows version of mod_shout. From tharindufit at gmail.com Fri Aug 20 23:47:17 2010 From: tharindufit at gmail.com (Tharindu Madushanka) Date: Sat, 21 Aug 2010 12:17:17 +0530 Subject: [Freeswitch-users] [FreeSWITCH] How to register SIP users with FreeSWITCH using a client. Message-ID: Hi, I have built a SIP client for my University project and I could make SIP calls between two accounts using third party free SIP accounts. I used sip2sip.org to test the functionality. Now I would like to install a SIP server at my local machine and test how it works. Then I was looking for SIP open source projects and found this project. What I would like to do is. from my client app I want to register ( sign up ) for SIP accounts with FreeSWITCH server I installed.. Could somebody help me to give some points what I have to look into achieve this functionality.. It's really useful.. Thank you and Kind Regards, Tharindu Madushanka tharindufit.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/8c103018/attachment.html From tayeb.meftah at gmail.com Sun Aug 22 01:51:24 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 22 Aug 2010 10:51:24 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <585587.29855.qm@web53407.mail.re2.yahoo.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com> Message-ID: <4C70E50C.4030502@gmail.com> use latest git Le 21/08/2010 11:39, Sadjad Seyed-Ahmadian a ?crit : > I use latest subversion. > > > ------------------------------------------------------------------------ > *From:* Tihomir Culjaga > *To:* FreeSWITCH Users Help > *Sent:* Sat, August 21, 2010 1:41:50 PM > *Subject:* Re: [Freeswitch-users] Mod_h323 fail to load > > what version of ptlib are u using ? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/3992a062/attachment.html From peter.olsson at visionutveckling.se Sat Aug 21 10:46:54 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 21 Aug 2010 19:46:54 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <4C70E50C.4030502@gmail.com> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com>, <4C70E50C.4030502@gmail.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC058156@cooper> I believe that it's the version of PTLib being discussed, not FreeSWITCH :) /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Meftah Tayeb [tayeb.meftah at gmail.com] Skickat: den 22 augusti 2010 10:51 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Mod_h323 fail to load use latest git Le 21/08/2010 11:39, Sadjad Seyed-Ahmadian a ?crit : I use latest subversion. ________________________________ From: Tihomir Culjaga To: FreeSWITCH Users Help Sent: Sat, August 21, 2010 1:41:50 PM Subject: Re: [Freeswitch-users] Mod_h323 fail to load what version of ptlib are u using ? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz !DSPAM:4c700d9332936056157045! From 12ukwn at gmail.com Sat Aug 21 10:50:43 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 21 Aug 2010 19:50:43 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> Message-ID: <20100821195043.36fab3f3@anubis.defcon1> Le Sat, 21 Aug 2010 18:55:50 +0200, David Ponzone a ?crit : > Guys, > > As it seems that this debate is going to last forever without an > acceptable conclusion Is there blood already?? -- From mike at jerris.com Sat Aug 21 11:10:53 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 14:10:53 -0400 Subject: [Freeswitch-users] playback options In-Reply-To: References: Message-ID: <51DD11C0-0EC2-47AF-A7B2-EEE3E909F547@jerris.com> You could use mod_local_stream to always have this stream running and ready to go or adjust the buffer settings to your preference http://wiki.freeswitch.org/wiki/Mod_local_stream http://wiki.freeswitch.org/wiki/Mod_shout On Aug 19, 2010, at 9:02 PM, broken dash wrote: > I'm pulling shout cast streams into freeswitch using the playback > action and I'm wondering if there is a variable like > playback_delimeter that I could set within my dialplan/scripts that > would essentially load up the stream and chop off a configurable > amount of time before it essentially bridges up the audio to the > caller? or perhaps there is a dialplan routine that could do this for > me? From mike at jerris.com Sat Aug 21 11:14:18 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 14:14:18 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: It might be best to wait for mod_rtmp and compare functionality. It probably won't do exactly what you want right out of the box, but will be extendable by the community. Who is interested in working on rtmp/flash solutions for the open source once this module is released? Mike On Aug 12, 2010, at 4:01 AM, Dennis wrote: > it sounds as if there is a way to stream audio from fs over red5 to a > flash-player. > > could someone explain how it is done? we do not want any telephony or > any bi-directional things. we just need a simple one-way solution to > listen to calls. > > in the moment we stream like this: fs -> icecast2 -> flash > audio-player (which makes it a http stream) > > we would like to go another way: fs -> red5 -> flash audio-player > (where the stream to the flash player ist rtmp) > > it would be great, if someone could explain how it works! From mike at jerris.com Sat Aug 21 11:16:12 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 14:16:12 -0400 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> Message-ID: <150CAE7F-0509-4073-A49D-8BA4D45FE1F7@jerris.com> if it is inband dtmf and you don't turn on the dtmf detector, we wont look for or touch the tones. On Aug 16, 2010, at 8:56 AM, Dennis wrote: > phillip, it seems that we have somehow the same problem, but we have > an additional strange behavior. > > we used wireshark on the incoming network-card and on the outgoing > side. there we see two strange things, we can not figure out: > > 1.) on the incoming side, we receive the in-band tone from the cirpack > with a tiny gap, but fs recognizes them as 2 tones. > in wireshark it look something like this: | |||||||| > as i said, fs sees this as two tones. > > 2.) fs sends the tone above to the outgoing side as two tones, but > seems to cut the first ms of the second part. > in wireshark it looks something like this: | ||| > > > i feel, that we have to find out the following: > > 1.) why does the tone already have a gap, when we receive it? > > 2.) why does fs not leave the tone untouched? something happens with > the tone, while passing fs. is there a setting to avoid this? > > 3.) why is there no easy way to completely delete dtmf tones, if it is > possible with vuvuzela-noise? ;-) > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/640e3efc/attachment.html From mike at jerris.com Sat Aug 21 11:20:59 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 14:20:59 -0400 Subject: [Freeswitch-users] Webapi Examples location In-Reply-To: <4C6EE9A7.4050108@gmail.com> References: <4C6EE9A7.4050108@gmail.com> Message-ID: <98FA8F5A-EE00-4AFA-BC65-A4478B82A556@jerris.com> They are built in to the freeswitch modules, there is no additional source. Mike On Aug 20, 2010, at 4:46 PM, Phone wrote: > I am exploring the Webapi. > > The sample applications "Voicemail" and "Telecast" are working as they > should, but where do I find the source for them?? From mike at jerris.com Sat Aug 21 11:22:10 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 14:22:10 -0400 Subject: [Freeswitch-users] mod_shout with vorbis support on Windows In-Reply-To: References: Message-ID: <27E936CE-B3F3-4AFE-B8EE-39452058ECB6@jerris.com> It will require a patch to the build system to add this functionality. Mike On Aug 21, 2010, at 1:14 AM, Jeffrey Leung wrote: > Is it possible to compile mod_shout on Windows to include vorbis > support? I seem not to be getting information on how I enable vorbis > support on the Windows version of mod_shout. From mike at jerris.com Sat Aug 21 11:23:24 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 14:23:24 -0400 Subject: [Freeswitch-users] [FreeSWITCH] How to register SIP users with FreeSWITCH using a client. In-Reply-To: References: Message-ID: <516972F4-A2FA-401D-A728-C36B1F326CA6@jerris.com> http://wiki.freeswitch.org/wiki/Getting_Started_Guide should have all the information you need to accomplish this. Mike On Aug 21, 2010, at 2:47 AM, Tharindu Madushanka wrote: > Hi, > > I have built a SIP client for my University project and I could make SIP calls between two accounts using third party free SIP accounts. I used sip2sip.org to test the functionality. > > Now I would like to install a SIP server at my local machine and test how it works. Then I was looking for SIP open source projects and found this project. > > What I would like to do is. from my client app I want to register ( sign up ) for SIP accounts with FreeSWITCH server I installed.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/45208d0a/attachment.html From ritzalam at gmail.com Sat Aug 21 12:07:16 2010 From: ritzalam at gmail.com (Richard Alam) Date: Sat, 21 Aug 2010 15:07:16 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: Hi Mike, On Sat, Aug 21, 2010 at 2:14 PM, Michael Jerris wrote: > It might be best to wait for mod_rtmp and compare functionality. ?It probably won't do exactly what you want right out of the box, but will be extendable by the community. ?Who is interested in working on rtmp/flash solutions for the open source once this module is released? > We are very interested in mod_rtmp. We currently have a flash client that connects to FS conference. However, we have to create an intermediary Red5 app that takes RTMP audio (Speex WB and Ulaw) and send it to FS in RTP. If we can get rid of the Red5 app and connect directly to FS with mod_rtmp, that would be perfect. Richard > Mike > > On Aug 12, 2010, at 4:01 AM, Dennis wrote: > >> it sounds as if there is a way to stream audio from fs over red5 to a >> flash-player. >> >> could someone explain how it is done? we do not want any telephony or >> any bi-directional things. we just need a simple one-way solution to >> listen to calls. >> >> in the moment we stream like this: fs -> icecast2 -> flash >> audio-player (which makes it a http stream) >> >> we would like to go another way: fs -> red5 -> flash audio-player >> (where the stream to the flash player ist rtmp) >> >> it would be great, if someone could explain how it works! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton From chat2jesse at gmail.com Sat Aug 21 12:24:07 2010 From: chat2jesse at gmail.com (jesse) Date: Sat, 21 Aug 2010 12:24:07 -0700 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: I integrated Flash based phone with red5 and SIP system before, it works. but the problem is voice quality and echo issue, it sucks. all you can blame is Adobe. the web phone Sergio is java applet based. way too last century. VOIP user conference also hosts one java applet web based phone for conference dialing. -jesse On Fri, Aug 20, 2010 at 6:44 AM, Doddle WebPhone wrote: > You can also use SIP and RTP:? http://widget.doddlephone.com/? (web driven > phone) > > ?Sergio > > On Thu, Aug 12, 2010 at 5:01 AM, Dennis wrote: >> >> it sounds as if there is a way to stream audio from fs over red5 to a >> flash-player. >> >> could someone explain how it is done? we do not want any telephony or >> any bi-directional things. we just need a simple one-way solution to >> listen to calls. >> >> in the moment we stream like this: fs -> icecast2 -> flash >> audio-player (which makes it a http stream) >> >> we would like to go another way: fs -> red5 -> flash audio-player >> (where the stream to the flash player ist rtmp) >> >> it would be great, if someone could explain how it works! >> >> thanks >> dennis >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From chat2jesse at gmail.com Sat Aug 21 12:26:42 2010 From: chat2jesse at gmail.com (jesse) Date: Sat, 21 Aug 2010 12:26:42 -0700 Subject: [Freeswitch-users] simple echo failed In-Reply-To: <146EFC92-F165-4931-99C1-DAAEE2267233@freeswitch.org> References: <146EFC92-F165-4931-99C1-DAAEE2267233@freeswitch.org> Message-ID: thanks for your advice. you are so nice. now I feel guilty. :-) -jesse On Sat, Aug 21, 2010 at 9:20 AM, Brian West wrote: > I don't recall any examples or documentation showing that.... How about slowing down a little and paying closer attention to things instead of rushing thru them and missing the little things... We are all guilty of this from time to time. > > /b > > On Aug 21, 2010, at 2:37 AM, jesse wrote: > >> freaking stupid, it because the extra space between & e. >> why it is designed like this? so error prone and misleading >> >> -jesse > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From tculjaga at gmail.com Sat Aug 21 13:42:51 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 21 Aug 2010 22:42:51 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57DC058156@cooper> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com> <4C70E50C.4030502@gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C57DC058156@cooper> Message-ID: ptlib i use has 11M compressed ... how can i send it to you ? T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/43ce853b/attachment.html From 12ukwn at gmail.com Sat Aug 21 14:22:07 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 21 Aug 2010 23:22:07 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com> <4C70E50C.4030502@gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C57DC058156@cooper> Message-ID: <20100821232207.680c0850@anubis.defcon1> Le Sat, 21 Aug 2010 22:42:51 +0200, Tihomir Culjaga a ?crit : > ptlib i use has 11M compressed ... how can i send it to you ? > > > T. from man zip: Use -s to set the split size and create a split archive. The size is given as a number followed optionally by one of k (kB), m (MB), g (GB), or t (TB) (the default is m). The -sp option can be used to pause zip between splits to allow changing removable media, for example, but read the descriptions and warnings for both -s and -sp below. then send each piece in a different email (careful: the mime encoding raise data size 33%) -- From tculjaga at gmail.com Sat Aug 21 14:34:15 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 21 Aug 2010 23:34:15 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <20100821232207.680c0850@anubis.defcon1> References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com> <4C70E50C.4030502@gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C57DC058156@cooper> <20100821232207.680c0850@anubis.defcon1> Message-ID: On Sat, Aug 21, 2010 at 11:22 PM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote: > Le Sat, 21 Aug 2010 22:42:51 +0200, > Tihomir Culjaga a ?crit : > > > ptlib i use has 11M compressed ... how can i send it to you ? > > > > > > T. > > from man zip: > > Use -s to set the split size and create a split archive. The size is given > as a number followed optionally by one of k (kB), m (MB), g (GB), or t > (TB) (the default is m). The -sp option can be used to pause zip between > splits to allow changing removable media, for example, but read the > descriptions and warnings for both -s and -sp below. > > then send each piece in a different email (careful: the mime encoding > raise data size 33%) > > i was thinking more of an FTP server where i can upload... T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/97f673d6/attachment.html From xyangni at gmail.com Sat Aug 21 14:42:37 2010 From: xyangni at gmail.com (xuyan yang) Date: Sun, 22 Aug 2010 05:42:37 +0800 Subject: [Freeswitch-users] Half calls end up in echo mode without counter-party's sound, since 26Jul git version Message-ID: I have a IVR written in js running on git version. During the test in Jul it works fine. But recently, I find that nearly half calls end up in silence and only echoing its own sound. Checked the recording on server, it is the same. IVR's speech is played back to the FS instead of the caller. The version is compiled on 26jul git. The same problem exist for both of my SIP providers and skypeopen, So it should not be the providers' fault. The git tree today is also tested with the same problem. But when I roll back to 1.0.6 everything back to normal. As I am updating the version quite often in Jul, this is very likely to be caused by some changes just before 26Jul. Does anyone using git version after 26Jul have the same problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/8b91e480/attachment.html From 12ukwn at gmail.com Sat Aug 21 14:48:52 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 21 Aug 2010 23:48:52 +0200 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com> <4C70E50C.4030502@gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C57DC058156@cooper> <20100821232207.680c0850@anubis.defcon1> Message-ID: <20100821234852.406fb1a2@anubis.defcon1> Le Sat, 21 Aug 2010 23:34:15 +0200, Tihomir Culjaga a ?crit : > i was thinking more of an FTP server where i can upload... > > T. then use something like http://rapidshare.com/ and send back the reference to your correspondant. -- Beauty, brains, availability, personality; pick any two. From mike at jerris.com Sat Aug 21 14:56:28 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 17:56:28 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: I am afraid that we have little control over the echo cancelation as that is all done in the client and doing so on the far end will be very cpu intensive and not that useful. Using flash for 2-way media will certainly require a headset for decent results at least with flash 10. If you want something better, your installing a full application, be that java or something else, at which point you might as well be using a sip softphone. Audio quality shouldn't be an issue with flash, indeed I have had quite good audio quality results in testing when using a headset. Also, push to talk applications should work quite well. Mike On Aug 21, 2010, at 3:24 PM, jesse wrote: > I integrated Flash based phone with red5 and SIP system before, it works. > but the problem is voice quality and echo issue, it sucks. all you can > blame is Adobe. > > the web phone Sergio is java applet based. way too last century. VOIP > user conference > also hosts one java applet web based phone for conference dialing. > From chris at cheeky.org Sat Aug 21 11:38:17 2010 From: chris at cheeky.org (Chris Hemmings) Date: Sat, 21 Aug 2010 19:38:17 +0100 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: Who is interested in working on rtmp/flash solutions for the open source > once this module is released? > Absolutely, could potentially open up some very interesting avenues. :-) Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/58b5cce9/attachment-0001.html From chris at cheeky.org Sat Aug 21 12:44:10 2010 From: chris at cheeky.org (Chris Hemmings) Date: Sat, 21 Aug 2010 20:44:10 +0100 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: I spotted this a few days ago, perhaps we will be able to live without Adobe all together one day. http://dev.w3.org/html5/html-device/ Chris On 21 August 2010 20:24, jesse wrote: > I integrated Flash based phone with red5 and SIP system before, it works. > but the problem is voice quality and echo issue, it sucks. all you can > blame is Adobe. > > the web phone Sergio is java applet based. way too last century. VOIP > user conference > also hosts one java applet web based phone for conference dialing. > > -jesse > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/1f6eb2ed/attachment-0001.html From mike at jerris.com Sat Aug 21 14:58:12 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 17:58:12 -0400 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: References: <40241.74408.qm@web53405.mail.re2.yahoo.com> <20100820222722.46e7ccb9@anubis.defcon1> <31192.48276.qm@web53406.mail.re2.yahoo.com> <739604.57080.qm@web53408.mail.re2.yahoo.com> <305309.80502.qm@web53403.mail.re2.yahoo.com> <585587.29855.qm@web53407.mail.re2.yahoo.com> <4C70E50C.4030502@gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C57DC058156@cooper> Message-ID: <98B2DE2B-A79D-47DD-BB23-F7CAA47BE533@jerris.com> If you send it to me via private email or some other method, I can toss it on http://files.freeswitch.org for you if you like? Mike On Aug 21, 2010, at 4:42 PM, Tihomir Culjaga wrote: > ptlib i use has 11M compressed ... how can i send it to you ? > > > T. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Sat Aug 21 14:59:13 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 17:59:13 -0400 Subject: [Freeswitch-users] Half calls end up in echo mode without counter-party's sound, since 26Jul git version In-Reply-To: References: Message-ID: can you git bisect to nail down the specific revision that caused this issue and post the issue to http://jira.freeswitch.org please? Mike On Aug 21, 2010, at 5:42 PM, xuyan yang wrote: > I have a IVR written in js running on git version. During the test in Jul it works fine. But recently, I find that nearly half calls end up in silence and only echoing its own sound. Checked the recording on server, it is the same. IVR's speech is played back to the FS instead of the caller. > The version is compiled on 26jul git. > The same problem exist for both of my SIP providers and skypeopen, So it should not be the providers' fault. > > The git tree today is also tested with the same problem. > But when I roll back to 1.0.6 everything back to normal. As I am updating the version quite often in Jul, this is very likely to be caused by some changes just before 26Jul. > Does anyone using git version after 26Jul have the same problem? From vince.freeswitch at hightek.org Sat Aug 21 15:03:15 2010 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Sat, 21 Aug 2010 17:03:15 -0500 Subject: [Freeswitch-users] Platform requrest (DragonflyBSD) for the Jira issue management system In-Reply-To: <20100728053058.GA26436@quark.hightek.org> References: <20100728053058.GA26436@quark.hightek.org> Message-ID: <20100821220315.GA52528@quark.hightek.org> On Wed, Jul 28, 2010 at 12:30:58AM -0500, Vincent Stemen wrote: > Hopefully this is an approprate place for this request. > > I would like to see "DragonFlyBSD/GCC" added as a Platform option. > > Regards, > - Vince Hi. Just a reminder. I just checked, and this platform it is still not in the list. From phone.bytes at gmail.com Sat Aug 21 15:27:44 2010 From: phone.bytes at gmail.com (phone.bytes) Date: Sat, 21 Aug 2010 16:27:44 -0600 Subject: [Freeswitch-users] Webapi Examples location In-Reply-To: <98FA8F5A-EE00-4AFA-BC65-A4478B82A556@jerris.com> References: <4C6EE9A7.4050108@gmail.com> <98FA8F5A-EE00-4AFA-BC65-A4478B82A556@jerris.com> Message-ID: <4C7052E0.6000108@gmail.com> Thanks for the reply. Are there any examples available to look at that would be helpful in expanding on these two sample applications shown? On 8/21/2010 12:20 PM, Michael Jerris wrote: > They are built in to the freeswitch modules, there is no additional source. > > Mike > > On Aug 20, 2010, at 4:46 PM, Phone wrote: > >> I am exploring the Webapi. >> >> The sample applications "Voicemail" and "Telecast" are working as they >> should, but where do I find the source for them?? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From xyangni at gmail.com Sat Aug 21 15:33:11 2010 From: xyangni at gmail.com (xuyan yang) Date: Sun, 22 Aug 2010 06:33:11 +0800 Subject: [Freeswitch-users] Half calls end up in echo mode without counter-party's sound, since 26Jul git version In-Reply-To: References: Message-ID: OK. Posted to jira. But after 20 test, 1 occurrence of the problem is also found in FS 1.0.6 I will test whether it is "js only" problem. On Sun, Aug 22, 2010 at 5:59 AM, Michael Jerris wrote: > can you git bisect to nail down the specific revision that caused this > issue and post the issue to http://jira.freeswitch.org please? > > Mike > > On Aug 21, 2010, at 5:42 PM, xuyan yang wrote: > > > I have a IVR written in js running on git version. During the test in Jul > it works fine. But recently, I find that nearly half calls end up in silence > and only echoing its own sound. Checked the recording on server, it is the > same. IVR's speech is played back to the FS instead of the caller. > > The version is compiled on 26jul git. > > The same problem exist for both of my SIP providers and skypeopen, So it > should not be the providers' fault. > > > > The git tree today is also tested with the same problem. > > But when I roll back to 1.0.6 everything back to normal. As I am updating > the version quite often in Jul, this is very likely to be caused by some > changes just before 26Jul. > > Does anyone using git version after 26Jul have the same problem? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/d26f70a7/attachment.html From mike at jerris.com Sat Aug 21 15:38:48 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 18:38:48 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: Are any browsers supporting 2-way media streams yet? Mike On Aug 21, 2010, at 3:44 PM, Chris Hemmings wrote: > I spotted this a few days ago, perhaps we will be able to live without Adobe all together one day. > > http://dev.w3.org/html5/html-device/ > > Chris > > > On 21 August 2010 20:24, jesse wrote: > I integrated Flash based phone with red5 and SIP system before, it works. > but the problem is voice quality and echo issue, it sucks. all you can > blame is Adobe. > > the web phone Sergio is java applet based. way too last century. VOIP > user conference > also hosts one java applet web based phone for conference dialing. > > -jesse -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/22faabab/attachment.html From mike at jerris.com Sat Aug 21 15:43:35 2010 From: mike at jerris.com (Michael Jerris) Date: Sat, 21 Aug 2010 18:43:35 -0400 Subject: [Freeswitch-users] Webapi Examples location In-Reply-To: <4C7052E0.6000108@gmail.com> References: <4C6EE9A7.4050108@gmail.com> <98FA8F5A-EE00-4AFA-BC65-A4478B82A556@jerris.com> <4C7052E0.6000108@gmail.com> Message-ID: <707FF4B9-03B7-401E-A9B6-80F5C45565B1@jerris.com> The examples are the ones you already mentioned. They use some stub code in mod_xml_rpc and api commands in the modules themselves. Check out in mod_voicemail.c:4227 SWITCH_STANDARD_API(voicemail_api_function) http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_voicemail/mod_voicemail.c?r=b5205c0bc0f3c6a0c834679f4abff0a3297f2bc5 On Aug 21, 2010, at 6:27 PM, phone.bytes wrote: > Thanks for the reply. > > Are there any examples available to look at that would be helpful in > expanding on these two sample applications shown? > > On 8/21/2010 12:20 PM, Michael Jerris wrote: >> They are built in to the freeswitch modules, there is no additional source. >> >> Mike >> >> On Aug 20, 2010, at 4:46 PM, Phone wrote: >> >>> I am exploring the Webapi. >>> >>> The sample applications "Voicemail" and "Telecast" are working as they >>> should, but where do I find the source for them?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/24200214/attachment.html From brian at freeswitch.org Sat Aug 21 15:45:47 2010 From: brian at freeswitch.org (Brian West) Date: Sat, 21 Aug 2010 17:45:47 -0500 Subject: [Freeswitch-users] Platform requrest (DragonflyBSD) for the Jira issue management system In-Reply-To: <20100821220315.GA52528@quark.hightek.org> References: <20100728053058.GA26436@quark.hightek.org> <20100821220315.GA52528@quark.hightek.org> Message-ID: <3BFC3227-59D0-4D45-974E-1E3C04EF6547@freeswitch.org> Vince, I see no value in adding it... but I added it anyway... Next time please open a jira about this stuff. /b On Aug 21, 2010, at 5:03 PM, Vincent Stemen wrote: > On Wed, Jul 28, 2010 at 12:30:58AM -0500, Vincent Stemen wrote: >> Hopefully this is an approprate place for this request. >> >> I would like to see "DragonFlyBSD/GCC" added as a Platform option. >> >> Regards, >> - Vince > > Hi. Just a reminder. I just checked, and this platform it is still not > in the list. From steveayre at gmail.com Sat Aug 21 15:49:32 2010 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 21 Aug 2010 23:49:32 +0100 Subject: [Freeswitch-users] [FreeSWITCH] How to register SIP users with FreeSWITCH using a client. In-Reply-To: References: Message-ID: I would look at mod_xml_curl (http://wiki.freeswitch.org/wiki/Mod_xml_curl) This allows you to offload the user directory to a web server with a database backend. You could develop a script on the same web server which the client uses to register the account. The script creates the account in the database. Your client then REGISTERs to Freeswitch, which will access the web server via mod_xml_curl to see whether the username+password were valid. The web script checks the database, and then returns that access is either allowed or denied. The web server (e.g. apache), database (e.g. mysql) and freeswitch can all run on the same server if you like. -Steve On 21 August 2010 07:47, Tharindu Madushanka wrote: > Hi, > > I have built a SIP client for my University project and I could make SIP > calls between two accounts using third party free SIP accounts. I used > sip2sip.org to test the functionality. > > Now I would like to install a SIP server at my local machine and test how > it works. Then I was looking for SIP open source projects and found this > project. > > What I would like to do is. from my client app I want to register ( sign up > ) for SIP accounts with FreeSWITCH server I installed.. > > Could somebody help me to give some points what I have to look into achieve > this functionality.. > > It's really useful.. > > Thank you and Kind Regards, > > Tharindu Madushanka > tharindufit.wordpress.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/bd6195c3/attachment.html From xyangni at gmail.com Sat Aug 21 16:15:40 2010 From: xyangni at gmail.com (xuyan yang) Date: Sun, 22 Aug 2010 07:15:40 +0800 Subject: [Freeswitch-users] Half calls end up in echo mode without counter-party's sound, since 26Jul git version In-Reply-To: References: Message-ID: This is a javascript only problem. When the speech is played through XML dialplan. There is no problem at least for my 30 test call. But such problem exist even for the simplest js script with 1 line of speak and 1 line of sleep. 1.0.6 also have this problem but probability is lower. I will update jira. On Sun, Aug 22, 2010 at 6:33 AM, xuyan yang wrote: > OK. Posted to jira. But after 20 test, 1 occurrence of the problem is also > found in FS 1.0.6 > I will test whether it is "js only" problem. > > > On Sun, Aug 22, 2010 at 5:59 AM, Michael Jerris wrote: > >> can you git bisect to nail down the specific revision that caused this >> issue and post the issue to http://jira.freeswitch.org please? >> >> Mike >> >> On Aug 21, 2010, at 5:42 PM, xuyan yang wrote: >> >> > I have a IVR written in js running on git version. During the test in >> Jul it works fine. But recently, I find that nearly half calls end up in >> silence and only echoing its own sound. Checked the recording on server, it >> is the same. IVR's speech is played back to the FS instead of the caller. >> > The version is compiled on 26jul git. >> > The same problem exist for both of my SIP providers and skypeopen, So it >> should not be the providers' fault. >> > >> > The git tree today is also tested with the same problem. >> > But when I roll back to 1.0.6 everything back to normal. As I am >> updating the version quite often in Jul, this is very likely to be caused by >> some changes just before 26Jul. >> > Does anyone using git version after 26Jul have the same problem? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/135f3630/attachment.html From vhatz at kinetix.gr Sat Aug 21 16:25:23 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Sun, 22 Aug 2010 02:25:23 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> Message-ID: <4C706063.1040906@kinetix.gr> Nyamul Hassan wrote: > Vlasis, Woody's 2nd email talked about 600 CPS on v1.04. And, perhaps > you can share your usage scenario, and how much pressure you could put > on FreeSwitch before it "crashed"? > I must have missed Woody's 2nd email... > Will like to see your implementation case, btw. > Sure, I'll be happy to. Should I contact you off list? Best regards, Vlasis. From adminjew at gmail.com Sat Aug 21 20:35:16 2010 From: adminjew at gmail.com (Yitzchok) Date: Sat, 21 Aug 2010 23:35:16 -0400 Subject: [Freeswitch-users] Phrase speak-text function return on first key press in phrase file on Windows Message-ID: NOTE: I tested this on Centos and I wasn't able to reproduce it. So it looks like a windows problem or it might be that since I got one error when trying to compile FreeSWITCH on windows I tried to fix it with the fix on this jira bug report http://jira.freeswitch.org/browse/FSBUILD-296?focusedCommentId=20921&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel so maybe that is the problem. (I am using Flite for TTS) -- I have a macro (phrase) file that is set up like this. Then when I use this phrase file in the play_and_get_digits application, something like this. If I try to enter a number on the phone for example "234#" what happens here is that if it is in the middle of playing the *speak-text *part of the phrase file then it will just return "2" and stop right away (as if he is done) if "234#" is pressed when the *play-file* part of phrase file is playing then it will work as it should work (returning 234). Yitzchok -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100821/86568593/attachment.html From mnhassan at usa.net Sat Aug 21 22:52:26 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Sun, 22 Aug 2010 11:52:26 +0600 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C706063.1040906@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> <4C706063.1040906@kinetix.gr> Message-ID: Why? On list is fine. Apparently many would like to see the "performance" that others get from their implementation. You could also contribute to the "wiki page", that David was referring to. Regards HASSAN On Sun, Aug 22, 2010 at 05:25, Vlasis Hatzistavrou (KTI) wrote: > Nyamul Hassan wrote: > > Vlasis, Woody's 2nd email talked about 600 CPS on v1.04. And, perhaps > > you can share your usage scenario, and how much pressure you could put > > on FreeSwitch before it "crashed"? > > > I must have missed Woody's 2nd email... > > > Will like to see your implementation case, btw. > > > > Sure, I'll be happy to. Should I contact you off list? > > > Best regards, > Vlasis. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/4b00ff0e/attachment.html From chris at cheeky.org Sun Aug 22 02:32:13 2010 From: chris at cheeky.org (Chris Hemmings) Date: Sun, 22 Aug 2010 10:32:13 +0100 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: Sadly, I believe there isn't yet. Looking at the draft, it's only a few days old so I imagine it will be a while before anything appears to play with. Chris On 21 August 2010 23:38, Michael Jerris wrote: > Are any browsers supporting 2-way media streams yet? > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/2c5fb1b3/attachment-0001.html From dujinfang at gmail.com Sun Aug 22 04:29:37 2010 From: dujinfang at gmail.com (Seven Du) Date: Sun, 22 Aug 2010 19:29:37 +0800 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: On Sun, Aug 22, 2010 at 2:14 AM, Michael Jerris wrote: > It might be best to wait for mod_rtmp and compare functionality. ?It probably won't do exactly what you want right out of the box, but will be extendable by the community. ?Who is interested in working on rtmp/flash solutions for the open source once this module is released? > Is there any ETA of the mod ? I'm interested. > Mike > > On Aug 12, 2010, at 4:01 AM, Dennis wrote: > >> it sounds as if there is a way to stream audio from fs over red5 to a >> flash-player. >> >> could someone explain how it is done? we do not want any telephony or >> any bi-directional things. we just need a simple one-way solution to >> listen to calls. >> >> in the moment we stream like this: fs -> icecast2 -> flash >> audio-player (which makes it a http stream) >> >> we would like to go another way: fs -> red5 -> flash audio-player >> (where the stream to the flash player ist rtmp) >> >> it would be great, if someone could explain how it works! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From ssa1357 at yahoo.com Sun Aug 22 04:40:55 2010 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Sun, 22 Aug 2010 04:40:55 -0700 (PDT) Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <4C70E50C.4030502@gmail.com> Message-ID: <881664.71177.qm@web53407.mail.re2.yahoo.com> Finally I compile mod_h323 correctly with copying lib files to freeswitch source directory. It is now loaded correctly but when I want to bridge it to sip I got user-busy!!! --- On Sun, 8/22/10, Meftah Tayeb wrote: From: Meftah Tayeb Subject: Re: [Freeswitch-users] Mod_h323 fail to load To: "FreeSWITCH Users Help" Cc: "Sadjad Seyed-Ahmadian" Date: Sunday, August 22, 2010, 1:21 PM use latest git Le 21/08/2010 11:39, Sadjad Seyed-Ahmadian a ?crit?: I use latest subversion. From: Tihomir Culjaga To: FreeSWITCH Users Help Sent: Sat, August 21, 2010 1:41:50 PM Subject: Re: [Freeswitch-users] Mod_h323 fail to load what version of ptlib are u using ? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/49344212/attachment.html From xyangni at gmail.com Sun Aug 22 05:19:18 2010 From: xyangni at gmail.com (xuyan yang) Date: Sun, 22 Aug 2010 20:19:18 +0800 Subject: [Freeswitch-users] how to answer skype call after recent change? Message-ID: Hi, I have met a problem that skype to skype call become silent when eavesdrop is working. And already reported to JIRA. When trying to test with latest git head, SKYPE never answer any call. It seems to be a recent upgrade which has not been documented. How should I change the config to let skype answer all calls as before? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/112f5331/attachment.html From vhatz at kinetix.gr Sun Aug 22 06:06:21 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Sun, 22 Aug 2010 16:06:21 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> <4C706063.1040906@kinetix.gr> Message-ID: <4C7120CD.9000200@kinetix.gr> Nyamul Hassan wrote: > Why? On list is fine. Apparently many would like to see the > "performance" that others get from their implementation. > Well, I thought too that many would like to see performance stats from other users, but after all the discussion in this thread I drew the conclusion that an on-list discussion about performance was actually _not_ fine. After all, you wrote that you understood the devs' irritation about performance discussions and this is why I proposed to send you the data off list, since you are interested. > You could also contribute to the "wiki page", that David was referring to. > I'll just proceed to post the data on the wiki page, then. Rgds, Vlasis. From vince.freeswitch at hightek.org Sun Aug 22 08:41:28 2010 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Sun, 22 Aug 2010 10:41:28 -0500 Subject: [Freeswitch-users] Platform requrest (DragonflyBSD) for the Jira issue management system In-Reply-To: <3BFC3227-59D0-4D45-974E-1E3C04EF6547@freeswitch.org> References: <20100728053058.GA26436@quark.hightek.org> <20100821220315.GA52528@quark.hightek.org> <3BFC3227-59D0-4D45-974E-1E3C04EF6547@freeswitch.org> Message-ID: <20100822154128.GA62723@quark.hightek.org> On Sat, Aug 21, 2010 at 05:45:47PM -0500, Brian West wrote: > Vince, > I see no value in adding it... but I added it anyway... I appreciate it. May I ask why you do not see any value in it though? If you prefer to encompass all the BSD's into one platform option, I would suggest changing the "FreeBSD/gcc" option to "BSD/gcc". > Next time please open a jira about this stuff. OK. I will try to keep that in mind. You might want to consider mentioning that in the instructions on the form. It just says The Platform the issue is experienced on. If not available please contact an Administrator to add your os/compiler combination For most projects, instructions to contact an administrator or maintainer usually means either emailing them directly or posting to the mailing list. I even asked on the IRC what was the most appropriate way to contact an administrator about the request, prior to posting it to the list, and nobody mentioned using jira for it. > On Aug 21, 2010, at 5:03 PM, Vincent Stemen wrote: > > > On Wed, Jul 28, 2010 at 12:30:58AM -0500, Vincent Stemen wrote: > >> Hopefully this is an approprate place for this request. > >> > >> I would like to see "DragonFlyBSD/GCC" added as a Platform option. > >> > >> Regards, > >> - Vince > > > > Hi. Just a reminder. I just checked, and this platform it is still not > > in the list. From gmaruzz at celliax.org Sun Aug 22 09:02:01 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 22 Aug 2010 18:02:01 +0200 Subject: [Freeswitch-users] how to answer skype call after recent change? In-Reply-To: References: Message-ID: On Sun, Aug 22, 2010 at 2:19 PM, xuyan yang wrote: > Hi, > I have met a problem that skype to skype call become silent when > eavesdrop?is working. And already reported to JIRA. When trying to test with > latest git head, SKYPE never answer any call. It seems to be a recent > upgrade which has not been documented. How should I change the config?to let > skype answer all calls as before? The new behavior of mod_skypopen is to answer incoming calls when it's directed to do so with an explicit "answer" command in dialplan (or equivalent from esl or whatever). >From the Jira I see you are on windows, and I had not tested this latest git commit on windows. So, maybe (just maybe) on windows it does not answer calls. Just revert to the mod_skypopen version prior to the last mod_skypopen git commit. For the eavesdropping silent problem, I answered on Jira: need more info. -giovanni > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From steveayre at gmail.com Sun Aug 22 10:42:36 2010 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 22 Aug 2010 18:42:36 +0100 Subject: [Freeswitch-users] Mod_h323 fail to load In-Reply-To: <881664.71177.qm@web53407.mail.re2.yahoo.com> References: <4C70E50C.4030502@gmail.com> <881664.71177.qm@web53407.mail.re2.yahoo.com> Message-ID: What does the freeswitch log say? -Steve On 22 August 2010 12:40, Sadjad Seyed-Ahmadian wrote: > > > Finally I compile mod_h323 correctly with copying lib files to freeswitch > source directory. It is now loaded correctly but when I want to bridge it to > sip I got user-busy!!! > > > --- On *Sun, 8/22/10, Meftah Tayeb * wrote: > > > From: Meftah Tayeb > > Subject: Re: [Freeswitch-users] Mod_h323 fail to load > To: "FreeSWITCH Users Help" > Cc: "Sadjad Seyed-Ahmadian" > Date: Sunday, August 22, 2010, 1:21 PM > > > use latest git > Le 21/08/2010 11:39, Sadjad Seyed-Ahmadian a ?crit : > > I use latest subversion. > > > ------------------------------ > *From:* Tihomir Culjaga > *To:* FreeSWITCH Users Help > *Sent:* Sat, August 21, 2010 1:41:50 PM > *Subject:* Re: [Freeswitch-users] Mod_h323 fail to load > > what version of ptlib are u using ? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Meftah Tayeb > alg?rie t?l?com SPA > phone: +21321761805 > phone (INUM): +883510001289101 > mobile : +213660347746 > mobile (INUM: +883510001289110 > http://www.algerietelecom.dz > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/ec6d07ef/attachment-0001.html From vince.freeswitch at hightek.org Sun Aug 22 10:53:11 2010 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Sun, 22 Aug 2010 12:53:11 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: Message-ID: <20100822175311.GA63925@quark.hightek.org> On Fri, Aug 20, 2010 at 05:29:31PM +0800, Woody Dickson wrote: > Hi, > > I am doing some experiments with Freeswitch by torturing it to see how > the machine's CPU response to heavy loaded situation. > The test is done on a 16 core 5550 dual quad core server running > fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. > > What I found so strange was that while CPU usage remains pretty low > and distributed among all cores at 190 - 200 calls per second. Then, > after added a few more calls per second, all CPU becomes fully > utilized. > > Is this due to some wrong setting? Any idea how I can tweak the > configuration and continue my test? > > Thanks, > Woody Hi Woody. I would hazard to guess that this could be a Linux resource management issue. I don't have any experience with SMP on Linux, but Linux has a long history of memory management (among other) problems. We ran Linux exclusively on all our servers and workstations for over 10 years before finally switching to BSD several years ago. We had continuous problems ranging from minor strange unexplained behaviours, as you describe, to what appeared to be bugs in applications, to outright crashes and freezes of the whole OS every day. When we switched to BSD nearly all the problems went away. Even some of the (what appeared to be) bugs in Linux binary applications went way, going from Linux to BSD running under Linux emulation (without re-compiling), using the same Linux libraries on the same hardware. An interesting test would be to try the same load test with BSD on the same machine and see if you get a similar result. Regards, Vince From vince.freeswitch at hightek.org Sun Aug 22 11:10:11 2010 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Sun, 22 Aug 2010 13:10:11 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C6EAB1F.9060904@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> Message-ID: <20100822181011.GB63925@quark.hightek.org> On Fri, Aug 20, 2010 at 07:19:43PM +0300, Vlasis Hatzistavrou wrote: > On 20/8/2010 5:52 ??, Anthony Minessale wrote: > > Yet another load test thread......................... > > > > This is why we don't like them. > > > > > > Too bad you missed my presentation at cluecon..... > > > > Do you hear yourself btw? Quibbling about completely free software > > *only* doing 200cps in your fake test? > I don't know if I missed any message in this thread, but why do you say > that the tester did a fake test? > > > 200cps is 4 times the industry standard FYI and if you were really > > doing 200cps in real life you would be so rich at our expense that you > > could afford more boxes. > > > > you clearly ignored all of our repeated recommendations. > > Use the following or you are on your own....... > > > > This OS > > Centos 5.x x86_64 > > If someone is on his own for any other OS than Centos 5.x x86_64 then > why is FS offered for so many different platforms? > > It's only natural for people to want to test it on their favorite > platform... Perhaps, as you recommend, Centos 5.x x86_64 gives the best > results over all other platforms, but last time I checked, a user is > still allowed to ask performance related questions for his favorite > platform in this mailing list, right? I agree. Woody Dickson said What I found so strange was that while CPU usage remains pretty low and distributed among all cores at 190 - 200 calls per second. Then, after added a few more calls per second, all CPU becomes fully utilized. I did not interpret his comments as having anything to do with criticizing or questioning the performance of freeswitch. His question was one of curiosity about the extreme non-linearity of the processor load suddenly surging by adding just a few calls. If I had seen this same behaviour I likely would have asked the same question. Just my two cents worth. Vince > Woody's post was not expressing any negative opinion about FS's > performance. He didn't comment on whether 200cps was too little or too > much. He just wrote that the CPU usage increased in an unexpected manner > and wonders how/if he can solve this. He wrote that he was doing a load > test scenario, he didn't write that he was making money at 200cps "at > your expense". And he wrote all this in a manner which was neither > insulting nor abusive. > > The poster just wanted to learn something, see if others have gotten > better results with other setups. He was trying to make a valid > discussion to solve a problem, and other users were actually replying to > him. > > This is what mailing lists are for, right? > > I really don't understand why red flags have to be raised every time the > word "performance" is mentioned in this mailing list. > > -- > Best regards, > Vlasis Hatzistavrou. From chat2jesse at gmail.com Sun Aug 22 12:10:29 2010 From: chat2jesse at gmail.com (jesse) Date: Sun, 22 Aug 2010 12:10:29 -0700 Subject: [Freeswitch-users] Does FS ESL provide session command? Message-ID: According to http://wiki.freeswitch.org/wiki/Event_Socket, ESL only supports: api, bapi, event, myevents, divert_events, filter,sendevent, sendmsg, hangup, auth, exit,log, nolog, noevents, nixevent. How come I don't see a way to create a session via ESL. like in lua, freeswitch.Session(dialstring); one way to accomplish this in ESL is : api originate dialstring &hold(), is this = Session(dialstring) ? thanks From chat2jesse at gmail.com Sun Aug 22 12:12:59 2010 From: chat2jesse at gmail.com (jesse) Date: Sun, 22 Aug 2010 12:12:59 -0700 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: I don't think so, HTML5 is still evolving slowly. It will take some time for HTML5 to take off. -jesse On Sat, Aug 21, 2010 at 3:38 PM, Michael Jerris wrote: > Are any browsers supporting 2-way media streams yet? > Mike > On Aug 21, 2010, at 3:44 PM, Chris Hemmings wrote: > > I spotted this a few days ago,?? perhaps we will be able to live without > Adobe all together one day. > > http://dev.w3.org/html5/html-device/ > > Chris > > > On 21 August 2010 20:24, jesse wrote: >> >> I integrated Flash based phone with red5 and SIP system before, it works. >> but the problem is voice quality and echo issue, it sucks. all you can >> blame is Adobe. >> >> the web phone Sergio is java applet based. way too last century. VOIP >> user conference >> also hosts one java applet web based phone for conference dialing. >> >> -jesse > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From chat2jesse at gmail.com Sun Aug 22 12:17:07 2010 From: chat2jesse at gmail.com (jesse) Date: Sun, 22 Aug 2010 12:17:07 -0700 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: " I have had quite good audio quality results in testing when using a headset. " , next time try to call a cell phone from your flash client. The root cause is FLASH only supports very limited codec set. anyway, please count me in for the testing of mod_xmpp. -jesse On Sat, Aug 21, 2010 at 2:56 PM, Michael Jerris wrote: > I am afraid that we have little control over the echo cancelation as that is all done in the client and doing so on the far end will be very cpu intensive and not that useful. ?Using flash for 2-way media will certainly require a headset for decent results at least with flash 10. ?If you want something better, your installing a full application, be that java or something else, at which point you might as well be using a sip softphone. ?Audio quality shouldn't be an issue with flash, indeed I have had quite good audio quality results in testing when using a headset. ?Also, push to talk applications should work quite well. > > Mike > > On Aug 21, 2010, at 3:24 PM, jesse wrote: > >> I integrated Flash based phone with red5 and SIP system before, it works. >> but the problem is voice quality and echo issue, it sucks. all you can >> blame is Adobe. >> >> the web phone Sergio is java applet based. way too last century. VOIP >> user conference >> also hosts one java applet web based phone for conference dialing. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From math.parent at gmail.com Sun Aug 22 15:27:18 2010 From: math.parent at gmail.com (Mathieu Parent) Date: Mon, 23 Aug 2010 00:27:18 +0200 Subject: [Freeswitch-users] mod_skinny issue In-Reply-To: References: Message-ID: On Fri, Aug 6, 2010 at 8:14 AM, Terry Moore-Read wrote: > When starting freeswitch, I'm seeing the following : > > > 2010-08-05 23:08:43.665581 [ERR] mod_skinny.c:357 SQL ERR: [DELETE > FROM skinny_devices] no such table: skinny_devices > 2010-08-05 23:08:43.666177 [ERR] mod_skinny.c:357 SQL ERR: [DELETE > FROM skinny_lines] no such table: skinny_lines > 2010-08-05 23:08:43.666761 [ERR] mod_skinny.c:357 SQL ERR: [DELETE > FROM skinny_buttons] no such table: skinny_buttons > 2010-08-05 23:08:43.667323 [ERR] mod_skinny.c:357 SQL ERR: [DELETE > FROM skinny_active_lines] no such table: skinny_active_lines > > > It appears from mod_skinny.c that these tables should have been > created by this point if they don't exist. ? I'm not sure why ?this > isn't happening. Yes, this should happen. This can be a permission problem in the db directory. check the file named skinny_internal.db. > FreeSWITCH Version 1.0.head (git-b60d6b3 2010-08-05 16-07-14 -0400) > using sqlite Mathieu Parent From dujinfang at gmail.com Sun Aug 22 19:05:23 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 23 Aug 2010 10:05:23 +0800 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: Message-ID: Talked with brian, he thought the there might not much difference between them and suggest a perf test by me. I test on my Mac 10.6.4 64bit. First I increased cps to 100 and max-sessions to 8000. It seems that it cannot create more than 2560 sessions on Mac, and I don't know how to increase the limit so I just use small numbers. The following ruby code create channels through ESL slowly. 30 * 10 * 2 means 600 channels, and because I used loopback, it actually use 1200 channels. bridge: 1200 threads. 300-400% cpu. (on Activity monitor) and load avg 600-800 (on top). Memory 300M. conf: 2100 threads. 300-400%cpu, load avg 600-800, or can be 39-1000(dosen't make sense?) memory 500M. 30.times do |i| puts i * 10 10.times do |j| if ARGV[0].nil? #bridge conn.bgapi("originate", "loopback/9664 &bridge(loopback/9664)") else #conference conf = "c#{i* 10 + j}@default" conn.bgapi("originate", "loopback/9664 &conference(#{conf})") conn.bgapi("originate", "loopback/9664 &conference(#{conf})") end end sleep 1 end when I double the channels( 30 * 20 = i * j), ESL stuck when threads up to 2560 and it throws "cannot create channels". But, when I run "hupall" in FS, it start creating channels again. Don't know why FS on Mac has a 2560 threads limit. And after "hupall" again, there are dead channels(9 channels and 9 threads): 4d5a2408-eaf6-4f3a-a401-7a916b1911f1,outbound,2010-08-23 09:52:35,1282528355,loopback/9664-a,CS_EXECUTE,,0000000000,,9664,conference,c193 at default,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,ffc4367e-bcde-43a5-a95e-e0fd5c4069ea f93fa824-a6ae-47ca-ae9c-ff6b8948df4f,outbound,2010-08-23 09:52:35,1282528355,loopback/9664-a,CS_EXECUTE,,0000000000,,9664,conference,c194 at default,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,4c8b1826-5afa-43ad-a73b-59a8c5d8f41c 7611288b-be9e-4f70-9880-3919de567222,inbound,2010-08-23 09:52:35,1282528355,loopback/9664-b,CS_REPORTING,,0000000000,,9664,playback,local_stream://moh,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,, 9142d487-f0b5-4636-a2eb-a0adaee19634,inbound,2010-08-23 09:52:35,1282528355,loopback/9664-b,CS_REPORTING,,0000000000,,9664,playback,local_stream://moh,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,, 0d8bc340-edf7-4461-b1b4-91056c68474b,outbound,2010-08-23 09:52:35,1282528355,loopback/9664-a,CS_NEW,,,,,,,,,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,a9a8369e-f509-4b39-b689-8e616d29d5c3 ed93c3d5-6780-4e75-bf48-6e326274be04,outbound,2010-08-23 09:52:35,1282528355,loopback/9664-a,CS_NEW,,,,,,,,,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,e321a328-0b7c-4f74-9af5-a5e3f083ff8a 62aabb4b-4bfc-4ba0-a87c-2e1a76323cce,outbound,2010-08-23 09:52:39,1282528359,loopback/9664-a,CS_EXECUTE,,0000000000,,9664,conference,c200 at default,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,3a86d88d-33b2-42b2-a57c-e2cb2c1ec486 b13c3e9d-7fb8-46f2-90c9-04a0ec770a2e,inbound,2010-08-23 09:52:39,1282528359,loopback/9664-b,CS_REPORTING,,0000000000,,9664,playback,local_stream://moh,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,, b8bb7b64-c57c-4c8f-9264-09eecf4aef57,outbound,2010-08-23 09:52:39,1282528359,loopback/9664-a,CS_NEW,,,,,,,,,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,e50d768b-37b8-4cfb-ab46-60e7ac029c83 I also tested on a Linux server. It performs better. no detail data collected though. And, interesting, when I run a 30 * 20 bridge + 30 * 20 conference, loadavg suddenly grew to 2000+. But I can still run a "hupall" in fs_cli with so high load. Note: 1) loopback might not typical in test. 2) This is not a FS performance test, I only want the conclusion that 2-way conference uses more resource than bridged calls. On Thu, Aug 19, 2010 at 8:31 AM, Seven Du wrote: > Hi, > > Can someone explain the performance difference between bridged calls > and 2-party conference? or just in the code point of view? > > Since in some scenarios third party may join into a bridged call, so > we need to transfer a bridged call into a conference first. Make a > conference anyway event for 2-parties will make logic simpler and > clear. > > Thanks. > > -- > Blog: http://www.dujinfang.com > Proj:? http://www.freeswitch.org.cn > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From jeff at jefflenk.com Sun Aug 22 19:21:42 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 22 Aug 2010 19:21:42 -0700 (PDT) Subject: [Freeswitch-users] Phrase speak-text function return on first key press in phrase file on Windows In-Reply-To: References: Message-ID: <1282530102895-5451221.post@n2.nabble.com> Please open a jira on this -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Phrase-speak-text-function-return-on-first-key-press-in-phrase-file-on-Windows-tp5449059p5451221.html Sent from the freeswitch-users mailing list archive at Nabble.com. From woodydickson at gmail.com Sun Aug 22 20:34:43 2010 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 23 Aug 2010 11:34:43 +0800 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <20100822175311.GA63925@quark.hightek.org> References: <20100822175311.GA63925@quark.hightek.org> Message-ID: Hi Vince, I have played with running Freeswitch on BSD too but the result is not great. The reason seems to be because BSD's threading is not as efficient as the one in Linux or there may be some other ways to tune it. BSD does give a better pure UDP throughput performance by the way. So what I ended up doing is developing my own UDP implementation which enable media to move through the ethernet at raw wire speed. I am able to max out the ethernet card limitation on Linux platform as a result of that. Woody On Mon, Aug 23, 2010 at 1:53 AM, Vincent Stemen wrote: > On Fri, Aug 20, 2010 at 05:29:31PM +0800, Woody Dickson wrote: >> Hi, >> >> I am doing some experiments with Freeswitch by torturing it to see how >> the machine's CPU response to heavy loaded situation. >> The test is done on a 16 core 5550 dual quad core server running >> fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. >> >> What I found so strange was that while CPU usage remains pretty low >> and distributed among all cores at 190 - 200 calls per second. ?Then, >> after added a few more calls per second, all CPU becomes fully >> utilized. >> >> Is this due to some wrong setting? ?Any idea how I can tweak the >> configuration and continue my test? >> >> Thanks, >> Woody > > Hi Woody. > > I would hazard to guess that this could be a Linux resource management > issue. ?I don't have any experience with SMP on Linux, but Linux has > a long history of memory management (among other) problems. ?We ran > Linux exclusively on all our servers and workstations for over 10 years > before finally switching to BSD several years ago. ?We had continuous > problems ranging from minor strange unexplained behaviours, as you > describe, to what appeared to be bugs in applications, to outright > crashes and freezes of the whole OS every day. ?When we switched to BSD > nearly all the problems went away. ?Even some of the (what appeared to > be) bugs in Linux binary applications went way, going from Linux to BSD > running under Linux emulation (without re-compiling), using the same > Linux libraries on the same hardware. > > An interesting test would be to try the same load test with BSD on the > same machine and see if you get a similar result. > > Regards, > Vince > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Sun Aug 22 22:00:35 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 23 Aug 2010 13:00:35 +0800 Subject: [Freeswitch-users] centos5.5 ready for FreeSWITCH? Message-ID: Hi, I see bluebox based on CentOS 5.5. Before that it is highly recommended on 5.3 over 5.4. So, is 5.5 safe for production use now? Thanks. -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From adminjew at gmail.com Sun Aug 22 22:19:11 2010 From: adminjew at gmail.com (Yitzchok) Date: Mon, 23 Aug 2010 01:19:11 -0400 Subject: [Freeswitch-users] Phrase speak-text function return on first key press in phrase file on Windows In-Reply-To: <1282530102895-5451221.post@n2.nabble.com> References: <1282530102895-5451221.post@n2.nabble.com> Message-ID: http://jira.freeswitch.org/browse/MODAPP-448 Yitzchok On Sun, Aug 22, 2010 at 10:21 PM, Jeff Lenk wrote: > > Please open a jira on this > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Phrase-speak-text-function-return-on-first-key-press-in-phrase-file-on-Windows-tp5449059p5451221.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/714ebb82/attachment.html From ken at ukgb.net Mon Aug 23 00:57:11 2010 From: ken at ukgb.net (Ken Gillett) Date: Mon, 23 Aug 2010 08:57:11 +0100 Subject: [Freeswitch-users] Account selection Message-ID: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> If one wishes to have use of several VOIP 'lines', but with no PBX, you need to register all those SIP accounts with the client (softphone etc). You should then be informed which account is receiving a call and can pick a particular account from which to make calls. Once a PBX is in use, you can register the client as a single extension of the PBX and direct calls as appropriate to that extension - I assume with the correct caller ID and incoming account information passed to the recipient so they know as much as in the 'no PBX' configuration. But what about outgoing calls. In this scenario, registered as a single extension, how would it be possible to pick the outgoing 'line' (i.e. account) to use? Would it have to be done by dialling a prefix or is there another way? Is it client dependent? Ken G i l l e t t _/_/_/_/_/_/_/_/ From tayeb.meftah at gmail.com Tue Aug 24 00:33:35 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 24 Aug 2010 09:33:35 +0200 Subject: [Freeswitch-users] Account selection In-Reply-To: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> Message-ID: <4C7375CF.7070905@gmail.com> voip lines is server registered the server pick the line acording to your dialplan setting, client haven't anything to do with the voip lines Le 23/08/2010 09:57, Ken Gillett a ?crit : > If one wishes to have use of several VOIP 'lines', but with no PBX, you need to register all those SIP accounts with the client (softphone etc). You should then be informed which account is receiving a call and can pick a particular account from which to make calls. Once a PBX is in use, you can register the client as a single extension of the PBX and direct calls as appropriate to that extension - I assume with the correct caller ID and incoming account information passed to the recipient so they know as much as in the 'no PBX' configuration. > > But what about outgoing calls. In this scenario, registered as a single extension, how would it be possible to pick the outgoing 'line' (i.e. account) to use? Would it have to be done by dialling a prefix or is there another way? Is it client dependent? > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz From tayeb.meftah at gmail.com Tue Aug 24 00:34:48 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 24 Aug 2010 09:34:48 +0200 Subject: [Freeswitch-users] centos5.5 ready for FreeSWITCH? In-Reply-To: References: Message-ID: <4C737618.1040105@gmail.com> sevan, i am using it in my VPS without issus but mayb something wrong elsewhere with developers so you should confirm :P Le 23/08/2010 07:00, Seven Du a ?crit : > Hi, > > I see bluebox based on CentOS 5.5. Before that it is highly > recommended on 5.3 over 5.4. So, is 5.5 safe for production use now? > > Thanks. > > -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz From tayeb.meftah at gmail.com Tue Aug 24 00:49:21 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 24 Aug 2010 09:49:21 +0200 Subject: [Freeswitch-users] scalability of FS vs yate In-Reply-To: References: Message-ID: <4C737981.3060206@gmail.com> hi, fs is much stable fs is doing patch of the used library and sending it to the upstream project fs is maintaining there own patch of the library fs have very gradfull voicemail fs have big conferencing module that never dead or give bad audio quality fs have nice xml configuration base that's easy to understand/extand the dialplan of freeswitch couldn't by found elsehere fs have HD/WB/NB audio including polycom siren aka G.722.1, celt, silk, G.722, speex-WB freeswitch could host 1000000000 of subscribers if you twick your RDBMS and your hardware and you load balance it through ODBC or opensips/kamailio thanks Le 21/08/2010 02:30, jesse zhao a ?crit : > I am wondering whether someone did testing of FreeSwitch vs Yate under > the same hardware system and testing profiles. Which one is more > scalable? > > Any usage case of FS in large service providers or organization in > terms of 100K subscribers? It is nice to list customers of FS in the > wiki page. > > > thanks! > > -jesse > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb alg?rie t?l?com SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz From dujinfang at gmail.com Mon Aug 23 02:12:18 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 23 Aug 2010 17:12:18 +0800 Subject: [Freeswitch-users] repeat a submenu in ivr? Message-ID: Hi, >From http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr , it can repeat a main menu by menu-top, But, how to repeat a submenu? -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From asilva at wirelessmundi.com Mon Aug 23 02:32:10 2010 From: asilva at wirelessmundi.com (Antonio) Date: Mon, 23 Aug 2010 11:32:10 +0200 Subject: [Freeswitch-users] core.db table channels application field store wrong data when executing voicemail In-Reply-To: References: <1282205390.25391.41.camel@marces.tc.commsmundi.com> Message-ID: <1282555930.25391.48.camel@marces.tc.commsmundi.com> I see. I've a couple of questions related: Is it recommended to directly use core.db for monitoring purposes instead of developing a custom event handler? Could there be any sqlite "locking" issues if attacking that database in a read-only manner? In case it is possible, how can I identify what was the initial application that was called from the dialplan (and not the applications called by it). Is there an API method containing the history of applications executed on a specific channel given its uuid? Thanks, Ant?nio On Thu, 2010-08-19 at 09:20 -0500, Anthony Minessale wrote: > No, every app that is ever executed will change that field, some apps > in turn execute more apps. > > > On Thu, Aug 19, 2010 at 3:09 AM, Antonio wrote: > > > > Hi, > > > > during multiple selects in the table "channels" just realized that when > > a call is executing the app voicemail, to check for messages, during the > > navigation menu of the voicemail it changes to application "sleep". > > > > Is it normal? or is a bug for fs? > > > > > > The call: > > > > During the auth: > > > > sqlite> select * from channels; > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > sqlite> select * from channels; > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > sqlite> select * from channels; > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > sqlite> select * from channels; > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > sqlite> select * from channels; > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > sqlite> select * from channels; > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com 100|XML| > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > in the navigation menu: > > > > sqlite> select * from channels; > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| > > 101 at 192.168.10.25||EARLY|||| > > > > sqlite> select * from channels; > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| > > 101 at 192.168.10.25||EARLY|||| > > > > > > > > Thanks, > > Ant?nio > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- Un cordial saludo / Best regards, _________________________ Ant?nio Silva E-mail:asilva at wirelessmundi.com From steveayre at gmail.com Mon Aug 23 02:52:33 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 23 Aug 2010 10:52:33 +0100 Subject: [Freeswitch-users] core.db table channels application field store wrong data when executing voicemail In-Reply-To: <1282555930.25391.48.camel@marces.tc.commsmundi.com> References: <1282205390.25391.41.camel@marces.tc.commsmundi.com> <1282555930.25391.48.camel@marces.tc.commsmundi.com> Message-ID: On 23 August 2010 10:32, Antonio wrote: > I see. I've a couple of questions related: > > Is it recommended to directly use core.db for monitoring purposes > instead of developing a custom event handler? Could there be any sqlite > "locking" issues if attacking that database in a read-only manner? > I would use ESL, I believe that's probably the recommended way to do it. It's a pretty simple protocol so shouldn't be any harder to implement than something reading an sqlite file. You can either use events and work out the channel states yourself, or use api commands like 'show channels' etc (which reads the core db so are pretty much what you'd already be doing). A rwlock can't be written while a reader has it, so it will possibly slow the DB down if you're reading the database. Shouldn't be noticeable unless you're doing so intensively though. Another approach would be to move the core into odbc to use a database system that's more advanced than sqlite, which would also them allow you to replicate the data to another server so none of the monitoring occurs on the FS host. > > In case it is possible, how can I identify what was the initial > application that was called from the dialplan (and not the applications > called by it). Is there an API method containing the history of > applications executed on a specific channel given its uuid? > Do you need it in realtime or after the call? mod_xml_cdr contains the callflow, I don't know whether that can be accessed during a call though. ESL events will give you that information I believe. > > Thanks, > Ant?nio > > > On Thu, 2010-08-19 at 09:20 -0500, Anthony Minessale wrote: > > No, every app that is ever executed will change that field, some apps > > in turn execute more apps. > > > > > > On Thu, Aug 19, 2010 at 3:09 AM, Antonio > wrote: > > > > > > Hi, > > > > > > during multiple selects in the table "channels" just realized that when > > > a call is executing the app voicemail, to check for messages, during > the > > > navigation menu of the voicemail it changes to application "sleep". > > > > > > Is it normal? or is a bug for fs? > > > > > > > > > The call: > > > > > > During the auth: > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com commsmundi.com100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25||EARLY|||| > > > > > > in the navigation menu: > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > > 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| > > > 101 at 192.168.10.25||EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25|CS_EXECUTE|101|101| > > > 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU|8000||marces| > > > 101 at 192.168.10.25||EARLY|||| > > > > > > > > > > > > Thanks, > > > Ant?nio > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > -- > > Un cordial saludo / Best regards, > > _________________________ > > Ant?nio Silva > > E-mail:asilva at wirelessmundi.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/cfa394dc/attachment-0001.html From ali.stgt at gmail.com Mon Aug 23 03:36:43 2010 From: ali.stgt at gmail.com (=?UTF-8?B?RHVybXXFnyBBbGkgw5Z6dMO8cms=?=) Date: Mon, 23 Aug 2010 13:36:43 +0300 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session Message-ID: Hello, we had some trouble while executing a bulk call process with originating a parallel call of 200. Because in many cases, FreeSWITCH has notified hangups (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong hangup-cause. Instead of notifying the successful state, we got the NO_ANSWER hangup cause. BUSY and NO_ANSWER states are candidates for retries, therefor many numbers are called/payed twice. See log below. Some other questions: B) Is the originate_timeout value an overall timer or a timer for the ringing (starts if SIP code 180 incomes?) stage. C) We are originating each number in a separate thread and listen to the channel events for updating the call result. Should we change this implementation or is this a good scenario/standard way. Related to the call result, if it is busy or not answered, the call is retried after 30 min. What are the recommends on this side to be ensured, the correct hangup case be got and the number is not called twice.. D) What do I have to bear in mind for bulk calls with parallel calls over 200. Thanks for your answer in advance. Ali An extraction of the log (regard to the first issue): 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable string 0 = [sip_from_uri=sip:xxxx at xxxxx] 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable string 1 = [ignore_early_media=true] 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable string 2 = [sip_cid_type=none] 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable string 3 = [originate_timeout=40] 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8-227e99f3f0ef] 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_INIT 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/9055599XXXXX) State INIT 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 sofia/internal/9055599XXXXX SOFIA INIT 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/9055599XXXXX) State INIT going to sleep 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/9055599XXXXX) State ROUTING 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 sofia/internal/9055599XXXXX SOFIA ROUTING 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/9055599XXXXX) State ROUTING going to sleep 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/9055599XXXXX) State CONSUME_MEDIA 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [calling][0] 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX Update Callee ID to "9055599XXXXX" <9055599XXXXX> 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [proceeding][183] 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready sofia/internal/9055599XXXXX! 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [proceeding][183] 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send payload to 101 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 port 18620 codec: 8 ms: 20 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer [soft] 160 bytes per 20ms 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send payload to 101 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive payload to 101 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer sofia/internal/9055599XXXXX! 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel sofia/internal/9055599XXXXX skipping state [proceeding][180] 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [completing][200] 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal sofia/internal/9055599XXXXX [KILL] 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9055599XXXXX) State HANGUP 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to sofia/internal/9055599XXXXX 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9055599XXXXX) State HANGUP going to sleep 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/9055599XXXXX) State REPORTING 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/9055599XXXXX) State REPORTING going to sleep 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session 18770 (sofia/internal/9055599XXXXX) Locked, Waiting on external entities 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session 18770 (sofia/internal/9055599XXXXX) Ended 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close Channel sofia/internal/9055599XXXXX [CS_DESTROY] 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/9055599XXXXX) State DESTROY 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 sofia/internal/9055599XXXXX SOFIA DESTROY 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 sofia/internal/9055599XXXXX Standard DESTROY 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/9055599XXXXX) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/a5bc2b48/attachment.html From david.ponzone at ipeva.fr Mon Aug 23 03:46:19 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 12:46:19 +0200 Subject: [Freeswitch-users] Account selection In-Reply-To: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> Message-ID: Ken, I am not really sure to understand your issue/question. Can you describe exactly the equipements involved and what you want to do ? Is FS used as a PBX or a a provider softswitch to terminate the trunk coming form the PBX ? Some various information that could help you in the meantime: -most softphones can have several SIP accounts, but you should check that they can register all of them at the same time -if your objective is to have FS sending calls to a specific external VoIP account when it receives a call from a specific internal account, like this: phone1-----> FS-------> Provider SIP Account 1 phone2----->FS--------> Provider SIP Account 2 you would need to split the outgoing calls one way or another: you could do that based on the caller-id, or you may put your internal accounts in different contexts, so they use different dialplans. There are probably other ways, like using a prefix, but this one is probably a burden for the user and a security issue possibly. A such configuration is really some sort of SBC, when you want to avoid your SIP devices to connect to the accounts provided by your carrier directly, because you are concerned with security or because you want to keep control on the calls to provide more services to your users. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 09:57, Ken Gillett a ?crit : > If one wishes to have use of several VOIP 'lines', but with no PBX, > you need to register all those SIP accounts with the client > (softphone etc). You should then be informed which account is > receiving a call and can pick a particular account from which to > make calls. Once a PBX is in use, you can register the client as a > single extension of the PBX and direct calls as appropriate to that > extension - I assume with the correct caller ID and incoming account > information passed to the recipient so they know as much as in the > 'no PBX' configuration. > > But what about outgoing calls. In this scenario, registered as a > single extension, how would it be possible to pick the outgoing > 'line' (i.e. account) to use? Would it have to be done by dialling a > prefix or is there another way? Is it client dependent? > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/6f8861bb/attachment-0001.html From steveayre at gmail.com Mon Aug 23 03:53:33 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 23 Aug 2010 11:53:33 +0100 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: Do you have a sip trace for those calls? On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: > Hello, > > we had some trouble while executing a bulk call process with originating a > parallel call of 200. Because in many cases, FreeSWITCH has notified hangups > (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong hangup-cause. Instead > of notifying the successful state, we got the NO_ANSWER hangup cause. BUSY > and NO_ANSWER states are candidates for retries, therefor many numbers are > called/payed twice. See log below. > > Some other questions: > > B) Is the originate_timeout value an overall timer or a timer for the > ringing (starts if SIP code 180 incomes?) stage. > > C) We are originating each number in a separate thread and listen to the > channel events for updating the call result. Should we change this > implementation or is this a good scenario/standard way. Related to the call > result, if it is busy or not answered, the call is retried after 30 min. > What are the recommends on this side to be ensured, the correct hangup case > be got and the number is not called twice.. > > D) What do I have to bear in mind for bulk calls with parallel calls over > 200. > > Thanks for your answer in advance. > > Ali > > > An extraction of the log (regard to the first issue): > > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 0 = [sip_from_uri=sip:xxxx at xxxxx] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 1 = [ignore_early_media=true] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 2 = [sip_cid_type=none] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 3 = [originate_timeout=40] > 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel > sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8-227e99f3f0ef] > 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 > (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 > sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_INIT > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9055599XXXXX) State INIT > 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 > sofia/internal/9055599XXXXX SOFIA INIT > 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 > (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9055599XXXXX) State INIT going to sleep > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 > (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9055599XXXXX) State ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 > sofia/internal/9055599XXXXX SOFIA ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9055599XXXXX) State ROUTING going to sleep > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA > 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep > 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [calling][0] > 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX > Update Callee ID to "9055599XXXXX" <9055599XXXXX> > 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [proceeding][183] > 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready > sofia/internal/9055599XXXXX! > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [proceeding][183] > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: > 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec > sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send > payload to 101 > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP > [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 port > 18620 codec: 8 ms: 20 > 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer [soft] > 160 bytes per 20ms > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send > payload to 101 > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive > payload to 101 > 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer > sofia/internal/9055599XXXXX! > 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 > (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY > 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel > sofia/internal/9055599XXXXX skipping state [proceeding][180] > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [completing][200] > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 > (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP > 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup > sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal > sofia/internal/9055599XXXXX [KILL] > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9055599XXXXX) State HANGUP > 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to > sofia/internal/9055599XXXXX > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9055599XXXXX) State HANGUP going to sleep > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/9055599XXXXX) State REPORTING > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/9055599XXXXX) State REPORTING going to sleep > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session 18770 > (sofia/internal/9055599XXXXX) Locked, Waiting on external entities > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session > 18770 (sofia/internal/9055599XXXXX) Ended > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/internal/9055599XXXXX [CS_DESTROY] > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 > (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 > (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/9055599XXXXX) State DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 > sofia/internal/9055599XXXXX SOFIA DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/9055599XXXXX Standard DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/9055599XXXXX) State DESTROY going to sleep > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/86265ffb/attachment.html From neil.burgess at redmatter.com Mon Aug 23 04:21:12 2010 From: neil.burgess at redmatter.com (Neil Burgess) Date: Mon, 23 Aug 2010 12:21:12 +0100 Subject: [Freeswitch-users] Custom SIP Notify via proxy Message-ID: <787302A89ACCE24DA8F56DA101E77C842B3AE95095@THHS2E12BE1X.hostedservice2.net> Hello, Sent this post a couple of weeks back. Still trying to find a solution, so trying again and see if anyone has any thoughts! I am trying to send CUSTOM NOTIFY events to devices, using the ESL interface with a PHP script of the following form. I am having partial success, but coming up against problems when using with an outbound SIP proxy:- $e = new ESLevent("NOTIFY"); $e->addHeader("profile", "internal"); $e->addHeader("from", "1020"); $e->addHeader("to-uri", "sip:1019 at 192.168.2.3:5063;fs_path=sip:82.24.214.226:5063"); $e->addHeader("from-uri", "sip:1020 at pbx.rm.com"); $e->addHeader("host", "pbx.rm.com"); $e->addHeader("event-string", "check-sync"); $e->addHeader("content-type", "application/simple-message-summary"); $e->addBody("ok"); $res = $sockFSServerCommand->sendEvent($e); In our setup, we have several FreeSwitch boxes sitting behind two OpenSIPS load balancers. These Load Balancers handle registrations, etc. When we originate calls from FreeSwitch, we use an originate URL of the form (sofia/internal/1001 at sip.redmatter.com;fs_path=sip:84.45.30.2), and use fs_path to push the sip call out via the right load balancer/ proxy. This works perfectly. So, with the custom NOTIFY's I also need to make them follow the same path. As mentioned, we already have the contact string in our hand from the OpenSIPS registration server, but can't see how I can use that plus force the outbound path for the Notify. Was thinking I could use fs_path in the uri (as above) but doesn't seem to work, i.e. the wrong IP is used, and NOTIFY is messed up somewhat (see example below)! Maybe another combination I am missing! ------------------------ NOTIFY sip:1019 at 192.168.2.3:5063;fs_path=sip:82.24.214.22:5063 SIP/2.0 Via: SIP/2.0/UDP 212.85.24.245;rport;branch=z9hG4bKv268QHcy539yc Max-Forwards: 70 From: ;tag=0rer8BS2etmmc To: ;fs_path=sip:82.24.214.22:5063 Call-ID: 5a08d5c6-1a8e-122e-3885-0024e86c401a CSeq: 134921 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.5-20100211-0400-16602 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Event: check-sync Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=timeout Content-Type: application/simple-message-summary Content-Length: 2 ok ------------------ Any thoughts gratefully appreciated. Rgds, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/31b4e896/attachment-0001.html From mnhassan at usa.net Mon Aug 23 05:25:58 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 23 Aug 2010 18:25:58 +0600 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100822175311.GA63925@quark.hightek.org> Message-ID: Is this also the case for the recommended CentOS / RHEL? Do you still have to resort to having your own UDP implementation to max out Eth Card limitation? In the past, I have found a limitation on Linux, that the eth driver is single-threaded. So, I couldn't push beyond 50K pps on a Intel Quad E5504 HP machine through 1 ethernet card. Regards HASSAN On Mon, Aug 23, 2010 at 09:34, Woody Dickson wrote: > Hi Vince, > > I have played with running Freeswitch on BSD too but the result is not > great. The reason seems to be because BSD's threading is not as > efficient as the one in Linux or there may be some other ways to tune > it. BSD does give a better pure UDP throughput performance by the > way. > > So what I ended up doing is developing my own UDP implementation which > enable media to move through the ethernet at raw wire speed. I am > able to max out the ethernet card limitation on Linux platform as a > result of that. > > Woody > > > On Mon, Aug 23, 2010 at 1:53 AM, Vincent Stemen > wrote: > > On Fri, Aug 20, 2010 at 05:29:31PM +0800, Woody Dickson wrote: > >> Hi, > >> > >> I am doing some experiments with Freeswitch by torturing it to see how > >> the machine's CPU response to heavy loaded situation. > >> The test is done on a 16 core 5550 dual quad core server running > >> fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. > >> > >> What I found so strange was that while CPU usage remains pretty low > >> and distributed among all cores at 190 - 200 calls per second. Then, > >> after added a few more calls per second, all CPU becomes fully > >> utilized. > >> > >> Is this due to some wrong setting? Any idea how I can tweak the > >> configuration and continue my test? > >> > >> Thanks, > >> Woody > > > > Hi Woody. > > > > I would hazard to guess that this could be a Linux resource management > > issue. I don't have any experience with SMP on Linux, but Linux has > > a long history of memory management (among other) problems. We ran > > Linux exclusively on all our servers and workstations for over 10 years > > before finally switching to BSD several years ago. We had continuous > > problems ranging from minor strange unexplained behaviours, as you > > describe, to what appeared to be bugs in applications, to outright > > crashes and freezes of the whole OS every day. When we switched to BSD > > nearly all the problems went away. Even some of the (what appeared to > > be) bugs in Linux binary applications went way, going from Linux to BSD > > running under Linux emulation (without re-compiling), using the same > > Linux libraries on the same hardware. > > > > An interesting test would be to try the same load test with BSD on the > > same machine and see if you get a similar result. > > > > Regards, > > Vince > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/726f7237/attachment.html From brian at freeswitch.org Mon Aug 23 06:25:14 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 08:25:14 -0500 Subject: [Freeswitch-users] repeat a submenu in ivr? In-Reply-To: References: Message-ID: <7611EB9A-7658-4BFF-BCFB-700EF1C4E689@freeswitch.org> they repeat already if you don't select anything. /b On Aug 23, 2010, at 4:12 AM, Seven Du wrote: > > > > But, how to repeat a submenu? From ken at ukgb.net Mon Aug 23 06:26:49 2010 From: ken at ukgb.net (Ken Gillett) Date: Mon, 23 Aug 2010 14:26:49 +0100 Subject: [Freeswitch-users] Account selection In-Reply-To: References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> Message-ID: <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> Let's say there are 6 SIP accounts to which FreeSwitch is 'registered' and it can make calls using any of them. But which one is used? One of the local extensions (let's say a softphone) needs to receive calls from all of these accounts. This is simple enough and the recipient should see the incoming call information i.e. which account the call is to. But when that extension makes an outgoing call, how can it specify which is the SIP account used to by FS to make the call? This can be very important when each SIP account represents a different company/business. Although one person is dealing with all those businesses, when an outgoing call is made it is imperative that the correct SIP account is used to make that call so that the recipient is correctly informed who is making the call. Currently (no PBX), my softphone registers to each of these 6 accounts and I can choose which account to use to make a call. But if I am registered to FS as a single extension, how can I tell FS which account to use when I place an outgoing call? Is there any way to do this without having to use Dial plans? On 23 Aug 2010, at 11:46, David Ponzone wrote: > Ken, > > I am not really sure to understand your issue/question. > Can you describe exactly the equipements involved and what you want to do ? > Is FS used as a PBX or a a provider softswitch to terminate the trunk coming form the PBX ? > > Some various information that could help you in the meantime: > -most softphones can have several SIP accounts, but you should check that they can register all of them at the same time > -if your objective is to have FS sending calls to a specific external VoIP account when it receives a call from a specific internal account, like this: > phone1-----> FS-------> Provider SIP Account 1 > phone2----->FS--------> Provider SIP Account 2 > you would need to split the outgoing calls one way or another: you could do that based on the caller-id, or you may put your internal accounts in different contexts, so they use different dialplans. > There are probably other ways, like using a prefix, but this one is probably a burden for the user and a security issue possibly. > A such configuration is really some sort of SBC, when you want to avoid your SIP devices to connect to the accounts provided by your carrier directly, because you are concerned with security or because you want to keep control on the calls to provide more services to your users. > > Le 23/08/2010 ? 09:57, Ken Gillett a ?crit : > >> If one wishes to have use of several VOIP 'lines', but with no PBX, you need to register all those SIP accounts with the client (softphone etc). You should then be informed which account is receiving a call and can pick a particular account from which to make calls. Once a PBX is in use, you can register the client as a single extension of the PBX and direct calls as appropriate to that extension - I assume with the correct caller ID and incoming account information passed to the recipient so they know as much as in the 'no PBX' configuration. >> >> But what about outgoing calls. In this scenario, registered as a single extension, how would it be possible to pick the outgoing 'line' (i.e. account) to use? Would it have to be done by dialling a prefix or is there another way? Is it client dependent? Ken G i l l e t t _/_/_/_/_/_/_/_/ From woodydickson at gmail.com Mon Aug 23 06:44:04 2010 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 23 Aug 2010 21:44:04 +0800 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100822175311.GA63925@quark.hightek.org> Message-ID: Based on my experiment, that is still the case with CentOS. After using my own UDP, I was able to get 70K pps on 1 ethernet card on a Intel 5550. The limitation is that I don't have enough machine power to fire off enough calls to max it out. On Mon, Aug 23, 2010 at 8:25 PM, Nyamul Hassan wrote: > Is this also the case for the recommended CentOS / RHEL? ?Do you still have > to resort to having your own UDP implementation to max out Eth Card > limitation? > In the past, I have found a limitation on Linux, that the eth driver is > single-threaded. ?So, I couldn't push beyond 50K pps on a Intel Quad E5504 > HP machine through 1 ethernet card. > Regards > HASSAN > > On Mon, Aug 23, 2010 at 09:34, Woody Dickson wrote: >> >> Hi Vince, >> >> I have played with running Freeswitch on BSD too but the result is not >> great. ?The reason seems to be because BSD's threading is not as >> efficient as the one in Linux or there may be some other ways to tune >> it. ?BSD does give a better pure UDP throughput performance by the >> way. >> >> So what I ended up doing is developing my own UDP implementation which >> enable media to move through the ethernet at raw wire speed. ?I am >> able to max out the ethernet card limitation on Linux platform as a >> result of that. >> >> Woody >> >> >> On Mon, Aug 23, 2010 at 1:53 AM, Vincent Stemen >> wrote: >> > On Fri, Aug 20, 2010 at 05:29:31PM +0800, Woody Dickson wrote: >> >> Hi, >> >> >> >> I am doing some experiments with Freeswitch by torturing it to see how >> >> the machine's CPU response to heavy loaded situation. >> >> The test is done on a 16 core 5550 dual quad core server running >> >> fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. >> >> >> >> What I found so strange was that while CPU usage remains pretty low >> >> and distributed among all cores at 190 - 200 calls per second. ?Then, >> >> after added a few more calls per second, all CPU becomes fully >> >> utilized. >> >> >> >> Is this due to some wrong setting? ?Any idea how I can tweak the >> >> configuration and continue my test? >> >> >> >> Thanks, >> >> Woody >> > >> > Hi Woody. >> > >> > I would hazard to guess that this could be a Linux resource management >> > issue. ?I don't have any experience with SMP on Linux, but Linux has >> > a long history of memory management (among other) problems. ?We ran >> > Linux exclusively on all our servers and workstations for over 10 years >> > before finally switching to BSD several years ago. ?We had continuous >> > problems ranging from minor strange unexplained behaviours, as you >> > describe, to what appeared to be bugs in applications, to outright >> > crashes and freezes of the whole OS every day. ?When we switched to BSD >> > nearly all the problems went away. ?Even some of the (what appeared to >> > be) bugs in Linux binary applications went way, going from Linux to BSD >> > running under Linux emulation (without re-compiling), using the same >> > Linux libraries on the same hardware. >> > >> > An interesting test would be to try the same load test with BSD on the >> > same machine and see if you get a similar result. >> > >> > Regards, >> > Vince >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From kris at kriskinc.com Mon Aug 23 06:55:46 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 23 Aug 2010 09:55:46 -0400 Subject: [Freeswitch-users] freeswitch CPU usage Message-ID: Is this implementation open source? -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces at lists.freeswitch.org To: FreeSWITCH Users Help Sent: Mon Aug 23 09:44:04 2010 Subject: Re: [Freeswitch-users] freeswitch CPU usage Based on my experiment, that is still the case with CentOS. After using my own UDP, I was able to get 70K pps on 1 ethernet card on a Intel 5550. The limitation is that I don't have enough machine power to fire off enough calls to max it out. On Mon, Aug 23, 2010 at 8:25 PM, Nyamul Hassan wrote: > Is this also the case for the recommended CentOS / RHEL? Do you still > have > to resort to having your own UDP implementation to max out Eth Card > limitation? > In the past, I have found a limitation on Linux, that the eth driver is > single-threaded. So, I couldn't push beyond 50K pps on a Intel Quad E5504 > HP machine through 1 ethernet card. > Regards > HASSAN > > On Mon, Aug 23, 2010 at 09:34, Woody Dickson > wrote: >> >> Hi Vince, >> >> I have played with running Freeswitch on BSD too but the result is not >> great. The reason seems to be because BSD's threading is not as >> efficient as the one in Linux or there may be some other ways to tune >> it. BSD does give a better pure UDP throughput performance by the >> way. >> >> So what I ended up doing is developing my own UDP implementation which >> enable media to move through the ethernet at raw wire speed. I am >> able to max out the ethernet card limitation on Linux platform as a >> result of that. >> >> Woody >> >> >> On Mon, Aug 23, 2010 at 1:53 AM, Vincent Stemen >> wrote: >> > On Fri, Aug 20, 2010 at 05:29:31PM +0800, Woody Dickson wrote: >> >> Hi, >> >> >> >> I am doing some experiments with Freeswitch by torturing it to see how >> >> the machine's CPU response to heavy loaded situation. >> >> The test is done on a 16 core 5550 dual quad core server running >> >> fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. >> >> >> >> What I found so strange was that while CPU usage remains pretty low >> >> and distributed among all cores at 190 - 200 calls per second. Then, >> >> after added a few more calls per second, all CPU becomes fully >> >> utilized. >> >> >> >> Is this due to some wrong setting? Any idea how I can tweak the >> >> configuration and continue my test? >> >> >> >> Thanks, >> >> Woody >> > >> > Hi Woody. >> > >> > I would hazard to guess that this could be a Linux resource management >> > issue. I don't have any experience with SMP on Linux, but Linux has >> > a long history of memory management (among other) problems. We ran >> > Linux exclusively on all our servers and workstations for over 10 years >> > before finally switching to BSD several years ago. We had continuous >> > problems ranging from minor strange unexplained behaviours, as you >> > describe, to what appeared to be bugs in applications, to outright >> > crashes and freezes of the whole OS every day. When we switched to BSD >> > nearly all the problems went away. Even some of the (what appeared to >> > be) bugs in Linux binary applications went way, going from Linux to BSD >> > running under Linux emulation (without re-compiling), using the same >> > Linux libraries on the same hardware. >> > >> > An interesting test would be to try the same load test with BSD on the >> > same machine and see if you get a similar result. >> > >> > Regards, >> > Vince >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Mon Aug 23 06:59:58 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 23 Aug 2010 21:59:58 +0800 Subject: [Freeswitch-users] Account selection In-Reply-To: <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> Message-ID: On Mon, Aug 23, 2010 at 9:26 PM, Ken Gillett wrote: > Let's say there are 6 SIP accounts to which FreeSwitch is 'registered' and it can make calls using any of them. But which one is used? One of the local extensions (let's say a softphone) needs to receive calls from all of these accounts. This is simple enough and the recipient should see the incoming call information i.e. which account the call is to. But when that extension makes an outgoing call, how can it specify which is the SIP account used to by FS to make the call? > > This can be very important when each SIP account represents a different company/business. Although one person is dealing with all those businesses, when an outgoing call is made it is imperative that the correct SIP account is used to make that call so that the recipient is correctly informed who is making the call. > > Currently (no PBX), my softphone registers to each of these 6 accounts and I can choose which account to use to make a call. But if I am registered to FS as a single extension, how can I tell FS which account to use when I place an outgoing call? Is there any way to do this without having to use Dial plans? > > There is a way to do this if you use dialplans. How do you dial out from even one line if you don't use dialplans(in FS)? > On 23 Aug 2010, at 11:46, David Ponzone wrote: > >> Ken, >> >> I am not really sure to understand your issue/question. >> Can you describe exactly the equipements involved and what you want to do ? >> Is FS used as a PBX or a a provider softswitch to terminate the trunk coming form the PBX ? >> >> Some various information that could help you in the meantime: >> -most softphones can have several SIP accounts, but you should check that they can register all of them at the same time >> -if your objective is to have FS sending calls to a specific external VoIP account when it receives a call from a specific internal account, like this: >> phone1-----> FS-------> Provider SIP Account 1 >> phone2----->FS--------> Provider SIP Account 2 >> you would need to split the outgoing calls one way or another: you could do that based on the caller-id, or you may put your internal accounts in different contexts, so they use different dialplans. >> There are probably other ways, like using a prefix, but this one is probably a burden for the user and a security issue possibly. >> A such configuration is really some sort of SBC, when you want to avoid your SIP devices to connect to the accounts provided by your carrier directly, because you are concerned with security or because you want to keep control on the calls to provide more services to your users. >> >> Le 23/08/2010 ? 09:57, Ken Gillett a ?crit : >> >>> If one wishes to have use of several VOIP 'lines', but with no PBX, you need to register all those SIP accounts with the client (softphone etc). You should then be informed which account is receiving a call and can pick a particular account from which to make calls. Once a PBX is in use, you can register the client as a single extension of the PBX and direct calls as appropriate to that extension - I assume with the correct caller ID and incoming account information passed to the recipient so they know as much as in the 'no PBX' configuration. >>> >>> But what about outgoing calls. In this scenario, registered as a single extension, how would it be possible to pick the outgoing 'line' (i.e. account) to use? Would it have to be done by dialling a prefix or is there another way? Is it client dependent? > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From dujinfang at gmail.com Mon Aug 23 07:01:38 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 23 Aug 2010 22:01:38 +0800 Subject: [Freeswitch-users] repeat a submenu in ivr? In-Reply-To: <7611EB9A-7658-4BFF-BCFB-700EF1C4E689@freeswitch.org> References: <7611EB9A-7658-4BFF-BCFB-700EF1C4E689@freeswitch.org> Message-ID: Yes, perhaps that's the right way - if you forget the menu, just think(wait) a little while... :p On Mon, Aug 23, 2010 at 9:25 PM, Brian West wrote: > they repeat already if you don't select anything. > > /b > > On Aug 23, 2010, at 4:12 AM, Seven Du wrote: > >> >> ? ? >> >> But, how to repeat a submenu? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From david.ponzone at ipeva.fr Mon Aug 23 07:02:14 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 16:02:14 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100822175311.GA63925@quark.hightek.org> Message-ID: Woody, what codec do you use to push 70kpps on 1 GigE card ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 15:44, Woody Dickson a ?crit : > Based on my experiment, that is still the case with CentOS. After > using my own UDP, I was able to get 70K pps on 1 ethernet card on a > Intel 5550. The limitation is that I don't have enough machine power > to fire off enough calls to max it out. > > > On Mon, Aug 23, 2010 at 8:25 PM, Nyamul Hassan > wrote: >> Is this also the case for the recommended CentOS / RHEL? Do you >> still have >> to resort to having your own UDP implementation to max out Eth Card >> limitation? >> In the past, I have found a limitation on Linux, that the eth >> driver is >> single-threaded. So, I couldn't push beyond 50K pps on a Intel >> Quad E5504 >> HP machine through 1 ethernet card. >> Regards >> HASSAN >> >> On Mon, Aug 23, 2010 at 09:34, Woody Dickson >> wrote: >>> >>> Hi Vince, >>> >>> I have played with running Freeswitch on BSD too but the result is >>> not >>> great. The reason seems to be because BSD's threading is not as >>> efficient as the one in Linux or there may be some other ways to >>> tune >>> it. BSD does give a better pure UDP throughput performance by the >>> way. >>> >>> So what I ended up doing is developing my own UDP implementation >>> which >>> enable media to move through the ethernet at raw wire speed. I am >>> able to max out the ethernet card limitation on Linux platform as a >>> result of that. >>> >>> Woody >>> >>> >>> On Mon, Aug 23, 2010 at 1:53 AM, Vincent Stemen >>> wrote: >>>> On Fri, Aug 20, 2010 at 05:29:31PM +0800, Woody Dickson wrote: >>>>> Hi, >>>>> >>>>> I am doing some experiments with Freeswitch by torturing it to >>>>> see how >>>>> the machine's CPU response to heavy loaded situation. >>>>> The test is done on a 16 core 5550 dual quad core server running >>>>> fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. >>>>> >>>>> What I found so strange was that while CPU usage remains pretty >>>>> low >>>>> and distributed among all cores at 190 - 200 calls per second. >>>>> Then, >>>>> after added a few more calls per second, all CPU becomes fully >>>>> utilized. >>>>> >>>>> Is this due to some wrong setting? Any idea how I can tweak the >>>>> configuration and continue my test? >>>>> >>>>> Thanks, >>>>> Woody >>>> >>>> Hi Woody. >>>> >>>> I would hazard to guess that this could be a Linux resource >>>> management >>>> issue. I don't have any experience with SMP on Linux, but Linux >>>> has >>>> a long history of memory management (among other) problems. We ran >>>> Linux exclusively on all our servers and workstations for over 10 >>>> years >>>> before finally switching to BSD several years ago. We had >>>> continuous >>>> problems ranging from minor strange unexplained behaviours, as you >>>> describe, to what appeared to be bugs in applications, to outright >>>> crashes and freezes of the whole OS every day. When we switched >>>> to BSD >>>> nearly all the problems went away. Even some of the (what >>>> appeared to >>>> be) bugs in Linux binary applications went way, going from Linux >>>> to BSD >>>> running under Linux emulation (without re-compiling), using the >>>> same >>>> Linux libraries on the same hardware. >>>> >>>> An interesting test would be to try the same load test with BSD >>>> on the >>>> same machine and see if you get a similar result. >>>> >>>> Regards, >>>> Vince >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/54b69b2d/attachment-0001.html From david.ponzone at ipeva.fr Mon Aug 23 07:07:28 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 16:07:28 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <4C7120CD.9000200@kinetix.gr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> <4C706063.1040906@kinetix.gr> <4C7120CD.9000200@kinetix.gr> Message-ID: <5AB271EB-93C6-42E2-8B67-90FF7499B9B0@ipeva.fr> Vlasis, I don't see your results on the wiki page yet. I don't want to harass you but I am sure if some people start to fill it, others will follow. If we don't do that, I am afraid the table will still be empty in 1 year... I would gladly put my own results but they are far from interesting, as we don't push much traffic through FS yet. Com'on everyone! 5 minutes to add one line in the table! If you feel lazy or if you are wiki-proof, send me a private mail with the information, and I'll update it. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 22/08/2010 ? 15:06, Vlasis Hatzistavrou (KTI) a ?crit : > Nyamul Hassan wrote: >> Why? On list is fine. Apparently many would like to see the >> "performance" that others get from their implementation. >> > > Well, I thought too that many would like to see performance stats from > other users, > > but > > after all the discussion in this thread I drew the conclusion that an > on-list discussion about performance was actually _not_ fine. > > After all, you wrote that you understood the devs' irritation about > performance discussions and this is why I proposed to send you the > data > off list, since you are interested. > >> You could also contribute to the "wiki page", that David was >> referring to. >> > > I'll just proceed to post the data on the wiki page, then. > > Rgds, > Vlasis. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/6281b6c9/attachment.html From david.ponzone at ipeva.fr Mon Aug 23 07:13:33 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 16:13:33 +0200 Subject: [Freeswitch-users] Account selection In-Reply-To: <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> Message-ID: <2DD257A8-A956-45F1-9F3C-40A6EE3E77D3@ipeva.fr> There are 2 simple ways to do that: -use only one account on your softphone, and use prefixes: easy to implement, but easy to make mistake for the users too -use 6 accounts on your softphone as you used to do, with a 1-to-1 mapping with the external SIP accounts. You have to use a different FS context for each account in order to do that. If you find a nice softphone with programmable keys, you can even assign the keys so you have a line key per account. That would be the easier to use I think. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 15:26, Ken Gillett a ?crit : > Let's say there are 6 SIP accounts to which FreeSwitch is > 'registered' and it can make calls using any of them. But which one > is used? One of the local extensions (let's say a softphone) needs > to receive calls from all of these accounts. This is simple enough > and the recipient should see the incoming call information i.e. > which account the call is to. But when that extension makes an > outgoing call, how can it specify which is the SIP account used to > by FS to make the call? > > This can be very important when each SIP account represents a > different company/business. Although one person is dealing with all > those businesses, when an outgoing call is made it is imperative > that the correct SIP account is used to make that call so that the > recipient is correctly informed who is making the call. > > Currently (no PBX), my softphone registers to each of these 6 > accounts and I can choose which account to use to make a call. But > if I am registered to FS as a single extension, how can I tell FS > which account to use when I place an outgoing call? Is there any way > to do this without having to use Dial plans? > > > On 23 Aug 2010, at 11:46, David Ponzone wrote: > >> Ken, >> >> I am not really sure to understand your issue/question. >> Can you describe exactly the equipements involved and what you want >> to do ? >> Is FS used as a PBX or a a provider softswitch to terminate the >> trunk coming form the PBX ? >> >> Some various information that could help you in the meantime: >> -most softphones can have several SIP accounts, but you should >> check that they can register all of them at the same time >> -if your objective is to have FS sending calls to a specific >> external VoIP account when it receives a call from a specific >> internal account, like this: >> phone1-----> FS-------> Provider SIP Account 1 >> phone2----->FS--------> Provider SIP Account 2 >> you would need to split the outgoing calls one way or another: you >> could do that based on the caller-id, or you may put your internal >> accounts in different contexts, so they use different dialplans. >> There are probably other ways, like using a prefix, but this one is >> probably a burden for the user and a security issue possibly. >> A such configuration is really some sort of SBC, when you want to >> avoid your SIP devices to connect to the accounts provided by your >> carrier directly, because you are concerned with security or >> because you want to keep control on the calls to provide more >> services to your users. >> >> Le 23/08/2010 ? 09:57, Ken Gillett a ?crit : >> >>> If one wishes to have use of several VOIP 'lines', but with no >>> PBX, you need to register all those SIP accounts with the client >>> (softphone etc). You should then be informed which account is >>> receiving a call and can pick a particular account from which to >>> make calls. Once a PBX is in use, you can register the client as a >>> single extension of the PBX and direct calls as appropriate to >>> that extension - I assume with the correct caller ID and incoming >>> account information passed to the recipient so they know as much >>> as in the 'no PBX' configuration. >>> >>> But what about outgoing calls. In this scenario, registered as a >>> single extension, how would it be possible to pick the outgoing >>> 'line' (i.e. account) to use? Would it have to be done by dialling >>> a prefix or is there another way? Is it client dependent? > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/1b19b4f8/attachment-0001.html From woodydickson at gmail.com Mon Aug 23 07:14:11 2010 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 23 Aug 2010 22:14:11 +0800 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100822175311.GA63925@quark.hightek.org> Message-ID: I just used normal g711. This involve some code changes in the freeswitch core as well. I need to package it a bit better and get some more experiments done to fully qualify it. The problem with this kind of tests is getting enough hardware to generate this amount of calls is tricky. On Mon, Aug 23, 2010 at 10:02 PM, David Ponzone wrote: > Woody, > what codec do you use to push 70kpps on 1 GigE card ? > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 23/08/2010 ? 15:44, Woody Dickson a ?crit : > > Based on my experiment, that is still the case with CentOS. ?After > using my own UDP, I was able to get 70K pps on 1 ethernet card on a > Intel 5550. ?The limitation is that I don't have enough machine power > to fire off enough calls to max it out. > > > On Mon, Aug 23, 2010 at 8:25 PM, Nyamul Hassan wrote: > > Is this also the case for the recommended CentOS / RHEL? ?Do you still have > > to resort to having your own UDP implementation to max out Eth Card > > limitation? > > In the past, I have found a limitation on Linux, that the eth driver is > > single-threaded. ?So, I couldn't push beyond 50K pps on a Intel Quad E5504 > > HP machine through 1 ethernet card. > > Regards > > HASSAN > > On Mon, Aug 23, 2010 at 09:34, Woody Dickson wrote: > > Hi Vince, > > I have played with running Freeswitch on BSD too but the result is not > > great. ?The reason seems to be because BSD's threading is not as > > efficient as the one in Linux or there may be some other ways to tune > > it. ?BSD does give a better pure UDP throughput performance by the > > way. > > So what I ended up doing is developing my own UDP implementation which > > enable media to move through the ethernet at raw wire speed. ?I am > > able to max out the ethernet card limitation on Linux platform as a > > result of that. > > Woody > > > On Mon, Aug 23, 2010 at 1:53 AM, Vincent Stemen > > wrote: > > On Fri, Aug 20, 2010 at 05:29:31PM +0800, Woody Dickson wrote: > > Hi, > > I am doing some experiments with Freeswitch by torturing it to see how > > the machine's CPU response to heavy loaded situation. > > The test is done on a 16 core 5550 dual quad core server running > > fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. > > What I found so strange was that while CPU usage remains pretty low > > and distributed among all cores at 190 - 200 calls per second. ?Then, > > after added a few more calls per second, all CPU becomes fully > > utilized. > > Is this due to some wrong setting? ?Any idea how I can tweak the > > configuration and continue my test? > > Thanks, > > Woody > > Hi Woody. > > I would hazard to guess that this could be a Linux resource management > > issue. ?I don't have any experience with SMP on Linux, but Linux has > > a long history of memory management (among other) problems. ?We ran > > Linux exclusively on all our servers and workstations for over 10 years > > before finally switching to BSD several years ago. ?We had continuous > > problems ranging from minor strange unexplained behaviours, as you > > describe, to what appeared to be bugs in applications, to outright > > crashes and freezes of the whole OS every day. ?When we switched to BSD > > nearly all the problems went away. ?Even some of the (what appeared to > > be) bugs in Linux binary applications went way, going from Linux to BSD > > running under Linux emulation (without re-compiling), using the same > > Linux libraries on the same hardware. > > An interesting test would be to try the same load test with BSD on the > > same machine and see if you get a similar result. > > Regards, > > Vince > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Rong.Shen at oa.com.au Mon Aug 23 00:12:07 2010 From: Rong.Shen at oa.com.au (Rong Shen) Date: Mon, 23 Aug 2010 17:12:07 +1000 Subject: [Freeswitch-users] bypass media after fifo bridge Message-ID: <965759A53E43FE439E43565A7715E5F02F9FC7487B@oa-exchange1.oa.com.au> Hi, We've been using mod_fifo to connect calls for a while, it has all worked well except we are never able to bypass the media after a consumer and caller are connected. Enabling bypass_media_after_bridge in caller's dailplan doesn't seem to pass the variable to the point where the caller is popped out of the queue. We also tried to set bypass_media_after_bridge in the consumer's session, and use variable_fifo_consumer_caller_import to import it into the caller's session, but it didn't seem to have any effect - media is still going through the freeswitch (an event dump attached). Any idea how to set the bypass media option when using mod_fifo? Thanks. Regards, Rong -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: 1.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/bf6b7ff3/attachment-0001.txt From shamun.toha at gmail.com Sun Aug 22 02:40:53 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 22 Aug 2010 11:40:53 +0200 Subject: [Freeswitch-users] FreeSwitch - Skypopen cant make call Message-ID: Hi, When i am calling from another skypeID to FreeSwitch->Mod Skypopen account. I get this following details, No call is able to establish between Server and Regular skype caller?: freeswitch at example> sk list sk console is: |||interface1||| F ID Name IB (F/T) OB (F/T) State CallFlw UUID = ==== ======== ======= ======= ====== ============ ====== 1 [interface1] 0/0 0/0 IDLE CALL_IDLE Total Interfaces: 1 IB Calls(Failed/Total): 0/0 OB Calls(Failed/Total): 0/0 freeswitch at example> sk MESSAGE test1 hi there sk console is: |||interface1||| freeswitch at example> 2010-08-22 11:36:52.675088 [NOTICE] switch_channel.c:669 New Channel skypopen/interface1 [cbb3b872-add0-11df-a4c6-75d067ab3b3d] 2010-08-22 11:36:52.675088 [NOTICE] mod_skypopen.c:1944 Channel [skypopen/interface1] has been answered 2010-08-22 11:36:52.677551 [INFO] mod_dialplan_xml.c:418 Processing alu malu->5000 in context default 2010-08-22 11:36:52.687099 [NOTICE] mod_skypopen.c:1171 Hangup skypopen/interface1 [CS_EXECUTE] [NORMAL_CLEARING] 2010-08-22 11:36:52.750656 [NOTICE] switch_core_session.c:1182 Session 1 (skypopen/interface1) Ended 2010-08-22 11:36:52.750656 [NOTICE] switch_core_session.c:1184 Close Channel skypopen/interface1 [CS_DESTROY] 2010-08-22 11:36:52.987482 [ERR] skypopen_protocol.c:233 rev [(nil)|37 ][ERRORA 233 ][interface1][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER CALL: unable to alter input/output||| 2010-08-22 11:36:52.987482 [ERR] skypopen_protocol.c:235 rev [(nil)|37 ][ERRORA 235 ][interface1][-1, 0,16] skype_call now is DOWN 2010-08-22 11:37:18.045973 [NOTICE] switch_channel.c:669 New Channel skypopen/interface1 [dad30380-add0-11df-a4c7-75d067ab3b3d] 2010-08-22 11:37:18.047140 [NOTICE] mod_skypopen.c:1944 Channel [skypopen/interface1] has been answered 2010-08-22 11:37:18.048145 [INFO] mod_dialplan_xml.c:418 Processing alu malu->5000 in context default 2010-08-22 11:37:18.058690 [NOTICE] mod_skypopen.c:1171 Hangup skypopen/interface1 [CS_EXECUTE] [NORMAL_CLEARING] 2010-08-22 11:37:18.072829 [NOTICE] switch_core_session.c:1182 Session 2 (skypopen/interface1) Ended 2010-08-22 11:37:18.072829 [NOTICE] switch_core_session.c:1184 Close Channel skypopen/interface1 [CS_DESTROY] 2010-08-22 11:37:18.359355 [ERR] skypopen_protocol.c:233 rev [(nil)|37 ][ERRORA 233 ][interface1][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER CALL: unable to alter input/output||| 2010-08-22 11:37:18.359355 [ERR] skypopen_protocol.c:235 rev [(nil)|37 ][ERRORA 235 ][interface1][-1, 0,16] skype_call now is DOWN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/8a717933/attachment.html From shamun.toha at gmail.com Sun Aug 22 07:06:30 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Sun, 22 Aug 2010 16:06:30 +0200 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 Message-ID: Hi, Which skype to use? Isn't that 2.0.72 version ancient skype? IF 2.0.72 to be used, where can we download this for Fedora/CentOS? Any idea guys!!! Which Skype Client to use on Linux Use the *static* build of the *stable* Skype client (2.0.72). *Don't* use the build for your distro, neither the 'dynamic build'. *Don't* use the *beta* Skype client or more recent "stable" (2.0.72 is the one you want to use) if you want to have multiple channels with the same skypename. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100822/22925b86/attachment.html From ali.stgt at gmail.com Mon Aug 23 07:24:55 2010 From: ali.stgt at gmail.com (=?UTF-8?B?RHVybXXFnyBBbGkgw5Z6dMO8cms=?=) Date: Mon, 23 Aug 2010 17:24:55 +0300 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session Message-ID: I'm afraid, sip trace wasnt active. ---------- Weitergeleitete Nachricht ---------- > From: Steven Ayre > To: FreeSWITCH Users Help > Date: Mon, 23 Aug 2010 11:53:33 +0100 > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > answered session > Do you have a sip trace for those calls? > > > On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: > >> Hello, >> >> we had some trouble while executing a bulk call process with originating a >> parallel call of 200. Because in many cases, FreeSWITCH has notified hangups >> (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong hangup-cause. Instead >> of notifying the successful state, we got the NO_ANSWER hangup cause. BUSY >> and NO_ANSWER states are candidates for retries, therefor many numbers are >> called/payed twice. See log below. >> >> Some other questions: >> >> B) Is the originate_timeout value an overall timer or a timer for the >> ringing (starts if SIP code 180 incomes?) stage. >> >> C) We are originating each number in a separate thread and listen to the >> channel events for updating the call result. Should we change this >> implementation or is this a good scenario/standard way. Related to the call >> result, if it is busy or not answered, the call is retried after 30 min. >> What are the recommends on this side to be ensured, the correct hangup case >> be got and the number is not called twice.. >> >> D) What do I have to bear in mind for bulk calls with parallel calls over >> 200. >> >> Thanks for your answer in advance. >> >> Ali >> >> >> An extraction of the log (regard to the first issue): >> >> 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> string 0 = [sip_from_uri=sip:xxxx at xxxxx] >> 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> string 1 = [ignore_early_media=true] >> 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> string 2 = [sip_cid_type=none] >> 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> string 3 = [originate_timeout=40] >> 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel >> sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8-227e99f3f0ef] >> 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 >> (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT >> 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 >> sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 >> 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_INIT >> 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/9055599XXXXX) State INIT >> 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 >> sofia/internal/9055599XXXXX SOFIA INIT >> 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 >> (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING >> 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/9055599XXXXX) State INIT going to sleep >> 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING >> 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 >> (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/9055599XXXXX) State ROUTING >> 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 >> sofia/internal/9055599XXXXX SOFIA ROUTING >> 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 >> (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/9055599XXXXX) State ROUTING going to sleep >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 >> (sofia/internal/9055599XXXXX) State CONSUME_MEDIA >> 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 >> (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep >> 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel >> sofia/internal/9055599XXXXX entering state [calling][0] >> 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX >> Update Callee ID to "9055599XXXXX" <9055599XXXXX> >> 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel >> sofia/internal/9055599XXXXX entering state [proceeding][183] >> 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready >> sofia/internal/9055599XXXXX! >> 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel >> sofia/internal/9055599XXXXX entering state [proceeding][183] >> 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: >> 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec >> sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples >> 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send >> payload to 101 >> 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP >> [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 port >> 18620 codec: 8 ms: 20 >> 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer [soft] >> 160 bytes per 20ms >> 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send >> payload to 101 >> 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive >> payload to 101 >> 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer >> sofia/internal/9055599XXXXX! >> 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 >> (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY >> 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel >> sofia/internal/9055599XXXXX skipping state [proceeding][180] >> 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel >> sofia/internal/9055599XXXXX entering state [completing][200] >> 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP >> 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 >> (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP >> 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup >> sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] >> 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal >> sofia/internal/9055599XXXXX [KILL] >> 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP >> 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 >> (sofia/internal/9055599XXXXX) State HANGUP >> 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel >> sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER >> 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to >> sofia/internal/9055599XXXXX >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 >> (sofia/internal/9055599XXXXX) State HANGUP going to sleep >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING >> 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 >> (sofia/internal/9055599XXXXX) State REPORTING >> 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER >> 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 >> (sofia/internal/9055599XXXXX) State REPORTING going to sleep >> 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY >> 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session >> 18770 (sofia/internal/9055599XXXXX) Locked, Waiting on external entities >> 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session >> 18770 (sofia/internal/9055599XXXXX) Ended >> 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close >> Channel sofia/internal/9055599XXXXX [CS_DESTROY] >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 >> (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 >> (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/9055599XXXXX) State DESTROY >> 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 >> sofia/internal/9055599XXXXX SOFIA DESTROY >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/9055599XXXXX Standard DESTROY >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/9055599XXXXX) State DESTROY going to sleep >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/92ed5b7f/attachment-0001.html From vhatz at kinetix.gr Mon Aug 23 07:26:59 2010 From: vhatz at kinetix.gr (Vlasis Hatzistavrou) Date: Mon, 23 Aug 2010 17:26:59 +0300 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <5AB271EB-93C6-42E2-8B67-90FF7499B9B0@ipeva.fr> References: <4C6EAB1F.9060904@kinetix.gr> <4C6EFEF6.2000106@kinetix.gr> <4C6F1238.6000109@kinetix.gr> <4C6FC969.6020801@kinetix.gr> <20100821154007.7a45cd51@anubis.defcon1> <4C6FE993.3090102@kinetix.gr> <1BB4458C-0863-4A25-BC1B-653394B81D7B@ipeva.fr> <4C706063.1040906@kinetix.gr> <4C7120CD.9000200@kinetix.gr> <5AB271EB-93C6-42E2-8B67-90FF7499B9B0@ipeva.fr> Message-ID: <4C728533.8070901@kinetix.gr> Hello David, Apologies for the delay, it was a very busy day at work, it always on Mondays. I'll post the results asap, as I'll have to do some digging around in old files first. Best regards, Vlasis. On 23/8/2010 5:07 ??, David Ponzone wrote: > Vlasis, > > I don't see your results on the wiki page yet. > I don't want to harass you but I am sure if some people start to fill > it, others will follow. > If we don't do that, I am afraid the table will still be empty in 1 > year... > > I would gladly put my own results but they are far from interesting, > as we don't push much traffic through FS yet. > > Com'on everyone! 5 minutes to add one line in the table! > If you feel lazy or if you are wiki-proof, send me a private mail with > the information, and I'll update it. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > > /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > ? l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion non autoris?e est interdite. Tout message ?lectronique est > susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > n'?tes pas destinataire de ce message, merci de le d?truire > imm?diatement et d'avertir l'exp?diteur./ > / > / > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/220d4733/attachment.html From brian at freeswitch.org Mon Aug 23 07:29:32 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 09:29:32 -0500 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: Without that we can't help. You also need to make sure you're on the very latest code. The one thing I'm sure we get right are the hangup causes unless you're doing something to override them. /b On Aug 23, 2010, at 9:24 AM, Durmu? Ali ?zt?rk wrote: > I'm afraid, sip trace wasnt active. From david.ponzone at ipeva.fr Mon Aug 23 07:38:43 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 16:38:43 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100822175311.GA63925@quark.hightek.org> Message-ID: <847FEB94-D341-4F00-A0A4-B5EA26181220@ipeva.fr> So the throughput on the card topped at around 150Mbps, if I calculate correctly ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 16:14, Woody Dickson a ?crit : > I just used normal g711. > > This involve some code changes in the freeswitch core as well. I need > to package it a bit better and get some more experiments done to fully > qualify it. > The problem with this kind of tests is getting enough hardware to > generate this amount of calls is tricky. > > > > > > On Mon, Aug 23, 2010 at 10:02 PM, David Ponzone > wrote: >> Woody, >> what codec do you use to push 70kpps on 1 GigE card ? >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou >> diffusion >> non autoris?e est interdite. Tout message ?lectronique est >> susceptible >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce >> message s'il >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire >> de ce >> message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur. >> >> >> >> Le 23/08/2010 ? 15:44, Woody Dickson a ?crit : >> >> Based on my experiment, that is still the case with CentOS. After >> using my own UDP, I was able to get 70K pps on 1 ethernet card on a >> Intel 5550. The limitation is that I don't have enough machine power >> to fire off enough calls to max it out. >> >> >> On Mon, Aug 23, 2010 at 8:25 PM, Nyamul Hassan >> wrote: >> >> Is this also the case for the recommended CentOS / RHEL? Do you >> still have >> >> to resort to having your own UDP implementation to max out Eth Card >> >> limitation? >> >> In the past, I have found a limitation on Linux, that the eth >> driver is >> >> single-threaded. So, I couldn't push beyond 50K pps on a Intel >> Quad E5504 >> >> HP machine through 1 ethernet card. >> >> Regards >> >> HASSAN >> >> On Mon, Aug 23, 2010 at 09:34, Woody Dickson >> wrote: >> >> Hi Vince, >> >> I have played with running Freeswitch on BSD too but the result is >> not >> >> great. The reason seems to be because BSD's threading is not as >> >> efficient as the one in Linux or there may be some other ways to tune >> >> it. BSD does give a better pure UDP throughput performance by the >> >> way. >> >> So what I ended up doing is developing my own UDP implementation >> which >> >> enable media to move through the ethernet at raw wire speed. I am >> >> able to max out the ethernet card limitation on Linux platform as a >> >> result of that. >> >> Woody >> >> >> On Mon, Aug 23, 2010 at 1:53 AM, Vincent Stemen >> >> wrote: >> >> On Fri, Aug 20, 2010 at 05:29:31PM +0800, Woody Dickson wrote: >> >> Hi, >> >> I am doing some experiments with Freeswitch by torturing it to see >> how >> >> the machine's CPU response to heavy loaded situation. >> >> The test is done on a 16 core 5550 dual quad core server running >> >> fedora 2.6.30.10-105.2.23.fc11.x86_64 OS. >> >> What I found so strange was that while CPU usage remains pretty low >> >> and distributed among all cores at 190 - 200 calls per second. Then, >> >> after added a few more calls per second, all CPU becomes fully >> >> utilized. >> >> Is this due to some wrong setting? Any idea how I can tweak the >> >> configuration and continue my test? >> >> Thanks, >> >> Woody >> >> Hi Woody. >> >> I would hazard to guess that this could be a Linux resource >> management >> >> issue. I don't have any experience with SMP on Linux, but Linux has >> >> a long history of memory management (among other) problems. We ran >> >> Linux exclusively on all our servers and workstations for over 10 >> years >> >> before finally switching to BSD several years ago. We had continuous >> >> problems ranging from minor strange unexplained behaviours, as you >> >> describe, to what appeared to be bugs in applications, to outright >> >> crashes and freezes of the whole OS every day. When we switched to >> BSD >> >> nearly all the problems went away. Even some of the (what appeared >> to >> >> be) bugs in Linux binary applications went way, going from Linux to >> BSD >> >> running under Linux emulation (without re-compiling), using the same >> >> Linux libraries on the same hardware. >> >> An interesting test would be to try the same load test with BSD on >> the >> >> same machine and see if you get a similar result. >> >> Regards, >> >> Vince >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/ff8bffc7/attachment-0001.html From brian at freeswitch.org Mon Aug 23 07:43:08 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 09:43:08 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <847FEB94-D341-4F00-A0A4-B5EA26181220@ipeva.fr> References: <20100822175311.GA63925@quark.hightek.org> <847FEB94-D341-4F00-A0A4-B5EA26181220@ipeva.fr> Message-ID: Chances are you're not hitting the bandwidth limits but the fact that you're moving tiny packets around. Smaller packets == more context switches == less throughput. You can increase your packet size to 60ms and gain performance. /b On Aug 23, 2010, at 9:38 AM, David Ponzone wrote: > So the throughput on the card topped at around 150Mbps, if I calculate correctly ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > From mnhassan at usa.net Mon Aug 23 07:47:26 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Mon, 23 Aug 2010 20:47:26 +0600 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100822175311.GA63925@quark.hightek.org> <847FEB94-D341-4F00-A0A4-B5EA26181220@ipeva.fr> Message-ID: David, In our experience, it is not about "bps", but about "pps". Because the ethernet driver in Linux is single threaded, you cannot push beyond what a single core can handle. In our case, after 60kpps (on Intel X5504), the quality deteriorated. In Woody's case, this limit goes as high as 70kpps (on Intel X5550), which is roughly equivalent to the single core clock speed difference between the servers. Brian is speaking the exact same jargon, that I came across when I was reading about this 1 year ago. Regards HASSAN 2010/8/23 Brian West > Chances are you're not hitting the bandwidth limits but the fact that > you're moving tiny packets around. Smaller packets == more context switches > == less throughput. You can increase your packet size to 60ms and gain > performance. > > /b > > On Aug 23, 2010, at 9:38 AM, David Ponzone wrote: > > > So the throughput on the card topped at around 150Mbps, if I calculate > correctly ? > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/1dcccbef/attachment.html From ali.stgt at gmail.com Mon Aug 23 08:08:35 2010 From: ali.stgt at gmail.com (=?UTF-8?B?RHVybXXFnyBBbGkgw5Z6dMO8cms=?=) Date: Mon, 23 Aug 2010 18:08:35 +0300 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session Message-ID: I have setup freeswitch by using git and followed the instructions on the wiki page. How can I retrieve the exact git version? There is no override of the hangup cause in our code. The algorithm is very simple; we call the originate function with the playback action and use a wav file as argument. Hangup is done automatically by FreeSWITCH, after eof of the wav file is reached. Or the other part hangs up before the file ends. We dont traced the communication but in other hand, the sip codes are traced out into the log file (183-->180-->200=call established). This show to me, that there is no problem with the SIP transactions. Please assume, that SIP messages are OK. What else could be happen? 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable string 0 = [sip_from_uri=sip:xxxx at xxxxx] 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable string 1 = [ignore_early_media=true] 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable string 2 = [sip_cid_type=none] 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable string 3 = [originate_timeout=40] 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8- 227e99f3f0ef] 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_INIT 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/9055599XXXXX) State INIT 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 sofia/internal/9055599XXXXX SOFIA INIT 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/9055599XXXXX) State INIT going to sleep 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/9055599XXXXX) State ROUTING 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 sofia/internal/9055599XXXXX SOFIA ROUTING 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/9055599XXXXX) State ROUTING going to sleep 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/9055599XXXXX) State CONSUME_MEDIA 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [calling][0] 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX Update Callee ID to "9055599XXXXX" <9055599XXXXX> 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [proceeding][183] 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready sofia/internal/9055599XXXXX! 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [proceeding][183] 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send payload to 101 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 port 18620 codec: 8 ms: 20 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer [soft] 160 bytes per 20ms 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send payload to 101 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive payload to 101 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer sofia/internal/9055599XXXXX! 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel sofia/internal/9055599XXXXX skipping state [proceeding][180] 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [completing][200] 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal sofia/internal/9055599XXXXX [KILL] 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9055599XXXXX) State HANGUP 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to sofia/internal/9055599XXXXX 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/9055599XXXXX) State HANGUP going to sleep 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/9055599XXXXX) State REPORTING 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 (sofia/internal/9055599XXXXX) State REPORTING going to sleep 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal sofia/internal/9055599XXXXX [BREAK] 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session 18770 (sofia/internal/9055599XXXXX) Locked, Waiting on external entities 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session 18770 (sofia/internal/9055599XXXXX) Ended 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close Channel sofia/internal/9055599XXXXX [CS_DESTROY] 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/9055599XXXXX) State DESTROY 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 sofia/internal/9055599XXXXX SOFIA DESTROY 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 sofia/internal/9055599XXXXX Standard DESTROY 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/9055599XXXXX) State DESTROY going to sleep ---------- Weitergeleitete Nachricht ---------- > From: Brian West > To: FreeSWITCH Users Help > Date: Mon, 23 Aug 2010 09:29:32 -0500 > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > answered session > Without that we can't help. You also need to make sure you're on the very > latest code. The one thing I'm sure we get right are the hangup causes > unless you're doing something to override them. > > /b > > On Aug 23, 2010, at 9:24 AM, Durmu? Ali ?zt?rk wrote: > > > I'm afraid, sip trace wasnt active. > > ---------- Weitergeleitete Nachricht ---------- > From: Steven Ayre > To: FreeSWITCH Users Help > Date: Mon, 23 Aug 2010 11:53:33 +0100 > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > answered session > Do you have a sip trace for those calls? > > > On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: > Hello, > > we had some trouble while executing a bulk call process with originating a > parallel call of 200. Because in many cases, FreeSWITCH has notified hangups > (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong hangup-cause. Instead > of notifying the successful state, we got the NO_ANSWER hangup cause. BUSY > and NO_ANSWER states are candidates for retries, therefor many numbers are > called/payed twice. See log below. > > Some other questions: > > B) Is the originate_timeout value an overall timer or a timer for the > ringing (starts if SIP code 180 incomes?) stage. > > C) We are originating each number in a separate thread and listen to the > channel events for updating the call result. Should we change this > implementation or is this a good scenario/standard way. Related to the call > result, if it is busy or not answered, the call is retried after 30 min. > What are the recommends on this side to be ensured, the correct hangup case > be got and the number is not called twice.. > > D) What do I have to bear in mind for bulk calls with parallel calls over > 200. > > Thanks for your answer in advance. > > Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/9b81d82f/attachment-0001.html From brian at freeswitch.org Mon Aug 23 08:17:18 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 10:17:18 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100822175311.GA63925@quark.hightek.org> <847FEB94-D341-4F00-A0A4-B5EA26181220@ipeva.fr> Message-ID: <31C26F84-23F2-4878-A68A-7DB231ABB54B@freeswitch.org> I think you explained it a bit more in detail... we are saying exactly the same thing just I'm a bit more brief. /b On Aug 23, 2010, at 9:47 AM, Nyamul Hassan wrote: > David, > > In our experience, it is not about "bps", but about "pps". Because the ethernet driver in Linux is single threaded, you cannot push beyond what a single core can handle. > > In our case, after 60kpps (on Intel X5504), the quality deteriorated. In Woody's case, this limit goes as high as 70kpps (on Intel X5550), which is roughly equivalent to the single core clock speed difference between the servers. > > Brian is speaking the exact same jargon, that I came across when I was reading about this 1 year ago. > > Regards > HASSAN > > > > 2010/8/23 Brian West > Chances are you're not hitting the bandwidth limits but the fact that you're moving tiny packets around. Smaller packets == more context switches == less throughput. You can increase your packet size to 60ms and gain performance. > > /b From testeador01 at gmail.com Mon Aug 23 08:17:37 2010 From: testeador01 at gmail.com (Milena) Date: Mon, 23 Aug 2010 10:17:37 -0500 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: type "version" on the CLI If the "sip trace wasnt active" then activate it and pastebin it, that's the only way you can get help with this, you know the developers are experts not soothsayers :) -Mile 2010/8/23 Durmu? Ali ?zt?rk > I have setup freeswitch by using git and followed the instructions on the > wiki page. How can I retrieve the exact git version? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/d39e032e/attachment.html From fs-list at communicatefreely.net Mon Aug 23 08:18:22 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 23 Aug 2010 11:18:22 -0400 Subject: [Freeswitch-users] french language implementation Message-ID: <4C72913E.50308@communicatefreely.net> Hello, I would like to make a proper bilingual PBX implementation here in Canada, and it looks like there isn't a stock set of Canadian French sound files. I talked to GM Voices, and they have some voice talent available to do this job. Our company is willing to pay for it (they gave us a pretty good deal), and we will contribute the prompt set back to the project once it's done. Being that it's a different language, and some numbers and phrases are said in a different format, what do I need to change to make all this work? I imagine I have to modify some phrase macros somewhere. Does anyone have some instructions, or is there someone fluent in French willing to take this on? I see references to fr in modules.conf, but I can't find the audio files. Is it possible that someone has already done this work, and all we have to do is record the prompts? Any help is appreciated. -Tim From brian at freeswitch.org Mon Aug 23 08:18:30 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 10:18:30 -0500 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: <4AE83B09-04BC-42B7-A40B-27613C83012C@freeswitch.org> Again without a SIP trace I can't say... You have to remember that FreeSWITCH is a B2BUA and a call thru it results in TWO legs. If you were to ANSWER the first but then bridge to the second then it didn't answer you might still get a NO_ANSWER even thou you see the 200ok for the one leg of the call. /b On Aug 23, 2010, at 10:08 AM, Durmu? Ali ?zt?rk wrote: > I have setup freeswitch by using git and followed the instructions on the wiki page. How can I retrieve the exact git version? > > There is no override of the hangup cause in our code. The algorithm is very simple; we call the originate function with the playback action and use a wav file as argument. Hangup is done automatically by FreeSWITCH, after eof of the wav file is reached. Or the other part hangs up before the file ends. > > We dont traced the communication but in other hand, the sip codes are traced out into the log file (183-->180-->200=call established). This show to me, that there is no problem with the SIP transactions. Please assume, that SIP messages are OK. > > What else could be happen? From sos at sokhapkin.dyndns.org Mon Aug 23 08:22:37 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 23 Aug 2010 11:22:37 -0400 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: <201008231122.37692.sos@sokhapkin.dyndns.org> You're running concurrent calls, hangup cause NO_ANSWER and log line with 200 OK belong to different calls. On Monday 23 August 2010, Durmu? Ali ?zt?rk wrote: > I have setup freeswitch by using git and followed the instructions on the > wiki page. How can I retrieve the exact git version? > > There is no override of the hangup cause in our code. The algorithm is very > simple; we call the originate function with the playback action and use a > wav file as argument. Hangup is done automatically by FreeSWITCH, after eof > of the wav file is reached. Or the other part hangs up before the file > ends. > > We dont traced the communication but in other hand, the sip codes are > traced out into the log file (183-->180-->200=call established). This show > to me, that there is no problem with the SIP transactions. Please assume, > that SIP messages are OK. > > What else could be happen? > > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 0 = [sip_from_uri=sip:xxxx at xxxxx] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 1 = [ignore_early_media=true] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 2 = [sip_cid_type=none] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 3 = [originate_timeout=40] > 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel > sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8- > 227e99f3f0ef] > 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 > (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 > sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_INIT > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9055599XXXXX) State INIT > 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 > sofia/internal/9055599XXXXX SOFIA INIT > 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 > (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9055599XXXXX) State INIT going to sleep > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 > (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9055599XXXXX) State ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 > sofia/internal/9055599XXXXX SOFIA ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9055599XXXXX) State ROUTING going to sleep > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA > 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep > 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [calling][0] > 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX > Update Callee ID to "9055599XXXXX" <9055599XXXXX> > 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [proceeding][183] > 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready > sofia/internal/9055599XXXXX! > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [proceeding][183] > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: > 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec > sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send > payload to 101 > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP > [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 > port 18620 codec: 8 ms: 20 > 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer [soft] > 160 bytes per 20ms > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send > payload to 101 > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive > payload to 101 > 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer > sofia/internal/9055599XXXXX! > 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 > (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY > 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel > sofia/internal/9055599XXXXX skipping state [proceeding][180] > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [completing][200] > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 > (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP > 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup > sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal > sofia/internal/9055599XXXXX [KILL] > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9055599XXXXX) State HANGUP > 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to > sofia/internal/9055599XXXXX > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9055599XXXXX) State HANGUP going to sleep > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/9055599XXXXX) State REPORTING > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/9055599XXXXX) State REPORTING going to sleep > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session 18770 > (sofia/internal/9055599XXXXX) Locked, Waiting on external entities > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session > 18770 (sofia/internal/9055599XXXXX) Ended > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/internal/9055599XXXXX [CS_DESTROY] > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 > (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 > (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/9055599XXXXX) State DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 > sofia/internal/9055599XXXXX SOFIA DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/9055599XXXXX Standard DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/9055599XXXXX) State DESTROY going to sleep > > > > > ---------- Weitergeleitete Nachricht ---------- > > > From: Brian West > > To: FreeSWITCH Users Help > > Date: Mon, 23 Aug 2010 09:29:32 -0500 > > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > > answered session > > Without that we can't help. You also need to make sure you're on the > > very latest code. The one thing I'm sure we get right are the hangup > > causes unless you're doing something to override them. > > > > /b > > > > On Aug 23, 2010, at 9:24 AM, Durmu? Ali ?zt?rk wrote: > > > I'm afraid, sip trace wasnt active. > > > > ---------- Weitergeleitete Nachricht ---------- > > From: Steven Ayre > > To: FreeSWITCH Users Help > > Date: Mon, 23 Aug 2010 11:53:33 +0100 > > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > > answered session > > Do you have a sip trace for those calls? > > > > > > On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: > > Hello, > > > > we had some trouble while executing a bulk call process with originating > > a parallel call of 200. Because in many cases, FreeSWITCH has notified > > hangups (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong > > hangup-cause. Instead of notifying the successful state, we got the > > NO_ANSWER hangup cause. BUSY and NO_ANSWER states are candidates for > > retries, therefor many numbers are called/payed twice. See log below. > > > > Some other questions: > > > > B) Is the originate_timeout value an overall timer or a timer for the > > ringing (starts if SIP code 180 incomes?) stage. > > > > C) We are originating each number in a separate thread and listen to the > > channel events for updating the call result. Should we change this > > implementation or is this a good scenario/standard way. Related to the > > call result, if it is busy or not answered, the call is retried after 30 > > min. What are the recommends on this side to be ensured, the correct > > hangup case be got and the number is not called twice.. > > > > D) What do I have to bear in mind for bulk calls with parallel calls over > > 200. > > > > Thanks for your answer in advance. > > > > > > Ali > From david.ponzone at ipeva.fr Mon Aug 23 08:34:32 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 17:34:32 +0200 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: <31C26F84-23F2-4878-A68A-7DB231ABB54B@freeswitch.org> References: <20100822175311.GA63925@quark.hightek.org> <847FEB94-D341-4F00-A0A4-B5EA26181220@ipeva.fr> <31C26F84-23F2-4878-A68A-7DB231ABB54B@freeswitch.org> Message-ID: Thank you guys, but I should have said that I am perfectly aware of the pps vs. bps limitations. I had several years of experience dealing with Cisco 7200/7500 to learn about how one vendor can lie about performances :) I was just asking this to be sure about the figures. 70k pps mean around 1400 concurrent calls if ptime is 20ms (I am assuming 70k pps means 70k for both ways. If it's 70k pps with input/ output aggregated, all the following figures have to be divided by 2). I heard that some people around get 3000 calls on one box. So they get those 3000 calls probably spreading the load among multiple gigE cards. It's an important information to have for people building quite large switches that one gigE card won't handle more than 1000 calls with a regular kernel. Perhaps even 800 if you want to stay on the safe side. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 17:17, Brian West a ?crit : > I think you explained it a bit more in detail... we are saying > exactly the same thing just I'm a bit more brief. > > /b > > On Aug 23, 2010, at 9:47 AM, Nyamul Hassan wrote: > >> David, >> >> In our experience, it is not about "bps", but about "pps". Because >> the ethernet driver in Linux is single threaded, you cannot push >> beyond what a single core can handle. >> >> In our case, after 60kpps (on Intel X5504), the quality >> deteriorated. In Woody's case, this limit goes as high as 70kpps >> (on Intel X5550), which is roughly equivalent to the single core >> clock speed difference between the servers. >> >> Brian is speaking the exact same jargon, that I came across when I >> was reading about this 1 year ago. >> >> Regards >> HASSAN >> >> >> >> 2010/8/23 Brian West >> Chances are you're not hitting the bandwidth limits but the fact >> that you're moving tiny packets around. Smaller packets == more >> context switches == less throughput. You can increase your packet >> size to 60ms and gain performance. >> >> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/a57fe837/attachment-0001.html From msc at freeswitch.org Mon Aug 23 08:53:28 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Aug 2010 08:53:28 -0700 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: Message-ID: What happened when you created several thousand bridged channels as opposed to 2-person conferences? Just curious to see where your upper limits came in to play there. -MC On Sun, Aug 22, 2010 at 7:05 PM, Seven Du wrote: > Talked with brian, he thought the there might not much difference > between them and suggest a perf test by me. > > I test on my Mac 10.6.4 64bit. First I increased cps to 100 and > max-sessions to 8000. It seems that it cannot create more than 2560 > sessions on Mac, and I don't know how to increase the limit so I just > use small numbers. > > The following ruby code create channels through ESL slowly. 30 * 10 * > 2 means 600 channels, and because I used loopback, it actually use > 1200 channels. > > bridge: > 1200 threads. > 300-400% cpu. (on Activity monitor) and load avg 600-800 (on top). > Memory 300M. > > conf: > 2100 threads. > 300-400%cpu, load avg 600-800, or can be 39-1000(dosen't make sense?) > memory 500M. > > > > > > 30.times do |i| > puts i * 10 > 10.times do |j| > if ARGV[0].nil? #bridge > conn.bgapi("originate", "loopback/9664 &bridge(loopback/9664)") > else #conference > conf = "c#{i* 10 + j}@default" > conn.bgapi("originate", "loopback/9664 &conference(#{conf})") > conn.bgapi("originate", "loopback/9664 &conference(#{conf})") > end > end > > sleep 1 > end > > > > when I double the channels( 30 * 20 = i * j), ESL stuck when threads > up to 2560 and it throws "cannot create channels". But, when I run > "hupall" in FS, it start creating channels again. Don't know why FS on > Mac has a 2560 threads limit. > > And after "hupall" again, there are dead channels(9 channels and 9 > threads): > > 4d5a2408-eaf6-4f3a-a401-7a916b1911f1,outbound,2010-08-23 > > 09:52:35,1282528355,loopback/9664-a,CS_EXECUTE,,0000000000,,9664,conference,c193 at default > ,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,ffc4367e-bcde-43a5-a95e-e0fd5c4069ea > f93fa824-a6ae-47ca-ae9c-ff6b8948df4f,outbound,2010-08-23 > > 09:52:35,1282528355,loopback/9664-a,CS_EXECUTE,,0000000000,,9664,conference,c194 at default > ,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,4c8b1826-5afa-43ad-a73b-59a8c5d8f41c > 7611288b-be9e-4f70-9880-3919de567222,inbound,2010-08-23 > > 09:52:35,1282528355,loopback/9664-b,CS_REPORTING,,0000000000,,9664,playback,local_stream://moh,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,, > 9142d487-f0b5-4636-a2eb-a0adaee19634,inbound,2010-08-23 > > 09:52:35,1282528355,loopback/9664-b,CS_REPORTING,,0000000000,,9664,playback,local_stream://moh,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,, > 0d8bc340-edf7-4461-b1b4-91056c68474b,outbound,2010-08-23 > > 09:52:35,1282528355,loopback/9664-a,CS_NEW,,,,,,,,,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,a9a8369e-f509-4b39-b689-8e616d29d5c3 > ed93c3d5-6780-4e75-bf48-6e326274be04,outbound,2010-08-23 > > 09:52:35,1282528355,loopback/9664-a,CS_NEW,,,,,,,,,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,e321a328-0b7c-4f74-9af5-a5e3f083ff8a > 62aabb4b-4bfc-4ba0-a87c-2e1a76323cce,outbound,2010-08-23 > > 09:52:39,1282528359,loopback/9664-a,CS_EXECUTE,,0000000000,,9664,conference,c200 at default > ,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,3a86d88d-33b2-42b2-a57c-e2cb2c1ec486 > b13c3e9d-7fb8-46f2-90c9-04a0ec770a2e,inbound,2010-08-23 > > 09:52:39,1282528359,loopback/9664-b,CS_REPORTING,,0000000000,,9664,playback,local_stream://moh,xml,default,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,, > b8bb7b64-c57c-4c8f-9264-09eecf4aef57,outbound,2010-08-23 > > 09:52:39,1282528359,loopback/9664-a,CS_NEW,,,,,,,,,L16,8000,L16,8000,,seven-macpro.local,,,HANGUP,,,,e50d768b-37b8-4cfb-ab46-60e7ac029c83 > > > I also tested on a Linux server. It performs better. no detail data > collected though. > > And, interesting, when I run a 30 * 20 bridge + 30 * 20 conference, > loadavg suddenly grew to 2000+. But I can still run a "hupall" in > fs_cli with so high load. > > Note: > 1) loopback might not typical in test. > 2) This is not a FS performance test, I only want the conclusion that > 2-way conference uses more resource than bridged calls. > > On Thu, Aug 19, 2010 at 8:31 AM, Seven Du wrote: > > Hi, > > > > Can someone explain the performance difference between bridged calls > > and 2-party conference? or just in the code point of view? > > > > Since in some scenarios third party may join into a bridged call, so > > we need to transfer a bridged call into a conference first. Make a > > conference anyway event for 2-parties will make logic simpler and > > clear. > > > > Thanks. > > > > -- > > Blog: http://www.dujinfang.com > > Proj: http://www.freeswitch.org.cn > > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/8cb144cb/attachment.html From stephen at stephenjc.com Mon Aug 23 08:56:00 2010 From: stephen at stephenjc.com (stephen at stephenjc) Date: Mon, 23 Aug 2010 11:56:00 -0400 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: Message-ID: how would a conference vs a bridge effect bypass_media? From msc at freeswitch.org Mon Aug 23 08:58:45 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Aug 2010 08:58:45 -0700 Subject: [Freeswitch-users] core.db table channels application field store wrong data when executing voicemail In-Reply-To: References: <1282205390.25391.41.camel@marces.tc.commsmundi.com> <1282555930.25391.48.camel@marces.tc.commsmundi.com> Message-ID: > > >> > Do you need it in realtime or after the call? mod_xml_cdr contains the > callflow, I don't know whether that can be accessed during a call though. > > ESL events will give you that information I believe. > > The event socket will give you all sorts of information. In fact, every time a channel changes state or executes a dialplan app you can get an event. If you truly needed something that watches the state of every active channel then the event socket (and ESL, naturally) would certainly fit the bill. It's more work than just waiting for the XML CDR after the call is over, of course. FreeSWITCH gives you the hooks and tools to do it either way - it's just a matter of choosing the ones that fit your needs. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/fb815da4/attachment.html From michael.scheidell at secnap.com Mon Aug 23 08:56:50 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Mon, 23 Aug 2010 11:56:50 -0400 Subject: [Freeswitch-users] Newbie question Message-ID: <4C729A42.3060201@secnap.com> looked at FAQ's first, just want to clarify. Some background, looking at replacing a running sipx install due to limitations on ITSP's (conflicting ITSP and sipx limitations) Biggest limitation I see is that sipx without an external SBC forces the ITSP to send calls to port 5080. (Big Sip trunk providers like ATT, Level3 and Verizon don't want to do that. I actually understand why) in the firewall ports section it notes port 5060 for 'default' internal, 5070 for default nat and 5080 for default external. I suppose the biggest question is can I set up freeswitch to listen for SIP inbound trunk calls on port 5060 without loosing functionality (with sipx, if I do this, I can't forward calls: I don't want to interfere with normal sip: url inbound calls, so I would expect (in an 'askerisk like' setup) to be able to receive sip calls to internal extensions as well as SIP trunk calls to port 5060. (and be able to put both on hold, take them off hold, play moh to callers, transfer, blind and attended, and join a 3 way conference call) Next, looking for someone who has moved from sipx 4.2.0 to freeswitch/fusionpbx to quote me on a conversion (assuming I can use the above) POC would be to use someone like voip.ms and static registration (which uses port 5060). and have full regression testing. inbound calls, outbound calls, calls transfterd, (correct caller id shows up!) Currently, this only works on voip.ms user/password authentication since the invite/auth exchange will register port 5080. voip.ms has no way to specify the calling port, and even if they did, this is just the POC. Level3, Att and Verizon will ONLY send to port 5060. (ps, ISN dialing. seems to be tied in to e.164 dialing, but I was not able to actually tell if that is the case. on sipx, I added a two digit 'trunk' code '**' to make outbound ISN calls) -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 sip:michael.scheidell at secnap.com > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/472d7bc1/attachment.html From anthony.minessale at gmail.com Mon Aug 23 09:24:03 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Aug 2010 11:24:03 -0500 Subject: [Freeswitch-users] freeswitch CPU usage In-Reply-To: References: <20100822175311.GA63925@quark.hightek.org> <847FEB94-D341-4F00-A0A4-B5EA26181220@ipeva.fr> <31C26F84-23F2-4878-A68A-7DB231ABB54B@freeswitch.org> Message-ID: It's Always this type of thread that gets attention an not others that actually need attention. I may not have been cute and cuddly in my reply to Woody but I gave him a series of load testing aids in that same reply. And I don't notice Woody complaining. If you are offended Woody, accept my regrets. You have indeed hit a sore note because this subject always creates a huge pain for us and I don't always have time for tact in my busy day. Did you receive the numerous tips I suggested in my replies? I'll restate the most important which is probably to use the -heavy-timer command line arg to FS on a box like yours. (This was default in 1.0.4) I believe most of what I said is already on our WIKI but if not it would be a good idea to get it in there. READERS: Please strike "at our expense" from my original reply and replace it with "with the help of our absolutely free software" as some of the finer nuances in our irregular english language especially idioms may be lost in translation for some individuals. To the rest of you who have stepped up to make a non-biased performance WIKI page, Thank you for your efforts perhaps that will save us all some time in the long run. On Mon, Aug 23, 2010 at 10:34 AM, David Ponzone wrote: > Thank you guys, but I should have said that I am perfectly aware of the pps > vs. bps limitations. > I had several years of experience dealing with Cisco 7200/7500 to learn > about how one vendor can lie about performances :) > I was just asking this to be sure about the figures. > 70k pps mean around 1400 concurrent calls if ptime is 20ms (I am assuming > 70k pps means 70k for both ways. If it's 70k pps with input/output > aggregated, all the following figures have to be divided by 2). > I heard that some people around get 3000 calls on one box. > So they get those 3000 calls probably spreading the load among multiple gigE > cards. > It's an important information to have for people building quite large > switches that one gigE card won't handle more than 1000 calls with a regular > kernel. > Perhaps even 800 if you want to stay on the safe side. > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 23/08/2010 ? 17:17, Brian West a ?crit : > > I think you explained it a bit more in detail... we are saying exactly the > same thing just I'm a bit more brief. > > /b > > On Aug 23, 2010, at 9:47 AM, Nyamul Hassan wrote: > > David, > > In our experience, it is not about "bps", but about "pps". ?Because the > ethernet driver in Linux is single threaded, you cannot push beyond what a > single core can handle. > > In our case, after 60kpps (on Intel X5504), the quality deteriorated. ?In > Woody's case, this limit goes as high as 70kpps (on Intel X5550), which is > roughly equivalent to the single core clock speed difference between the > servers. > > Brian is speaking the exact same jargon, that I came across when I was > reading about this 1 year ago. > > Regards > > HASSAN > > > > 2010/8/23 Brian West > > Chances are you're not hitting the bandwidth limits but the fact that you're > moving tiny packets around. ?Smaller packets == more context switches == > less throughput. ?You can increase your packet size to 60ms and gain > performance. > > /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From gmaruzz at celliax.org Mon Aug 23 09:26:53 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 23 Aug 2010 18:26:53 +0200 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: As I yet told you, you have to find someone that has it, and that has it in the static build. But this is a problem only if you use multiple skype interfaces, I believe. Anyway, I'll contact you off list -giovanni On Sun, Aug 22, 2010 at 4:06 PM, Shamun toha md wrote: > Hi, Which skype to use? Isn't that 2.0.72 version ancient skype? IF 2.0.72 > to be used, where can we download this for Fedora/CentOS? Any idea guys!!! > > Which Skype Client to use on Linux > > Use the static build of the stable Skype client (2.0.72). > > Don't use the build for your distro, neither the 'dynamic build'. > > Don't use the beta Skype client or more recent "stable" (2.0.72 is the one > you want to use) if you want to have multiple channels with the same > skypename. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From david.ponzone at ipeva.fr Mon Aug 23 09:32:35 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 18:32:35 +0200 Subject: [Freeswitch-users] Newbie question In-Reply-To: <4C729A42.3060201@secnap.com> References: <4C729A42.3060201@secnap.com> Message-ID: <26EF30D0-1447-4C44-9475-83D72CC6C708@ipeva.fr> Michael, with FS, you do what you want. The default conf provided uses 5080 for external profile, 5060 for internal, but you can totally change that (I do), and anyway, what you do of one profile is totally up to you. External/internal are just names you can also change, they don't not imply anything. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 17:56, Michael Scheidell a ?crit : > looked at FAQ's first, just want to clarify. > Some background, looking at replacing a running sipx install due to > limitations on ITSP's > (conflicting ITSP and sipx limitations) > Biggest limitation I see is that sipx without an external SBC forces > the ITSP to send calls to port 5080. > (Big Sip trunk providers like ATT, Level3 and Verizon don't want to > do that. I actually understand why) > > in the firewall ports section it notes port 5060 for 'default' > internal, 5070 for default nat and 5080 for default external. > > I suppose the biggest question is can I set up freeswitch to listen > for SIP inbound trunk calls on port 5060 without loosing functionality > (with sipx, if I do this, I can't forward calls: > > > I don't want to interfere with normal sip: url inbound calls, so I > would expect (in an 'askerisk like' setup) to be able to receive sip > calls to internal extensions as well as SIP trunk calls to port > 5060. (and be able to put both on hold, take them off hold, play > moh to callers, transfer, blind and attended, and join a 3 way > conference call) > > Next, looking for someone who has moved from sipx 4.2.0 to > freeswitch/fusionpbx to quote me on a conversion (assuming I can use > the above) > > POC would be to use someone like voip.ms and static registration > (which uses port 5060). and have full regression testing. inbound > calls, outbound calls, calls transfterd, (correct caller id shows up!) > > Currently, this only works on voip.ms user/password authentication > since the invite/auth exchange will register port 5080. > voip.ms has no way to specify the calling port, and even if they > did, this is just the POC. Level3, Att and Verizon will ONLY send > to port 5060. > > (ps, ISN dialing. seems to be tied in to e.164 dialing, but I was > not able to actually tell if that is the case. > on sipx, I added a two digit 'trunk' code '**' to make outbound ISN > calls) > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > sip:michael.scheidell at secnap.com > > | SECNAP Network Security Corporation > Certified SNORT Integrator > 2008-9 Hot Company Award Winner, World Executive Alliance > Five-Star Partner Program 2009, VARBusiness > Best in Email Security,2010: Network Products Guide > King of Spam Filters, SC Magazine 2008 > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see http://www.secnap.com/products/spammertrap/ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/d1640d69/attachment.html From gmaruzz at celliax.org Mon Aug 23 09:33:18 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 23 Aug 2010 18:33:18 +0200 Subject: [Freeswitch-users] FreeSwitch - Skypopen cant make call In-Reply-To: References: Message-ID: are you still on your virtual platform? error says it probably cannot open the tcp sockets... please activate debug with: cli> fsctl loglevel 9 cli> console loglevel 9 then run the same calls and post all the log, since beginning, in the freeswitch pastebin (not in the mailing list). Then write to themailing list with the pastebin url. -giovanni On Sun, Aug 22, 2010 at 11:40 AM, Shamun toha md wrote: > Hi, When i am calling from another skypeID to FreeSwitch->Mod Skypopen > account. I get this following details, No call is able to establish between > Server and Regular skype caller?: > > freeswitch at example> sk list > > sk console is: |||interface1||| > F ID??? ??? Name??? ??? IB (F/T)??? OB (F/T)??? State??? CallFlw??? ??? UUID > = ====??? ? ========? ??? =======???? =======??? ======??? ============ > ====== > ? 1??? [interface1]??? ? 0/0??? ???? 0/0??? IDLE??? CALL_IDLE > > Total Interfaces: 1? IB Calls(Failed/Total): 0/0? OB Calls(Failed/Total): > 0/0 > > freeswitch at example> sk MESSAGE test1 hi there > > sk console is: |||interface1||| > > freeswitch at example> 2010-08-22 11:36:52.675088 [NOTICE] switch_channel.c:669 > New Channel skypopen/interface1 [cbb3b872-add0-11df-a4c6-75d067ab3b3d] > 2010-08-22 11:36:52.675088 [NOTICE] mod_skypopen.c:1944 Channel > [skypopen/interface1] has been answered > 2010-08-22 11:36:52.677551 [INFO] mod_dialplan_xml.c:418 Processing alu > malu->5000 in context default > 2010-08-22 11:36:52.687099 [NOTICE] mod_skypopen.c:1171 Hangup > skypopen/interface1 [CS_EXECUTE] [NORMAL_CLEARING] > 2010-08-22 11:36:52.750656 [NOTICE] switch_core_session.c:1182 Session 1 > (skypopen/interface1) Ended > 2010-08-22 11:36:52.750656 [NOTICE] switch_core_session.c:1184 Close Channel > skypopen/interface1 [CS_DESTROY] > 2010-08-22 11:36:52.987482 [ERR] skypopen_protocol.c:233 rev [(nil)|37 > ][ERRORA? 233? ][interface1][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER > CALL: unable to alter input/output||| > 2010-08-22 11:36:52.987482 [ERR] skypopen_protocol.c:235 rev [(nil)|37 > ][ERRORA? 235? ][interface1][-1, 0,16] skype_call now is DOWN > 2010-08-22 11:37:18.045973 [NOTICE] switch_channel.c:669 New Channel > skypopen/interface1 [dad30380-add0-11df-a4c7-75d067ab3b3d] > 2010-08-22 11:37:18.047140 [NOTICE] mod_skypopen.c:1944 Channel > [skypopen/interface1] has been answered > 2010-08-22 11:37:18.048145 [INFO] mod_dialplan_xml.c:418 Processing alu > malu->5000 in context default > 2010-08-22 11:37:18.058690 [NOTICE] mod_skypopen.c:1171 Hangup > skypopen/interface1 [CS_EXECUTE] [NORMAL_CLEARING] > 2010-08-22 11:37:18.072829 [NOTICE] switch_core_session.c:1182 Session 2 > (skypopen/interface1) Ended > 2010-08-22 11:37:18.072829 [NOTICE] switch_core_session.c:1184 Close Channel > skypopen/interface1 [CS_DESTROY] > 2010-08-22 11:37:18.359355 [ERR] skypopen_protocol.c:233 rev [(nil)|37 > ][ERRORA? 233? ][interface1][-1, 0, 0] Skype got ERROR: |||ERROR 589 ALTER > CALL: unable to alter input/output||| > 2010-08-22 11:37:18.359355 [ERR] skypopen_protocol.c:235 rev [(nil)|37 > ][ERRORA? 235? ][interface1][-1, 0,16] skype_call now is DOWN > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From msc at freeswitch.org Mon Aug 23 09:36:17 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Aug 2010 09:36:17 -0700 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: Message-ID: On Mon, Aug 23, 2010 at 8:56 AM, stephen at stephenjc wrote: > how would a conference vs a bridge effect bypass_media? > A conference can't bypass media - by design it is mixing media streams. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/ffbf7196/attachment.html From mike at jerris.com Mon Aug 23 11:04:25 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Aug 2010 14:04:25 -0400 Subject: [Freeswitch-users] Platform requrest (DragonflyBSD) for the Jira issue management system In-Reply-To: <20100822154128.GA62723@quark.hightek.org> References: <20100728053058.GA26436@quark.hightek.org> <20100821220315.GA52528@quark.hightek.org> <3BFC3227-59D0-4D45-974E-1E3C04EF6547@freeswitch.org> <20100822154128.GA62723@quark.hightek.org> Message-ID: On Aug 22, 2010, at 11:41 AM, Vincent Stemen wrote: > On Sat, Aug 21, 2010 at 05:45:47PM -0500, Brian West wrote: >> Vince, >> I see no value in adding it... but I added it anyway... > > I appreciate it. May I ask why you do not see any value in it though? > If you prefer to encompass all the BSD's into one platform option, I > would suggest changing the "FreeBSD/gcc" option to "BSD/gcc". It can't be the same classification. The reason it didn't build in the past was due to dragonfly specific issues. There is no issue having it in the list if there are active community members providing patches to fix these issues. Are you able to volunteer to provide patches for all dragonfly specific issues? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/e6238b1d/attachment-0001.html From mike at jerris.com Mon Aug 23 11:06:45 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Aug 2010 14:06:45 -0400 Subject: [Freeswitch-users] FS with Red5 Streaming-Server (RTMP-Stream) In-Reply-To: References: <4C5FAE23.3000406@infosecurity.ch> <4C602C60.4010803@infosecurity.ch> Message-ID: <8B043C0C-8765-409A-A5AB-94FDB9A78A1E@jerris.com> why would calling a cell phone make a difference? As it happens the other person was on a cell phone. On Aug 22, 2010, at 3:17 PM, jesse wrote: > " I have had quite good audio quality results in testing when using a > headset. " , next time try to call a cell phone from your flash > client. The root cause is FLASH only supports very limited codec set. From anthony.minessale at gmail.com Mon Aug 23 11:26:02 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Aug 2010 13:26:02 -0500 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: Message-ID: A conference for sure uses more resources, inline muxing, always transcoding and 2 threads per call. It still scales well but not as well as bridges. On Mon, Aug 23, 2010 at 11:36 AM, Michael Collins wrote: > > > On Mon, Aug 23, 2010 at 8:56 AM, stephen at stephenjc > wrote: >> >> how would a conference vs a bridge effect bypass_media? > > A conference can't bypass media - by design it is mixing media streams. > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jason at cloudtree.net Mon Aug 23 11:28:37 2010 From: jason at cloudtree.net (Jason Jeffords) Date: Mon, 23 Aug 2010 14:28:37 -0400 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Hi Giovanni, We got some test cycles today and have found the following: 1) SIP to Skype calls work as desired - A-leg is SIP, B-leg is Skype 2) Skype to Internal Extensions Work as desired - A-leg is Skype, B-leg is FreeSwitch (internal extension) 3) Skype to SIP does not work - A-leg is Skype, B-leg is SIP In the last case: 1) We hear Skype ringing until the B-leg (SIP) is answered - this is correct 2) We then get dead air and the SIP B-leg hangs up on its own (i.e. the user answering does not hang up, but the connection hangs up) 3) The Skype client connection stays up with dead air until it is manually hung up We are running the latest code as of today. Any insight you could provide will be most appreciated. Thanks of all your work, Jason On Fri, Aug 13, 2010 at 5:31 PM, Giovanni Maruzzelli wrote: > Jason, > > please test with the latest git version. > > Many messy changes to an already messy code, so maybe some bug or side > effect (ok, bug) has been introduced. > > commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 > Author: Giovanni Maruzzelli > Date: Fri Aug 13 16:19:20 2010 -0500 > > skypopen: now answer a call only when directed to do it (before > was trying to answer any incoming call). Lot of changes to a messy > part, so maybe some problem will come out... > > Signed-off-by: Giovanni Maruzzelli > > -giovanni > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/e07307d6/attachment.html From ken at ukgb.net Mon Aug 23 11:35:20 2010 From: ken at ukgb.net (Ken Gillett) Date: Mon, 23 Aug 2010 19:35:20 +0100 Subject: [Freeswitch-users] Account selection In-Reply-To: <2DD257A8-A956-45F1-9F3C-40A6EE3E77D3@ipeva.fr> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> <2DD257A8-A956-45F1-9F3C-40A6EE3E77D3@ipeva.fr> Message-ID: <773D1529-5F42-4C53-BE5C-7A2C1B0546B6@ukgb.net> On 23 Aug 2010, at 15:13, David Ponzone wrote: > There are 2 simple ways to do that: > -use only one account on your softphone, and use prefixes: easy to implement, but easy to make mistake for the users too > -use 6 accounts on your softphone as you used to do, with a 1-to-1 mapping with the external SIP accounts. You have to use a different FS context for each account in order to do that. That's what I am thinking would be the best solution, but could it be 'many to 1'? IOW, could I have several different extensions all mapped to the same external account, so incoming calls on that account go to the grouped extensions and if any of them ring out, it goes out via that same external account? Here's another related question. Is it possible to register more than one SIP client to a single extension, or would you just use a different extension for each and set them up in a (hunt) group? > If you find a nice softphone with programmable keys, you can even assign the keys so you have a line key per account. That would be the easier to use I think. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 23/08/2010 ? 15:26, Ken Gillett a ?crit : > >> Let's say there are 6 SIP accounts to which FreeSwitch is 'registered' and it can make calls using any of them. But which one is used? One of the local extensions (let's say a softphone) needs to receive calls from all of these accounts. This is simple enough and the recipient should see the incoming call information i.e. which account the call is to. But when that extension makes an outgoing call, how can it specify which is the SIP account used to by FS to make the call? >> >> This can be very important when each SIP account represents a different company/business. Although one person is dealing with all those businesses, when an outgoing call is made it is imperative that the correct SIP account is used to make that call so that the recipient is correctly informed who is making the call. >> >> Currently (no PBX), my softphone registers to each of these 6 accounts and I can choose which account to use to make a call. But if I am registered to FS as a single extension, how can I tell FS which account to use when I place an outgoing call? Is there any way to do this without having to use Dial plans? >> >> >> On 23 Aug 2010, at 11:46, David Ponzone wrote: >> >>> Ken, >>> >>> I am not really sure to understand your issue/question. >>> Can you describe exactly the equipements involved and what you want to do ? >>> Is FS used as a PBX or a a provider softswitch to terminate the trunk coming form the PBX ? >>> >>> Some various information that could help you in the meantime: >>> -most softphones can have several SIP accounts, but you should check that they can register all of them at the same time >>> -if your objective is to have FS sending calls to a specific external VoIP account when it receives a call from a specific internal account, like this: >>> phone1-----> FS-------> Provider SIP Account 1 >>> phone2----->FS--------> Provider SIP Account 2 >>> you would need to split the outgoing calls one way or another: you could do that based on the caller-id, or you may put your internal accounts in different contexts, so they use different dialplans. >>> There are probably other ways, like using a prefix, but this one is probably a burden for the user and a security issue possibly. >>> A such configuration is really some sort of SBC, when you want to avoid your SIP devices to connect to the accounts provided by your carrier directly, because you are concerned with security or because you want to keep control on the calls to provide more services to your users. >>> >>> Le 23/08/2010 ? 09:57, Ken Gillett a ?crit : >>> >>>> If one wishes to have use of several VOIP 'lines', but with no PBX, you need to register all those SIP accounts with the client (softphone etc). You should then be informed which account is receiving a call and can pick a particular account from which to make calls. Once a PBX is in use, you can register the client as a single extension of the PBX and direct calls as appropriate to that extension - I assume with the correct caller ID and incoming account information passed to the recipient so they know as much as in the 'no PBX' configuration. >>>> >>>> But what about outgoing calls. In this scenario, registered as a single extension, how would it be possible to pick the outgoing 'line' (i.e. account) to use? Would it have to be done by dialling a prefix or is there another way? Is it client dependent? >> Ken G i l l e t t _/_/_/_/_/_/_/_/ From mrene_lists at avgs.ca Mon Aug 23 11:53:00 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 23 Aug 2010 14:53:00 -0400 Subject: [Freeswitch-users] french language implementation In-Reply-To: <4C72913E.50308@communicatefreely.net> References: <4C72913E.50308@communicatefreely.net> Message-ID: Hi, The french canadian sound files are available at: http://svn.freeswitch.org/svn/sounds/trunk/fr/ca/june/ Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-08-23, at 11:18 AM, Tim St. Pierre wrote: > Hello, > > I would like to make a proper bilingual PBX implementation here in Canada, and it looks like there > isn't a stock set of Canadian French sound files. > > I talked to GM Voices, and they have some voice talent available to do this job. Our company is > willing to pay for it (they gave us a pretty good deal), and we will contribute the prompt set back > to the project once it's done. > > Being that it's a different language, and some numbers and phrases are said in a different format, > what do I need to change to make all this work? > > I imagine I have to modify some phrase macros somewhere. > > Does anyone have some instructions, or is there someone fluent in French willing to take this on? > > I see references to fr in modules.conf, but I can't find the audio files. Is it possible that > someone has already done this work, and all we have to do is record the prompts? > > Any help is appreciated. > > -Tim > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/a89eab47/attachment.html From mthakershi at gmail.com Mon Aug 23 12:19:29 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Mon, 23 Aug 2010 14:19:29 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? Message-ID: Hello, Since last few days my freeswitch server receives SPAM calls. 25 to 30 calls at the same time. It makes my mod_managed program crash. Unlike regular phone numbers in US, these seem to come from sip IP addresses. When I see location of these IP addresses, most of them seem to be from China. Sample from CDR: sip XML sip sip 113.105.152.220 1442073479999 1056 bef84579-d25a-ab4b-aa52-dbcf9d7509be mod_sofia default sofia/sipinterface_1/sip at 113.105.152.220 I am only expecting call from regular phone lines from USA. I use Vitelity and I have secured my Vitelity configuration (SOFIA) with secret code provided by them. How do I stop these anonymous SIP callers from crashing my system? Please help. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/96dcd9b6/attachment-0001.html From brian at freeswitch.org Mon Aug 23 12:26:09 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 14:26:09 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: References: Message-ID: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> Well you're using FreePBX right? The only corse of action you have is to find out why its crashing and reporting the issue on our Jira. Without any more info to go on you're SOL. http://www.google.com/search?hl=en&client=safari&rls=en&defl=en&q=define:Vishing&sa=X&ei=RstyTO24JI_Znge7-6yNCw&ved=0CBIQkAE /b On Aug 23, 2010, at 2:19 PM, Malay Thakershi wrote: > Hello, > > Since last few days my freeswitch server receives SPAM calls. 25 to 30 calls at the same time. It makes my mod_managed program crash. > > Unlike regular phone numbers in US, these seem to come from sip IP addresses. When I see location of these IP addresses, most of them seem to be from China. > > Sample from CDR: > > sip > XML > sip > > > sip > 113.105.152.220 > 1442073479999 > 1056 > bef84579-d25a-ab4b-aa52-dbcf9d7509be > mod_sofia > default > sofia/sipinterface_1/sip at 113.105.152.220 > > > I am only expecting call from regular phone lines from USA. I use Vitelity and I have secured my Vitelity configuration (SOFIA) with secret code provided by them. > > How do I stop these anonymous SIP callers from crashing my system? > > Please help. > > Thank you. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From david.ponzone at ipeva.fr Mon Aug 23 12:32:35 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 21:32:35 +0200 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> Message-ID: <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> Brian he can't add an ACL with FreePBX ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 21:26, Brian West a ?crit : > Well you're using FreePBX right? The only corse of action you have > is to find out why its crashing and reporting the issue on our > Jira. Without any more info to go on you're SOL. > > http://www.google.com/search?hl=en&client=safari&rls=en&defl=en&q=define:Vishing&sa=X&ei=RstyTO24JI_Znge7-6yNCw&ved=0CBIQkAE > > /b > > On Aug 23, 2010, at 2:19 PM, Malay Thakershi wrote: > >> Hello, >> >> Since last few days my freeswitch server receives SPAM calls. 25 to >> 30 calls at the same time. It makes my mod_managed program crash. >> >> Unlike regular phone numbers in US, these seem to come from sip IP >> addresses. When I see location of these IP addresses, most of them >> seem to be from China. >> >> Sample from CDR: >> >> sip >> XML >> sip >> >> >> sip >> 113.105.152.220 >> 1442073479999 >> 1056 >> bef84579-d25a-ab4b-aa52-dbcf9d7509be >> mod_sofia >> default >> sofia/sipinterface_1/sip at 113.105.152.220 >> >> >> I am only expecting call from regular phone lines from USA. I use >> Vitelity and I have secured my Vitelity configuration (SOFIA) with >> secret code provided by them. >> >> How do I stop these anonymous SIP callers from crashing my system? >> >> Please help. >> >> Thank you. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/7e5161f7/attachment.html From brian at freeswitch.org Mon Aug 23 12:36:28 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 14:36:28 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> Message-ID: <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> David, No Clue, Never Used It, Can't Say... /b On Aug 23, 2010, at 2:32 PM, David Ponzone wrote: > Brian > > he can't add an ACL with FreePBX ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 23/08/2010 ? 21:26, Brian West a ?crit : > >> Well you're using FreePBX right? The only corse of action you have is to find out why its crashing and reporting the issue on our Jira. Without any more info to go on you're SOL. >> >> http://www.google.com/search?hl=en&client=safari&rls=en&defl=en&q=define:Vishing&sa=X&ei=RstyTO24JI_Znge7-6yNCw&ved=0CBIQkAE >> >> /b From shamun.toha at gmail.com Mon Aug 23 12:30:19 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Mon, 23 Aug 2010 21:30:19 +0200 Subject: [Freeswitch-users] FreeSwitch - Skypopen cant make call In-Reply-To: References: Message-ID: No Sorry, This is solved just by updating the idiot Skype Client from 2.1.x to 2.0.72. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/b2bde99c/attachment.html From michael.scheidell at secnap.com Mon Aug 23 12:30:40 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Mon, 23 Aug 2010 15:30:40 -0400 Subject: [Freeswitch-users] How to stop SPAM calls? References: Message-ID: <83A9F79016D8A440BC4EEA4357E97B22AA4C93@secnap3.secnap.com> -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Malay Thakershi Sent: Mon 8/23/2010 3:19 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How to stop SPAM calls? Hello, Since last few days my freeswitch server receives SPAM calls. 25 to 30 calls at the same time. It makes my mod_managed program crash. Unlike regular phone numbers in US, these seem to come from sip IP addresses. When I see location of these IP addresses, most of them seem to be from China. Sample from CDR: sip XML sip sip 113.105.152.220 1442073479999 1056 bef84579-d25a-ab4b-aa52-dbcf9d7509be mod_sofia default sofia/sipinterface_1/sip at 113.105.152.220 I am only expecting call from regular phone lines from USA. I use Vitelity and I have secured my Vitelity configuration (SOFIA) with secret code provided by them. How do I stop these anonymous SIP callers from crashing my system? Please help. Thank you. google for sipvisious. they also have a (python 25) based program that might intercept these and block them (crashing THEIR system) ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ From mthakershi at gmail.com Mon Aug 23 12:47:04 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Mon, 23 Aug 2010 14:47:04 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> Message-ID: I am going through documentation but seems iptables can eliminate calls being made on ports other than required ones. But my server is Windows. How do I run iptables command? Also, could you tell me if I block all incoming ports other than 5060 and 5061, will my regular inbound calls work? Thank you. 2010/8/23 Brian West > David, > No Clue, Never Used It, Can't Say... > > /b > > On Aug 23, 2010, at 2:32 PM, David Ponzone wrote: > > > Brian > > > > he can't add an ACL with FreePBX ? > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > Le 23/08/2010 ? 21:26, Brian West a ?crit : > > > >> Well you're using FreePBX right? The only corse of action you have is to > find out why its crashing and reporting the issue on our Jira. Without any > more info to go on you're SOL. > >> > >> > http://www.google.com/search?hl=en&client=safari&rls=en&defl=en&q=define:Vishing&sa=X&ei=RstyTO24JI_Znge7-6yNCw&ved=0CBIQkAE > >> > >> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/8bea1bf5/attachment-0001.html From 12ukwn at gmail.com Mon Aug 23 12:47:21 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Mon, 23 Aug 2010 21:47:21 +0200 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: References: Message-ID: <20100823214721.290108b0@anubis.defcon1> Le Mon, 23 Aug 2010 14:19:29 -0500, Malay Thakershi a ?crit : > Unlike regular phone numbers in US, these seem to come from sip IP > addresses. When I see location of these IP addresses, most of them seem to > be from China. ... > I am only expecting call from regular phone lines from USA. I use Vitelity > and I have secured my Vitelity configuration (SOFIA) with secret code > provided by them. > > How do I stop these anonymous SIP callers from crashing my system? The crash isn't normal, file a bug. There are some ways to do what you want. First: you could refuse anonymous calls (but it might be a no-no in your job.) Faster: forbid (drop, not reject) the whole IP adresses range that bothers you to at the firewall level (113.96.0.0 - 113.111.255.255) Other: may be there's another possibility into FS but the former is still faster as it instantly refuse the connexion instead sending it to FS thus eating CPU to do that. Last: Crack defense computers and nuke China. -- From brian at freeswitch.org Mon Aug 23 12:53:04 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 14:53:04 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <83A9F79016D8A440BC4EEA4357E97B22AA4C93@secnap3.secnap.com> References: <83A9F79016D8A440BC4EEA4357E97B22AA4C93@secnap3.secnap.com> Message-ID: <10EA2260-7199-4AC0-A5F9-CDE1BB634F37@freeswitch.org> Lets not land the lad in prison mmmkay. Things like this sound highly illegal. /b On Aug 23, 2010, at 2:30 PM, Michael Scheidell wrote: > google for sipvisious. they also have a (python 25) based program that might intercept these and block them (crashing THEIR system) From david.ponzone at ipeva.fr Mon Aug 23 12:57:39 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 21:57:39 +0200 Subject: [Freeswitch-users] Account selection In-Reply-To: <773D1529-5F42-4C53-BE5C-7A2C1B0546B6@ukgb.net> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> <2DD257A8-A956-45F1-9F3C-40A6EE3E77D3@ipeva.fr> <773D1529-5F42-4C53-BE5C-7A2C1B0546B6@ukgb.net> Message-ID: Answers inline: Le 23/08/2010 ? 20:35, Ken Gillett a ?crit : > On 23 Aug 2010, at 15:13, David Ponzone wrote: > >> There are 2 simple ways to do that: >> -use only one account on your softphone, and use prefixes: easy to >> implement, but easy to make mistake for the users too >> -use 6 accounts on your softphone as you used to do, with a 1-to-1 >> mapping with the external SIP accounts. You have to use a different >> FS context for each account in order to do that. > > > That's what I am thinking would be the best solution, but could it > be 'many to 1'? IOW, could I have several different extensions all > mapped to the same external account, so incoming calls on that > account go to the grouped extensions and if any of them ring out, it > goes out via that same external account? I am not sure I get it but I think :) Let's say you have 3 external accounts and you have 3 users (softphone). You want all 3 users to receive calls for all external accounts and also to be able to dial out through them, don't you ? Well, of course, you can: Softphone1 would have 3 users: user11 in context ext1, user12 in contet ext2, user13 in context ext3. Softphone2 would have 3 users: user21 in context ext1, user22 in contet ext2, user23 in context ext3. etc.. for Softphone3. ext1 is a context where calls going out will use external account 1. ext2 is a context where calls going out will use external account 2. etc.. For incoming calls, it's going to depend on what information allows you do distinguish the calls. If it's a DID, it's easy. All incoming calls will hit the public dialplan. You just have to add extensions (FreeSWITCH extensions, so rules) so that if the call is for DID1, then user11, user21 and user31 will be called. A such bridge command allows you to dial them all simultaneously or sequentially. > Here's another related question. Is it possible to register more > than one SIP client to a single extension, or would you just use a > different extension for each and set them up in a (hunt) group? > Yes you can! You have to enable that in the SIP profile with: Value can also be "contact", but I don't know the difference. For outbound, it's the same than previously, except you only need 3 accounts (one per external account). For inbound calls, you will be able to ring all devices at the same time in an easier way with: Was this helpful ? >> If you find a nice softphone with programmable keys, you can even >> assign the keys so you have a line key per account. That would be >> the easier to use I think. >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? l'intention exclusive de ses destinataires. Toute >> utilisation ou diffusion non autoris?e est interdite. Tout message >> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute >> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou >> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de >> le d?truire imm?diatement et d'avertir l'exp?diteur. >> >> >> >> >> Le 23/08/2010 ? 15:26, Ken Gillett a ?crit : >> >>> Let's say there are 6 SIP accounts to which FreeSwitch is >>> 'registered' and it can make calls using any of them. But which >>> one is used? One of the local extensions (let's say a softphone) >>> needs to receive calls from all of these accounts. This is simple >>> enough and the recipient should see the incoming call information >>> i.e. which account the call is to. But when that extension makes >>> an outgoing call, how can it specify which is the SIP account used >>> to by FS to make the call? >>> >>> This can be very important when each SIP account represents a >>> different company/business. Although one person is dealing with >>> all those businesses, when an outgoing call is made it is >>> imperative that the correct SIP account is used to make that call >>> so that the recipient is correctly informed who is making the call. >>> >>> Currently (no PBX), my softphone registers to each of these 6 >>> accounts and I can choose which account to use to make a call. But >>> if I am registered to FS as a single extension, how can I tell FS >>> which account to use when I place an outgoing call? Is there any >>> way to do this without having to use Dial plans? >>> >>> >>> On 23 Aug 2010, at 11:46, David Ponzone wrote: >>> >>>> Ken, >>>> >>>> I am not really sure to understand your issue/question. >>>> Can you describe exactly the equipements involved and what you >>>> want to do ? >>>> Is FS used as a PBX or a a provider softswitch to terminate the >>>> trunk coming form the PBX ? >>>> >>>> Some various information that could help you in the meantime: >>>> -most softphones can have several SIP accounts, but you should >>>> check that they can register all of them at the same time >>>> -if your objective is to have FS sending calls to a specific >>>> external VoIP account when it receives a call from a specific >>>> internal account, like this: >>>> phone1-----> FS-------> Provider SIP Account 1 >>>> phone2----->FS--------> Provider SIP Account 2 >>>> you would need to split the outgoing calls one way or another: >>>> you could do that based on the caller-id, or you may put your >>>> internal accounts in different contexts, so they use different >>>> dialplans. >>>> There are probably other ways, like using a prefix, but this one >>>> is probably a burden for the user and a security issue possibly. >>>> A such configuration is really some sort of SBC, when you want to >>>> avoid your SIP devices to connect to the accounts provided by >>>> your carrier directly, because you are concerned with security or >>>> because you want to keep control on the calls to provide more >>>> services to your users. >>>> >>>> Le 23/08/2010 ? 09:57, Ken Gillett a ?crit : >>>> >>>>> If one wishes to have use of several VOIP 'lines', but with no >>>>> PBX, you need to register all those SIP accounts with the client >>>>> (softphone etc). You should then be informed which account is >>>>> receiving a call and can pick a particular account from which to >>>>> make calls. Once a PBX is in use, you can register the client as >>>>> a single extension of the PBX and direct calls as appropriate to >>>>> that extension - I assume with the correct caller ID and >>>>> incoming account information passed to the recipient so they >>>>> know as much as in the 'no PBX' configuration. >>>>> >>>>> But what about outgoing calls. In this scenario, registered as a >>>>> single extension, how would it be possible to pick the outgoing >>>>> 'line' (i.e. account) to use? Would it have to be done by >>>>> dialling a prefix or is there another way? Is it client dependent? >>> > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/7627ea7c/attachment.html From david.ponzone at ipeva.fr Mon Aug 23 13:03:43 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 23 Aug 2010 22:03:43 +0200 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> Message-ID: <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> If I understand correctly, you expect calls form PSTN, so only from the known IPs of your provider ? You can then filter all other IPs going to your port X (5060, 5080, your mileage may vary). Also, a call coming to a port you don't use (so not opened) should not have ANY impact. It should not even hit the dialplan. it should be rejected with ICMP port unreachable by the Windows TCP/IP stack. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 21:47, Malay Thakershi a ?crit : > I am going through documentation but seems iptables can eliminate > calls being made on ports other than required ones. > > But my server is Windows. How do I run iptables command? > > Also, could you tell me if I block all incoming ports other than > 5060 and 5061, will my regular inbound calls work? > > Thank you. > > > > 2010/8/23 Brian West > David, > No Clue, Never Used It, Can't Say... > > /b > > On Aug 23, 2010, at 2:32 PM, David Ponzone wrote: > > > Brian > > > > he can't add an ACL with FreePBX ? > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > > > > > > Le 23/08/2010 ? 21:26, Brian West a ?crit : > > > >> Well you're using FreePBX right? The only corse of action you > have is to find out why its crashing and reporting the issue on our > Jira. Without any more info to go on you're SOL. > >> > >> http://www.google.com/search?hl=en&client=safari&rls=en&defl=en&q=define:Vishing&sa=X&ei=RstyTO24JI_Znge7-6yNCw&ved=0CBIQkAE > >> > >> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/855250b9/attachment-0001.html From mthakershi at gmail.com Mon Aug 23 13:10:43 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Mon, 23 Aug 2010 15:10:43 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <10EA2260-7199-4AC0-A5F9-CDE1BB634F37@freeswitch.org> References: <83A9F79016D8A440BC4EEA4357E97B22AA4C93@secnap3.secnap.com> <10EA2260-7199-4AC0-A5F9-CDE1BB634F37@freeswitch.org> Message-ID: I am checking sipvisious. A correction: freeswitch doesn't crash. Program I have written in mod_managed on receiving calls does crash (probably due to issue with my code). 1. I would like to stop calls from being answered. That would be ideal. May be sipvisious can help. I am checking. 2. If in short term I can do something from my code, such as checking port/caller id, and then not going through the prompt (my app) process. How can I do that? Thank you. On Mon, Aug 23, 2010 at 2:53 PM, Brian West wrote: > Lets not land the lad in prison mmmkay. Things like this sound highly > illegal. > > /b > > On Aug 23, 2010, at 2:30 PM, Michael Scheidell wrote: > > > google for sipvisious. they also have a (python 25) based program that > might intercept these and block them (crashing THEIR system) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/3083f580/attachment.html From jeff at jefflenk.com Mon Aug 23 14:01:17 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 23 Aug 2010 14:01:17 -0700 (PDT) Subject: [Freeswitch-users] Phrase speak-text function return on first key press in phrase file on Windows In-Reply-To: References: <1282530102895-5451221.post@n2.nabble.com> Message-ID: <1282597277910-5454286.post@n2.nabble.com> Fixed in git head -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Phrase-speak-text-function-return-on-first-key-press-in-phrase-file-on-Windows-tp5449059p5454286.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cliff at develix.com Mon Aug 23 14:04:11 2010 From: cliff at develix.com (Cliff Wells) Date: Mon, 23 Aug 2010 14:04:11 -0700 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT Message-ID: <1282597451.3516.54.camel@portable-evil> I have two servers, both seemingly configured identically. One can make outbound calls via the gateway, the other cannot. Here's the error: EXECUTE FreeTDM/1:25/7771112222 bridge(sofia/gateway/outbound/18001231234) 2010-08-23 08:52:05.918700 [ERR] mod_sofia.c:3668 Invalid Gateway 2010-08-23 08:52:05.918700 [NOTICE] mod_sofia.c:3984 Close Channel N/A [CS_NEW] 2010-08-23 08:52:05.918700 [DEBUG] switch_core_state_machine.c:430 () Running State Change CS_DESTROY 2010-08-23 08:52:05.918700 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY 2010-08-23 08:52:05.918700 [DEBUG] mod_sofia.c:358 N/A SOFIA DESTROY 2010-08-23 08:52:05.918700 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY going to sleep 2010-08-23 08:52:05.918700 [ERR] switch_ivr_originate.c:2600 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2010-08-23 08:52:05.918700 [DEBUG] switch_ivr_originate.c:3408 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2010-08-23 08:52:05.918700 [INFO] mod_dptools.c:2393 Originate Failed. Cause: INVALID_NUMBER_FORMAT 2010-08-23 08:52:05.918700 [DEBUG] switch_channel.c:2261 (FreeTDM/1:25/7017120999) Callstate Change RINGING -> HANGUP 2010-08-23 08:52:05.918700 [NOTICE] mod_dptools.c:2456 Hangup FreeTDM/1:25/7017120999 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] 2010-08-23 08:52:05.918700 [DEBUG] switch_channel.c:2277 Send signal FreeTDM/1:25/7771112222 [KILL] Dialplan is via mod_xml_curl (both servers get the same info from the same http server) and looks like:
The gateway (identical on both systems): Again, this setup works on one system but not the other and I'm at a loss since I can't be sure what number FS is complaining about (I assume the outbound). Thanks in advance, Cliff -- From brian at freeswitch.org Mon Aug 23 14:10:56 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 16:10:56 -0500 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT In-Reply-To: <1282597451.3516.54.camel@portable-evil> References: <1282597451.3516.54.camel@portable-evil> Message-ID: <64247BA4-F7FC-4AF5-91DC-4EF03D5608AC@freeswitch.org> This should clue you in... the gateway doesn't exist. /b On Aug 23, 2010, at 4:04 PM, Cliff Wells wrote: > 2010-08-23 08:52:05.918700 [ERR] mod_sofia.c:3668 Invalid Gateway From mthakershi at gmail.com Mon Aug 23 14:11:43 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Mon, 23 Aug 2010 16:11:43 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> Message-ID: That is true. So do I block all other IP in my firewall? Or do I configure that in FreeSwitch? Also, How can be sure my provider's IP to remain same? (I use vitelity) Please let me know. On Mon, Aug 23, 2010 at 3:03 PM, David Ponzone wrote: > If I understand correctly, you expect calls form PSTN, so only from the > known IPs of your provider ? > You can then filter all other IPs going to your port X (5060, 5080, your > mileage may vary). > > Also, a call coming to a port you don't use (so not opened) should not have > ANY impact. > It should not even hit the dialplan. > it should be rejected with ICMP port unreachable by the Windows TCP/IP > stack. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 23/08/2010 ? 21:47, Malay Thakershi a ?crit : > > I am going through documentation but seems iptables can eliminate calls > being made on ports other than required ones. > > But my server is Windows. How do I run iptables command? > > Also, could you tell me if I block all incoming ports other than 5060 and > 5061, will my regular inbound calls work? > > Thank you. > > > > 2010/8/23 Brian West > >> David, >> No Clue, Never Used It, Can't Say... >> >> /b >> >> On Aug 23, 2010, at 2:32 PM, David Ponzone wrote: >> >> > Brian >> > >> > he can't add an ACL with FreePBX ? >> > >> > David Ponzone Direction Technique >> > email: david.ponzone at ipeva.fr >> > tel: 01 74 03 18 97 >> > gsm: 06 66 98 76 34 >> > >> > Service Client IPeva >> > tel: 0811 46 26 26 >> > www.ipeva.fr - www.ipeva-studio.com >> > >> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. >> > >> > >> > >> > >> > Le 23/08/2010 ? 21:26, Brian West a ?crit : >> > >> >> Well you're using FreePBX right? The only corse of action you have is >> to find out why its crashing and reporting the issue on our Jira. Without >> any more info to go on you're SOL. >> >> >> >> >> http://www.google.com/search?hl=en&client=safari&rls=en&defl=en&q=define:Vishing&sa=X&ei=RstyTO24JI_Znge7-6yNCw&ved=0CBIQkAE >> >> >> >> /b >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/680689c2/attachment-0001.html From msc at freeswitch.org Mon Aug 23 14:24:56 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Aug 2010 14:24:56 -0700 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT In-Reply-To: <64247BA4-F7FC-4AF5-91DC-4EF03D5608AC@freeswitch.org> References: <1282597451.3516.54.camel@portable-evil> <64247BA4-F7FC-4AF5-91DC-4EF03D5608AC@freeswitch.org> Message-ID: Two commands to help you see what's up: sofia status sofia status profile external (Assuming you defined the gateway in the external profile) Make sure that your gateway is loading properly. If you can restart the profile without messing things up then try doing that from the command line and watch for errors. (You can do "sofia profile external restart") -MC On Mon, Aug 23, 2010 at 2:10 PM, Brian West wrote: > This should clue you in... the gateway doesn't exist. > > /b > > On Aug 23, 2010, at 4:04 PM, Cliff Wells wrote: > > > 2010-08-23 08:52:05.918700 [ERR] mod_sofia.c:3668 Invalid Gateway > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/49850878/attachment.html From chat2jesse at gmail.com Mon Aug 23 14:39:47 2010 From: chat2jesse at gmail.com (jesse) Date: Mon, 23 Aug 2010 14:39:47 -0700 Subject: [Freeswitch-users] How could install PHP ESL? Message-ID: The current FS version doesn't include source code for PHP. What are the steps to install it on top of my current version? where to get the source? how to build and install? Iif you guys want to keep the new system slim by get rid of PHP, please at least keep a doc about how to add it in case of need. thanks! -jesse From brian at freeswitch.org Mon Aug 23 14:52:44 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 16:52:44 -0500 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: You'll need php-devel and all the headers to build it installed... then try from there. I personally do not like or use PHP so someone that uses and likes it would need to write up some documentation on the wiki. Thanks, Brian On Aug 23, 2010, at 4:39 PM, jesse wrote: > The current FS version doesn't include source code for PHP. > What are the steps to install it on top of my current version? where > to get the source? how to build and install? > > Iif you guys want to keep the new system slim by get rid of PHP, > please at least keep a doc about how to add it in case of need. > > thanks! > > -jesse From cliff at develix.com Mon Aug 23 15:05:15 2010 From: cliff at develix.com (Cliff Wells) Date: Mon, 23 Aug 2010 15:05:15 -0700 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT In-Reply-To: <64247BA4-F7FC-4AF5-91DC-4EF03D5608AC@freeswitch.org> References: <1282597451.3516.54.camel@portable-evil> <64247BA4-F7FC-4AF5-91DC-4EF03D5608AC@freeswitch.org> Message-ID: <1282601115.3516.93.camel@portable-evil> Yeah, I'm an idiot. My gateway file was named outbound.conf, not outbound.conf.xml so it wasn't getting included. Thanks. On Mon, 2010-08-23 at 16:10 -0500, Brian West wrote: > This should clue you in... the gateway doesn't exist. > > /b > > On Aug 23, 2010, at 4:04 PM, Cliff Wells wrote: > > > 2010-08-23 08:52:05.918700 [ERR] mod_sofia.c:3668 Invalid Gateway > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- From david.ponzone at ipeva.fr Mon Aug 23 15:34:54 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Tue, 24 Aug 2010 00:34:54 +0200 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> Message-ID: <54AB9D1D-BFE7-4685-B098-ECB3806C3666@ipeva.fr> You should do that in your firewall. The quicker you filter, the better. I would not care much about the RTP traffic. So you need to filter SIP. And I would really don't think Vitelity is going to change the IP of their softswitch/SBC very often, and if they do, they should tell you. If Vitelity's IP is X and your SIP port is 5060, what you should do as filters is: allow UDP from X to yourIP:5060 (this will match SIP packets coming from Vitelity) deny UDP from all to yourIP:5060 (this will match malicious SIP packets) allow UDP from all to all (this will match the RTP traffic and other UDP traffic) and then add your other usual filters David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 23/08/2010 ? 23:11, Malay Thakershi a ?crit : > That is true. So do I block all other IP in my firewall? Or do I > configure that in FreeSwitch? Also, How can be sure my provider's IP > to remain same? (I use vitelity) > > Please let me know. > > On Mon, Aug 23, 2010 at 3:03 PM, David Ponzone > wrote: > If I understand correctly, you expect calls form PSTN, so only from > the known IPs of your provider ? > You can then filter all other IPs going to your port X (5060, 5080, > your mileage may vary). > > Also, a call coming to a port you don't use (so not opened) should > not have ANY impact. > It should not even hit the dialplan. > it should be rejected with ICMP port unreachable by the Windows TCP/ > IP stack. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 23/08/2010 ? 21:47, Malay Thakershi a ?crit : > >> I am going through documentation but seems iptables can eliminate >> calls being made on ports other than required ones. >> >> But my server is Windows. How do I run iptables command? >> >> Also, could you tell me if I block all incoming ports other than >> 5060 and 5061, will my regular inbound calls work? >> >> Thank you. >> >> >> >> 2010/8/23 Brian West >> David, >> No Clue, Never Used It, Can't Say... >> >> /b >> >> On Aug 23, 2010, at 2:32 PM, David Ponzone wrote: >> >> > Brian >> > >> > he can't add an ACL with FreePBX ? >> > >> > David Ponzone Direction Technique >> > email: david.ponzone at ipeva.fr >> > tel: 01 74 03 18 97 >> > gsm: 06 66 98 76 34 >> > >> > Service Client IPeva >> > tel: 0811 46 26 26 >> > www.ipeva.fr - www.ipeva-studio.com >> > >> > Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? l'intention exclusive de ses destinataires. Toute >> utilisation ou diffusion non autoris?e est interdite. Tout message >> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute >> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou >> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de >> le d?truire imm?diatement et d'avertir l'exp?diteur. >> > >> > >> > >> > >> > Le 23/08/2010 ? 21:26, Brian West a ?crit : >> > >> >> Well you're using FreePBX right? The only corse of action you >> have is to find out why its crashing and reporting the issue on our >> Jira. Without any more info to go on you're SOL. >> >> >> >> http://www.google.com/search?hl=en&client=safari&rls=en&defl=en&q=define:Vishing&sa=X&ei=RstyTO24JI_Znge7-6yNCw&ved=0CBIQkAE >> >> >> >> /b >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/d592baaa/attachment-0001.html From gmaruzz at celliax.org Mon Aug 23 15:36:42 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 24 Aug 2010 00:36:42 +0200 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Hi Jason, with 3) - the problematic one - you mean an incoming call from a remote skype to the fs-skype, that is then bridged to an outbound sip call? I'm asking this because this is the most tested case, eg: in the gateway for the FS-conference, where the incoming skype calls are then bridged (through dialplan, like pressing "1" in the standard IVR) with the sip conference. And it works with no problems... Let me know more, please. -giovanni On Mon, Aug 23, 2010 at 8:28 PM, Jason Jeffords wrote: > Hi Giovanni, > We got some test cycles today and have found the following: > 1) SIP to Skype calls work as desired - A-leg is SIP, B-leg is Skype > 2) Skype to Internal Extensions Work as desired - A-leg is Skype, B-leg is > FreeSwitch (internal extension) > 3) Skype to SIP does not work - A-leg is Skype, B-leg is SIP > In the last case: > 1) We hear Skype ringing until the B-leg (SIP) is answered - this is correct > 2) We then get dead air and the SIP B-leg hangs up on its own (i.e. the user > answering does not hang up, but the connection hangs up) > 3) The Skype client connection stays up with dead air until it is manually > hung up > We are running the latest code as of today. > Any insight you could provide will be most appreciated. > Thanks of all your work, > Jason > On Fri, Aug 13, 2010 at 5:31 PM, Giovanni Maruzzelli > wrote: >> >> Jason, >> >> please test with the latest git version. >> >> Many messy changes to an already messy code, so maybe some bug or side >> effect (ok, bug) has been introduced. >> >> commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 >> Author: Giovanni Maruzzelli >> Date: ? Fri Aug 13 16:19:20 2010 -0500 >> >> ? ?skypopen: now answer a call only when directed to do it (before >> was trying to answer any incoming call). Lot of changes to a messy >> part, so maybe some problem will come out... >> >> ? ?Signed-off-by: Giovanni Maruzzelli >> >> -giovanni >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Mon Aug 23 15:38:24 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 24 Aug 2010 00:38:24 +0200 Subject: [Freeswitch-users] FreeSwitch - Skypopen cant make call In-Reply-To: References: Message-ID: Good! -giovanni On Mon, Aug 23, 2010 at 9:30 PM, Shamun toha md wrote: > No Sorry, This is solved just by updating the idiot Skype Client from 2.1.x > to 2.0.72. > > Thanks > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From msc at freeswitch.org Mon Aug 23 15:50:16 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Aug 2010 15:50:16 -0700 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT In-Reply-To: <1282601115.3516.93.camel@portable-evil> References: <1282597451.3516.54.camel@portable-evil> <64247BA4-F7FC-4AF5-91DC-4EF03D5608AC@freeswitch.org> <1282601115.3516.93.camel@portable-evil> Message-ID: On Mon, Aug 23, 2010 at 3:05 PM, Cliff Wells wrote: > Yeah, I'm an idiot. > > "Admitting you have a problem is always the first step to recovery." ;) Good job on figuring out the issue. Thanks for letting us know the resolution. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/06bd01a8/attachment.html From steveayre at gmail.com Mon Aug 23 15:52:14 2010 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 23 Aug 2010 23:52:14 +0100 Subject: [Freeswitch-users] Newbie question In-Reply-To: <4C729A42.3060201@secnap.com> References: <4C729A42.3060201@secnap.com> Message-ID: Profiles are bound to IP + port, not just port. You could have an IP for internal and a different IP for external, and then use 5060 on both. That can either be either 2xLAN, 2xWAN, or 1LAN+1WAN IPs, it doesn't really matter. Which IP calls go out from just depends which profile you use in the dialstring. -Steve On 23 August 2010 16:56, Michael Scheidell wrote: > looked at FAQ's first, just want to clarify. > Some background, looking at replacing a running sipx install due to > limitations on ITSP's > (conflicting ITSP and sipx limitations) > Biggest limitation I see is that sipx without an external SBC forces the > ITSP to send calls to port 5080. > (Big Sip trunk providers like ATT, Level3 and Verizon don't want to do > that. I actually understand why) > > in the firewall ports section it notes port 5060 for 'default' internal, > 5070 for default nat and 5080 for default external. > > I suppose the biggest question is can I set up freeswitch to listen for SIP > inbound trunk calls on port 5060 without loosing functionality > (with sipx, if I do this, I can't forward calls: > > > I don't want to interfere with normal sip: url inbound calls, so I would > expect (in an 'askerisk like' setup) to be able to receive sip calls to > internal extensions as well as SIP trunk calls to port 5060. (and be able > to put both on hold, take them off hold, play moh to callers, transfer, > blind and attended, and join a 3 way conference call) > > Next, looking for someone who has moved from sipx 4.2.0 to > freeswitch/fusionpbx to quote me on a conversion (assuming I can use the > above) > > POC would be to use someone like voip.ms and static registration (which > uses port 5060). and have full regression testing. inbound calls, outbound > calls, calls transfterd, (correct caller id shows up!) > > Currently, this only works on voip.ms user/password authentication since > the invite/auth exchange will register port 5080. > voip.ms has no way to specify the calling port, and even if they did, this > is just the POC. Level3, Att and Verizon will ONLY send to port 5060. > > (ps, ISN dialing. seems to be tied in to e.164 dialing, but I was not able > to actually tell if that is the case. > on sipx, I added a two digit 'trunk' code '**' to make outbound ISN calls) > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > sip:michael.scheidell at secnap.com > > *| *SECNAP Network Security Corporation > > - Certified SNORT Integrator > - 2008-9 Hot Company Award Winner, World Executive Alliance > - Five-Star Partner Program 2009, VARBusiness > - Best in Email Security,2010: Network Products Guide > - King of Spam Filters, SC Magazine 2008 > > > ------------------------------ > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see http://www.secnap.com/products/spammertrap/ > ------------------------------ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/5a04ff75/attachment.html From dujinfang at gmail.com Mon Aug 23 15:58:20 2010 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Aug 2010 06:58:20 +0800 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: Message-ID: Thanks all replies to this thread, I got more clear idea. On Mon, Aug 23, 2010 at 11:53 PM, Michael Collins wrote: > What happened when you created several thousand bridged channels as opposed > to 2-person conferences? Just curious to see where your upper limits came in > to play there. > -MC > MC, I'm not sure I get what do you mean. I think the fact is that conferences use more resources(threads) than bridges, it just failed to create new threads at about 2560 threads on Mac. I used the default ulimit values. Since I was trying to compare conferences over bridges but not load test, so I didn't care about much of upper limits. Anyway I will never use in production on Mac. On Linux, I created 2400 bridged channels(think about loopback using double than normal calls), and without hangup all the calls, I start to create 2400 conference channels, loadavg boost to over 2000 from some where, so I executed "hupall" to avoid crash the whole sever. ( before I saw 2000 loadavg, the max loadavg I seen is about 60 on linux and that server is no-response at all, I was lucky that I can still control the server on loadavg 2000. I almost used the default config(say loglevel 7), so bottle necks might be everywhere. Anyway I got my conclusion. Thanks. From Victor at isptelecom.net Mon Aug 23 16:00:09 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Mon, 23 Aug 2010 19:00:09 -0400 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <54AB9D1D-BFE7-4685-B098-ECB3806C3666@ipeva.fr> References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> <54AB9D1D-BFE7-4685-B098-ECB3806C3666@ipeva.fr> Message-ID: <4C72FD79.5030401@isptelecom.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/3ae6334f/attachment-0001.html From anthony.minessale at gmail.com Mon Aug 23 16:27:52 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Aug 2010 18:27:52 -0500 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: Message-ID: Mac indeed has some unfair hard-coded 2500 thread limit that we can't explain, we've seen it all along. Don't trust the load avg on linux when you have lots of rtp, it's not real its thrown off because of all the rtp traffic causing scheduler wait. anyway, you learned the limits of your box for yourself, good job! That's what I like to see. naturally, you get better or worse results depending on how many cpu you have or how fast each one is. On Mon, Aug 23, 2010 at 5:58 PM, Seven Du wrote: > Thanks all replies to this thread, I got more clear idea. > > On Mon, Aug 23, 2010 at 11:53 PM, Michael Collins wrote: >> What happened when you created several thousand bridged channels as opposed >> to 2-person conferences? Just curious to see where your upper limits came in >> to play there. >> -MC >> > > MC, I'm not sure I get what do you mean. I think the fact is that > conferences use more resources(threads) than bridges, it just failed > to create new threads at about 2560 threads on Mac. I used the default > ulimit values. ?Since I was trying to compare conferences over bridges > but not load test, so I didn't care about much of upper limits. Anyway > I will never use in production on Mac. > > On Linux, I created 2400 bridged channels(think about loopback using > double than normal calls), and without hangup all the calls, I start > to create 2400 conference channels, loadavg boost to over 2000 from > some where, so I executed "hupall" to avoid crash the whole sever. ( > before I saw 2000 loadavg, the max loadavg I seen is about 60 on linux > and that server is no-response at all, I was lucky that I can still > control the server on loadavg 2000. > > I almost used the default config(say loglevel 7), so bottle necks > might be everywhere. Anyway I got my conclusion. > > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From brian at freeswitch.org Mon Aug 23 16:34:35 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Aug 2010 18:34:35 -0500 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: Message-ID: <059F5E02-AD34-4C1B-B96C-DE742933085D@freeswitch.org> Sad part is that 2560 thread limit on OS X is still there... and they sell a freakin 12 core Mac Pro now.. I have a bug open on apples bug tracker over this issue for like two years now. /b On Aug 23, 2010, at 6:27 PM, Anthony Minessale wrote: > Mac indeed has some unfair hard-coded 2500 thread limit that we can't > explain, we've seen it all along. > > Don't trust the load avg on linux when you have lots of rtp, it's not > real its thrown off because of all the rtp traffic causing scheduler > wait. > > anyway, you learned the limits of your box for yourself, good job! > That's what I like to see. > > naturally, you get better or worse results depending on how many cpu > you have or how fast each one is. From jason at cloudtree.net Mon Aug 23 16:34:26 2010 From: jason at cloudtree.net (Jason Jeffords) Date: Mon, 23 Aug 2010 19:34:26 -0400 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Hi Giovanni, Our case 3 test was actually being bridged to a SIP endpoint (not in a conference, although this probably should not matter). We tested two type 3 cases: 1) Skype to FreeSWITCH Skype bridged to an extension of a registered SIP phone 2) Skype to FreeSWITCH Skype bridged to an outbound call through a PSTN gateway In both cases we are transiting FreeSWITCH, not terminating on it (could there be a race condition when doing signaling coordination to remote SIP endpoints, not FreeSWITCH itself?). When we terminate Skype calls on FreeSWITCH this works (case 2). It also works for outbound Skype calls (case1 - SIP to FreeSWITCH, FreeSWITCH Skype to Skype). Also, we are running the very latest (well, as of this morning) git version, so that may introduce one more variable (if you are running an older version). Thanks for your help, Jason On Mon, Aug 23, 2010 at 6:36 PM, Giovanni Maruzzelli wrote: > Hi Jason, > > with 3) - the problematic one - you mean an incoming call from a > remote skype to the fs-skype, that is then bridged to an outbound sip > call? > > I'm asking this because this is the most tested case, eg: in the > gateway for the FS-conference, where the incoming skype calls are then > bridged (through dialplan, like pressing "1" in the standard IVR) with > the sip conference. And it works with no problems... > > Let me know more, please. > > -giovanni > > On Mon, Aug 23, 2010 at 8:28 PM, Jason Jeffords > wrote: > > Hi Giovanni, > > We got some test cycles today and have found the following: > > 1) SIP to Skype calls work as desired - A-leg is SIP, B-leg is Skype > > 2) Skype to Internal Extensions Work as desired - A-leg is Skype, B-leg > is > > FreeSwitch (internal extension) > > 3) Skype to SIP does not work - A-leg is Skype, B-leg is SIP > > In the last case: > > 1) We hear Skype ringing until the B-leg (SIP) is answered - this is > correct > > 2) We then get dead air and the SIP B-leg hangs up on its own (i.e. the > user > > answering does not hang up, but the connection hangs up) > > 3) The Skype client connection stays up with dead air until it is > manually > > hung up > > We are running the latest code as of today. > > Any insight you could provide will be most appreciated. > > Thanks of all your work, > > Jason > > On Fri, Aug 13, 2010 at 5:31 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> Jason, > >> > >> please test with the latest git version. > >> > >> Many messy changes to an already messy code, so maybe some bug or side > >> effect (ok, bug) has been introduced. > >> > >> commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 > >> Author: Giovanni Maruzzelli > >> Date: Fri Aug 13 16:19:20 2010 -0500 > >> > >> skypopen: now answer a call only when directed to do it (before > >> was trying to answer any incoming call). Lot of changes to a messy > >> part, so maybe some problem will come out... > >> > >> Signed-off-by: Giovanni Maruzzelli > >> > >> -giovanni > >> -- > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/ede2cb50/attachment.html From anthony.minessale at gmail.com Mon Aug 23 16:41:40 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Aug 2010 18:41:40 -0500 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: We never have had PHP in our code distribution. On Mon, Aug 23, 2010 at 4:39 PM, jesse wrote: > The current FS version doesn't include source code for PHP. > What are the steps to install it on top of my current version? ?where > to get the source? how to build and install? > > Iif you guys want to keep the new system slim by get rid of PHP, > please at least keep a doc about how to add it in case of need. > > thanks! > > -jesse > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From gmaruzz at celliax.org Mon Aug 23 16:46:36 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 24 Aug 2010 01:46:36 +0200 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Hi Jason, I repetedly tested the scenario: Skype->skypopen->FS->SIP->FS and this mens, a skype call incoming to FS and then bridged via SIP to another FS you can test the same on your machine if you make the destination for skypopen interface pointing to the extension 5000 of the default dialplan (the standard IVR) and then press 1. Can you please test the 5000 ext press 1, and confirm that it works (or not) for you? Can you then post (better in a Jira) the exact way to replicate your problem? (dialplan, actions, configs, whatever) Thanks for reporting, -giovanni On Tue, Aug 24, 2010 at 1:34 AM, Jason Jeffords wrote: > Hi Giovanni, > Our case 3 test was actually being bridged to a SIP endpoint (not in a > conference, although > this probably should not matter). > We tested two type 3 cases: > 1) Skype to FreeSWITCH Skype bridged to an extension of a registered SIP > phone > 2) Skype to FreeSWITCH Skype bridged to an outbound call through a PSTN > gateway > In both cases we are transiting FreeSWITCH, not terminating on it (could > there be a > race condition when doing signaling coordination to remote SIP endpoints, > not FreeSWITCH > itself?). > When we terminate Skype calls on FreeSWITCH this works (case 2). ?It also > works > for outbound Skype calls (case1 - SIP to FreeSWITCH, FreeSWITCH Skype to > Skype). > Also, we are running the very latest (well, as of this morning) git version, > so that > may introduce one more variable (if you are running an older version). > Thanks for your help, > Jason > > On Mon, Aug 23, 2010 at 6:36 PM, Giovanni Maruzzelli > wrote: >> >> Hi Jason, >> >> with 3) - the problematic one - you mean an incoming call from a >> remote skype to the fs-skype, that is then bridged to an outbound sip >> call? >> >> I'm asking this because this is the most tested case, eg: in the >> gateway for the FS-conference, where the incoming skype calls are then >> bridged (through dialplan, like pressing "1" in the standard IVR) with >> the sip conference. And it works with no problems... >> >> Let me know more, please. >> >> -giovanni >> >> On Mon, Aug 23, 2010 at 8:28 PM, Jason Jeffords >> wrote: >> > Hi Giovanni, >> > We got some test cycles today and have found the following: >> > 1) SIP to Skype calls work as desired - A-leg is SIP, B-leg is Skype >> > 2) Skype to Internal Extensions Work as desired - A-leg is Skype, B-leg >> > is >> > FreeSwitch (internal extension) >> > 3) Skype to SIP does not work - A-leg is Skype, B-leg is SIP >> > In the last case: >> > 1) We hear Skype ringing until the B-leg (SIP) is answered - this is >> > correct >> > 2) We then get dead air and the SIP B-leg hangs up on its own (i.e. the >> > user >> > answering does not hang up, but the connection hangs up) >> > 3) The Skype client connection stays up with dead air until it is >> > manually >> > hung up >> > We are running the latest code as of today. >> > Any insight you could provide will be most appreciated. >> > Thanks of all your work, >> > Jason >> > On Fri, Aug 13, 2010 at 5:31 PM, Giovanni Maruzzelli >> > >> > wrote: >> >> >> >> Jason, >> >> >> >> please test with the latest git version. >> >> >> >> Many messy changes to an already messy code, so maybe some bug or side >> >> effect (ok, bug) has been introduced. >> >> >> >> commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 >> >> Author: Giovanni Maruzzelli >> >> Date: ? Fri Aug 13 16:19:20 2010 -0500 >> >> >> >> ? ?skypopen: now answer a call only when directed to do it (before >> >> was trying to answer any incoming call). Lot of changes to a messy >> >> part, so maybe some problem will come out... >> >> >> >> ? ?Signed-off-by: Giovanni Maruzzelli >> >> >> >> -giovanni >> >> -- >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> Cell : +39-347-2665618 >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From mrene_lists at avgs.ca Mon Aug 23 16:52:20 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 23 Aug 2010 19:52:20 -0400 Subject: [Freeswitch-users] Account selection In-Reply-To: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> Message-ID: <13158305-434E-475D-A2DA-D0AEEB546755@avgs.ca> Hi, You can set the current account using the "set_user" dialplan application. You could define a list of prefixes to select which account you use when dialing outbound. For example: And then in your dialplan Once set_user runs, it loads in everything from the user directory, like if the call came from that user. You can now dial 81[number] to dial the number using the phone's line 1, etc. This is a good example of how one can be creative in their ways of configuring freeswitch :) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-08-23, at 3:57 AM, Ken Gillett wrote: > If one wishes to have use of several VOIP 'lines', but with no PBX, you need to register all those SIP accounts with the client (softphone etc). You should then be informed which account is receiving a call and can pick a particular account from which to make calls. Once a PBX is in use, you can register the client as a single extension of the PBX and direct calls as appropriate to that extension - I assume with the correct caller ID and incoming account information passed to the recipient so they know as much as in the 'no PBX' configuration. > > But what about outgoing calls. In this scenario, registered as a single extension, how would it be possible to pick the outgoing 'line' (i.e. account) to use? Would it have to be done by dialling a prefix or is there another way? Is it client dependent? > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jim at k4gvo.com Mon Aug 23 16:55:43 2010 From: jim at k4gvo.com (Jim) Date: Mon, 23 Aug 2010 19:55:43 -0400 Subject: [Freeswitch-users] Still can't dial gateway from ZAP phone. Message-ID: <4C730A7F.6030408@k4gvo.com> Michael Collins gave me a suggestions a while back: I looked in mod_openzap.c and I didn't see any references to channel variables. However, you have context and dialplan options. I suggest that you create a dialplan context just for your FXS port(s). Try this. Create conf/dialplan/fxs-ports.xml: Then in your openzap.conf.xml change the context for the analog span(s) with the FXS ports: Restart FS after making these changes and then give it a shot. You should see the call from the analog phone going into context "fxs-ports" and then get transferred over to the default context where it will act like your SIP phones because we manually set the ${toll_allow} chan var. -MC Unfortunately that did not work. The "default_gateway" variable used by this line: Dialplan: OpenZAP/1:1/17705550068 Action bridge(sofia/gateway/${default_gateway}/17705550068) ended up looking like: EXECUTE OpenZAP/1:1/17705550068 bridge(sofia/gateway//17705550068) whereas a successful dialout looks like: EXECUTE sofia/internal/1002 at 192.168.2.51 bridge(sofia/gateway/gw4.telasip.com/17705550068) Somehow the information in the directory/default/default.xml file never got included and I'm not sure how to fix it. Thanks for any guidance. Jim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/813e3f55/attachment.html From dujinfang at gmail.com Mon Aug 23 17:00:54 2010 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Aug 2010 08:00:54 +0800 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: <059F5E02-AD34-4C1B-B96C-DE742933085D@freeswitch.org> References: <059F5E02-AD34-4C1B-B96C-DE742933085D@freeswitch.org> Message-ID: Thanks Anthony & Brian. I learned more than the limits of my box in this thread :). Hard to believe it is hard coded. I did google on this subject but got no answer before changed my test case to generate less threads. Even harder to believe it still has this limit on a 12 core. So, I guess no one is running (or will run) in production on Mac. But Mac is a good platform for developers and we still can do fancy things on it like fscomm that's why FS support Mac isn't it? On Tue, Aug 24, 2010 at 7:34 AM, Brian West wrote: > Sad part is that 2560 thread limit on OS X is still there... and they sell a freakin 12 core Mac Pro now.. ?I have a bug open on apples bug tracker over this issue for like two years now. > > /b > > On Aug 23, 2010, at 6:27 PM, Anthony Minessale wrote: > >> Mac indeed has some unfair hard-coded 2500 thread limit that we can't >> explain, we've seen it all along. >> >> Don't trust the load avg on linux when you have lots of rtp, it's not >> real its thrown off because of all the rtp traffic causing scheduler >> wait. >> >> anyway, you learned the limits of your box for yourself, good job! >> That's what I like to see. >> >> naturally, you get better or worse results depending on how many cpu >> you have or how fast each one is. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From mthakershi at gmail.com Mon Aug 23 17:01:08 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Mon, 23 Aug 2010 19:01:08 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <4C72FD79.5030401@isptelecom.net> References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> <54AB9D1D-BFE7-4685-B098-ECB3806C3666@ipeva.fr> <4C72FD79.5030401@isptelecom.net> Message-ID: Thank you. Could you please share your configuration section from the two files? I tried what you suggested. I put another IP as my ACL (which should be rejected) but it goes through. So 41.XXX is not the IP I am calling from but it connects me anyway. acl.conf.xml ----------------- ------------- Base_Settings.xml file under sip_profiles folder: ----------------- Please let me know what am I doing wrong or missing? Thank you. On Mon, Aug 23, 2010 at 6:00 PM, Victor Chukalovskiy wrote: > Malay, > > I use apply-inbound-acl="providers" in my sip profile. Then I define my > providers IP addresses in ACL "providers" (within acl.conf.xml) > This way all other IPs are forced to authorize in order to place calls > through. > Why bother with firewall if freeswitch has built-in ACL functionality? > > Regards, > Victor > > > On -10/01/37 02:59 PM, David Ponzone wrote: > > You should do that in your firewall. > The quicker you filter, the better. > > I would not care much about the RTP traffic. > So you need to filter SIP. > And I would really don't think Vitelity is going to change the IP of their > softswitch/SBC very often, and if they do, they should tell you. > > If Vitelity's IP is X and your SIP port is 5060, what you should do as > filters is: > allow UDP from X to yourIP:5060 (this will match SIP packets coming from > Vitelity) > deny UDP from all to yourIP:5060 (this will match malicious SIP packets) > allow UDP from all to all (this will match the RTP traffic and other UDP > traffic) > and then add your other usual filters > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 23/08/2010 ? 23:11, Malay Thakershi a ?crit : > > That is true. So do I block all other IP in my firewall? Or do I configure > that in FreeSwitch? Also, How can be sure my provider's IP to remain same? > (I use vitelity) > > Please let me know. > > On Mon, Aug 23, 2010 at 3:03 PM, David Ponzone wrote: > >> If I understand correctly, you expect calls form PSTN, so only from the >> known IPs of your provider ? >> You can then filter all other IPs going to your port X (5060, 5080, your >> mileage may vary). >> >> Also, a call coming to a port you don't use (so not opened) should not >> have ANY impact. >> It should not even hit the dialplan. >> it should be rejected with ICMP port unreachable by the Windows TCP/IP >> stack. >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis >> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 23/08/2010 ? 21:47, Malay Thakershi a ?crit : >> >> I am going through documentation but seems iptables can eliminate calls >> being made on ports other than required ones. >> >> But my server is Windows. How do I run iptables command? >> >> Also, could you tell me if I block all incoming ports other than 5060 >> and 5061, will my regular inbound calls work? >> >> Thank you. >> >> >> >> 2010/8/23 Brian West >> >>> David, >>> No Clue, Never Used It, Can't Say... >>> >>> /b >>> >>> On Aug 23, 2010, at 2:32 PM, David Ponzone wrote: >>> >>> > Brian >>> > >>> > he can't add an ACL with FreePBX ? >>> > >>> > David Ponzone Direction Technique >>> > email: david.ponzone at ipeva.fr >>> > tel: 01 74 03 18 97 >>> > gsm: 06 66 98 76 34 >>> > >>> > Service Client IPeva >>> > tel: 0811 46 26 26 >>> > www.ipeva.fr - www.ipeva-studio.com >>> > >>> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il >>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce >>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. >>> > >>> > >>> > >>> > >>> > Le 23/08/2010 ? 21:26, Brian West a ?crit : >>> > >>> >> Well you're using FreePBX right? The only corse of action you have is >>> to find out why its crashing and reporting the issue on our Jira. Without >>> any more info to go on you're SOL. >>> >> >>> >> >>> http://www.google.com/search?hl=en&client=safari&rls=en&defl=en&q=define:Vishing&sa=X&ei=RstyTO24JI_Znge7-6yNCw&ved=0CBIQkAE >>> >> >>> >> /b >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/84802ec9/attachment-0001.html From david.ponzone at ipeva.fr Mon Aug 23 17:12:10 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Tue, 24 Aug 2010 02:12:10 +0200 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: References: <059F5E02-AD34-4C1B-B96C-DE742933085D@freeswitch.org> Message-ID: <48487B04-981E-40A8-BEA1-BC8011183980@ipeva.fr> Seven, is that on a regular workstation Mac or on a XServe. If the limitation is still there on a XServe, that's insane! David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 24/08/2010 ? 02:00, Seven Du a ?crit : > Thanks Anthony & Brian. I learned more than the limits of my box in > this thread :). Hard to believe it is hard coded. I did google on this > subject but got no answer before changed my test case to generate less > threads. > > Even harder to believe it still has this limit on a 12 core. So, I > guess no one is running (or will run) in production on Mac. But Mac is > a good platform for developers and we still can do fancy things on it > like fscomm that's why FS support Mac isn't it? > > On Tue, Aug 24, 2010 at 7:34 AM, Brian West > wrote: >> Sad part is that 2560 thread limit on OS X is still there... and >> they sell a freakin 12 core Mac Pro now.. I have a bug open on >> apples bug tracker over this issue for like two years now. >> >> /b >> >> On Aug 23, 2010, at 6:27 PM, Anthony Minessale wrote: >> >>> Mac indeed has some unfair hard-coded 2500 thread limit that we >>> can't >>> explain, we've seen it all along. >>> >>> Don't trust the load avg on linux when you have lots of rtp, it's >>> not >>> real its thrown off because of all the rtp traffic causing scheduler >>> wait. >>> >>> anyway, you learned the limits of your box for yourself, good job! >>> That's what I like to see. >>> >>> naturally, you get better or worse results depending on how many cpu >>> you have or how fast each one is. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/557c0f3b/attachment.html From dujinfang at gmail.com Mon Aug 23 17:13:45 2010 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Aug 2010 08:13:45 +0800 Subject: [Freeswitch-users] Account selection In-Reply-To: <13158305-434E-475D-A2DA-D0AEEB546755@avgs.ca> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <13158305-434E-475D-A2DA-D0AEEB546755@avgs.ca> Message-ID: Mathieu, It is very good but it's not the right answer to the question. Ken asked "Is there any way to do this without having to use Dial plans?" ;) On Tue, Aug 24, 2010 at 7:52 AM, Mathieu Rene wrote: > Hi, > > You can set the current account using the "set_user" dialplan application. > > You could define a list of prefixes to select which account you use when dialing outbound. > > For example: > > > ? > ? > ? > ? > > > And then in your dialplan > > ? > ? > ? > ? > > > Once set_user runs, it loads in everything from the user directory, like if the call came from that user. > You can now dial 81[number] to dial the number using the phone's line 1, etc. > > This is a good example of how one can be creative in their ways of configuring freeswitch :) > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 2010-08-23, at 3:57 AM, Ken Gillett wrote: > >> If one wishes to have use of several VOIP 'lines', but with no PBX, you need to register all those SIP accounts with the client (softphone etc). You should then be informed which account is receiving a call and can pick a particular account from which to make calls. Once a PBX is in use, you can register the client as a single extension of the PBX and direct calls as appropriate to that extension - I assume with the correct caller ID and incoming account information passed to the recipient so they know as much as in the 'no PBX' configuration. >> >> But what about outgoing calls. In this scenario, registered as a single extension, how would it be possible to pick the outgoing 'line' (i.e. account) to use? Would it have to be done by dialling a prefix or is there another way? Is it client dependent? >> >> >> Ken G i l l e t t >> >> _/_/_/_/_/_/_/_/ >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From Victor at isptelecom.net Mon Aug 23 18:24:48 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Mon, 23 Aug 2010 21:24:48 -0400 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> <54AB9D1D-BFE7-4685-B098-ECB3806C3666@ipeva.fr> <4C72FD79.5030401@isptelecom.net> Message-ID: <4C731F60.6030801@isptelecom.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/240b0241/attachment-0001.html From dujinfang at gmail.com Mon Aug 23 18:54:34 2010 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Aug 2010 09:54:34 +0800 Subject: [Freeswitch-users] performance between bridged call and conference In-Reply-To: <48487B04-981E-40A8-BEA1-BC8011183980@ipeva.fr> References: <059F5E02-AD34-4C1B-B96C-DE742933085D@freeswitch.org> <48487B04-981E-40A8-BEA1-BC8011183980@ipeva.fr> Message-ID: It was a Mac book pro. On Tue, Aug 24, 2010 at 8:12 AM, David Ponzone wrote: > Seven, > is that on a regular workstation Mac or on a XServe. > If the limitation is still there on a XServe, that's insane! > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 24/08/2010 ? 02:00, Seven Du a ?crit : > > Thanks Anthony & Brian. I learned more than the limits of my box in > this thread :). Hard to believe it is hard coded. I did google on this > subject but got no answer before changed my test case to generate less > threads. > > Even harder to believe it still has this limit on a 12 core. So, I > guess no one is running (or will run) in production on Mac. But Mac is > a good platform for developers and we still can do fancy things on it > like fscomm that's why FS support Mac isn't it? > > On Tue, Aug 24, 2010 at 7:34 AM, Brian West wrote: > > Sad part is that 2560 thread limit on OS X is still there... and they sell a > freakin 12 core Mac Pro now.. ?I have a bug open on apples bug tracker over > this issue for like two years now. > > /b > > On Aug 23, 2010, at 6:27 PM, Anthony Minessale wrote: > > Mac indeed has some unfair hard-coded 2500 thread limit that we can't > > explain, we've seen it all along. > > Don't trust the load avg on linux when you have lots of rtp, it's not > > real its thrown off because of all the rtp traffic causing scheduler > > wait. > > anyway, you learned the limits of your box for yourself, good job! > > That's what I like to see. > > naturally, you get better or worse results depending on how many cpu > > you have or how fast each one is. > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Blog: http://www.dujinfang.com > Proj:? http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From macedoslm at gmail.com Mon Aug 23 19:08:47 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Mon, 23 Aug 2010 23:08:47 -0300 Subject: [Freeswitch-users] Suppress DTMF Message-ID: Hi, I want to suppress the dtmf tone in the conference room. When a user sends a DTMF in the conference all others users can hear that DTMF tone. How can I suppress it? My = "waste". Thanks, -- Samuel Macedo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100823/179184cb/attachment.html From kris at kriskinc.com Mon Aug 23 21:52:28 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 24 Aug 2010 00:52:28 -0400 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation Message-ID: Hello everyone, While extensive tweaks to Mod_voicemail and phrases can get "pretty close" to emulating existing voicemail systems like Asterisk, "pretty close" doesn't cut it when you have many discerning users that have become accustomed to the behavior of a given system. I want to move to FreeSWITCH voicemail but I don't want to have to reprint quick reference cards and retrain users and support staff. I want to make FreeSWITCH mod_voicemail even better by expanding its already impressive customization. A few tweaks here, and few new features there and we should be able to do drop in replacements for Asterisk and presumably other voicemail systems. Official bounty here: http://jira.freeswitch.org/browse/BOUNTY-22 You'll notice I haven't included a dollar/euro/pound amount... Well it's because I'm not even sure of the scope of this work. I'm hoping to get some discussion going first (of course any work we pay for will be open source and given back). So, what are your thoughts? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From adminjew at gmail.com Mon Aug 23 22:45:28 2010 From: adminjew at gmail.com (Yitzchok) Date: Tue, 24 Aug 2010 01:45:28 -0400 Subject: [Freeswitch-users] Phrase speak-text function return on first key press in phrase file on Windows In-Reply-To: <1282597277910-5454286.post@n2.nabble.com> References: <1282530102895-5451221.post@n2.nabble.com> <1282597277910-5454286.post@n2.nabble.com> Message-ID: Thanks Jeff seems like it works now. Yitzchok On Mon, Aug 23, 2010 at 5:01 PM, Jeff Lenk wrote: > > Fixed in git head > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Phrase-speak-text-function-return-on-first-key-press-in-phrase-file-on-Windows-tp5449059p5454286.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/eb7ce400/attachment.html From shamun.toha at gmail.com Tue Aug 24 00:17:22 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Tue, 24 Aug 2010 09:17:22 +0200 Subject: [Freeswitch-users] FreeSwitch - uuid find list dump Message-ID: Hi, i have few questions, don't know why its not working. Q. How do i get the uuid? To make a test from the CLI. freeswitch at example> sofia profile internal siptrace on Enabled sip debugging on internal freeswitch at example> uuid_exists 3f7f5dc2-af4f-11df-b704-69c5ea8291c5 false freeswitch at example> Q. When i register SIP phone, nonce = uuid? Authorization: Digest username="2001", realm="78.23.89.55", nonce="3f7f5dc2-af4f-11df-b704-69c5ea8291c5", uri="sip:78.23.89.55", algorithm=MD5, qop=auth, cnonce="339774495957948623487dc37177be3f", nc=00000001, response="8fa80654a0c7bfba71c81f5e9cbafee6" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/cccc7d75/attachment.html From david.ponzone at ipeva.fr Tue Aug 24 00:30:39 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Tue, 24 Aug 2010 09:30:39 +0200 Subject: [Freeswitch-users] FreeSwitch - uuid find list dump In-Reply-To: References: Message-ID: <148EEEEF-F9D6-470A-8A60-5D58F92BB11A@ipeva.fr> show calls or show channels David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 24/08/2010 ? 09:17, Shamun toha md a ?crit : > Hi, i have few questions, don't know why its not working. > > Q. How do i get the uuid? To make a test from the CLI. > > freeswitch at example> sofia profile internal siptrace on > > Enabled sip debugging on internal > freeswitch at example> uuid_exists 3f7f5dc2-af4f-11df-b704-69c5ea8291c5 > > false > freeswitch at example> > > > Q. When i register SIP phone, nonce = uuid? > Authorization: Digest username="2001", realm="78.23.89.55", > nonce="3f7f5dc2-af4f-11df-b704-69c5ea8291c5", uri="sip:78.23.89.55", > algorithm=MD5, qop=auth, cnonce="339774495957948623487dc37177be3f", > nc=00000001, response="8fa80654a0c7bfba71c81f5e9cbafee6" > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/468e16b7/attachment.html From shamun.toha at gmail.com Tue Aug 24 00:44:05 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Tue, 24 Aug 2010 09:44:05 +0200 Subject: [Freeswitch-users] FreeSwitch - uuid find list dump In-Reply-To: <148EEEEF-F9D6-470A-8A60-5D58F92BB11A@ipeva.fr> References: <148EEEEF-F9D6-470A-8A60-5D58F92BB11A@ipeva.fr> Message-ID: Thank you, works. freeswitch at example> show calls 0 total. freeswitch at example> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data 02eebd72-af53-11df-b708-69c5ea8291c5,inbound,2010-08-24 09:41:31,1282635691,sofia/internal/2001 at 78.23.89.xxx ,CS_EXECUTE,2001,2001,78.23.89.xxx,5000,ivr,demo_ivr,XML,default,PCMA,8000,PCMA,8000,,example,2001 at 78.23.89.xxx , 1 total. freeswitch at example> uuid_exists 02eebd72-af53-11df-b708-69c5ea8291c5 true freeswitch at example> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/72dd23c3/attachment-0001.html From ken at ukgb.net Tue Aug 24 00:44:14 2010 From: ken at ukgb.net (Ken Gillett) Date: Tue, 24 Aug 2010 08:44:14 +0100 Subject: [Freeswitch-users] Account selection In-Reply-To: References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> <2DD257A8-A956-45F1-9F3C-40A6EE3E77D3@ipeva.fr> <773D1529-5F42-4C53-BE5C-7A2C1B0546B6@ukgb.net> Message-ID: <869311D3-5342-44CD-B7EB-EC0435C564EE@ukgb.net> On 23 Aug 2010, at 20:57, David Ponzone wrote: > > Was this helpful ? Yes indeed. Although I don't want dialled prefixes to be the only way to choose an outbound account, I do want it to be possible, so thanks to everyone for all the replies. Ken G i l l e t t _/_/_/_/_/_/_/_/ From asilva at wirelessmundi.com Tue Aug 24 00:56:46 2010 From: asilva at wirelessmundi.com (Antonio) Date: Tue, 24 Aug 2010 09:56:46 +0200 Subject: [Freeswitch-users] core.db table channels application field store wrong data when executing voicemail In-Reply-To: References: <1282205390.25391.41.camel@marces.tc.commsmundi.com> <1282555930.25391.48.camel@marces.tc.commsmundi.com> Message-ID: <1282636606.13028.7.camel@marces.tc.commsmundi.com> Yes, i want to use it for realtime monitoring. I will give a try to ESL. My system is not that load, so in the core.db i would have every thing that i need, but i can't trust the fields... using ESL i will "re-do" what is been done fore the core.db . On Mon, 2010-08-23 at 10:52 +0100, Steven Ayre wrote: > > > On 23 August 2010 10:32, Antonio wrote: > I see. I've a couple of questions related: > > Is it recommended to directly use core.db for monitoring > purposes > instead of developing a custom event handler? Could there be > any sqlite > "locking" issues if attacking that database in a read-only > manner? > > I would use ESL, I believe that's probably the recommended way to do > it. It's a pretty simple protocol so shouldn't be any harder to > implement than something reading an sqlite file. You can either use > events and work out the channel states yourself, or use api commands > like 'show channels' etc (which reads the core db so are pretty much > what you'd already be doing). > > A rwlock can't be written while a reader has it, so it will possibly > slow the DB down if you're reading the database. Shouldn't be > noticeable unless you're doing so intensively though. > > Another approach would be to move the core into odbc to use a database > system that's more advanced than sqlite, which would also them allow > you to replicate the data to another server so none of the monitoring > occurs on the FS host. > > > In case it is possible, how can I identify what was the > initial > application that was called from the dialplan (and not the > applications > called by it). Is there an API method containing the history > of > applications executed on a specific channel given its uuid? > > Do you need it in realtime or after the call? mod_xml_cdr contains the > callflow, I don't know whether that can be accessed during a call > though. > > ESL events will give you that information I believe. > > > Thanks, > Ant?nio > > > > On Thu, 2010-08-19 at 09:20 -0500, Anthony Minessale wrote: > > No, every app that is ever executed will change that field, > some apps > > in turn execute more apps. > > > > > > On Thu, Aug 19, 2010 at 3:09 AM, Antonio > wrote: > > > > > > Hi, > > > > > > during multiple selects in the table "channels" just > realized that when > > > a call is executing the app voicemail, to check for > messages, during the > > > navigation menu of the voicemail it changes to application > "sleep". > > > > > > Is it normal? or is a bug for fs? > > > > > > > > > The call: > > > > > > During the auth: > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 > 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25| > CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com > commsmundi.com 100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25|| > EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 > 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25| > CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com > commsmundi.com 100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25|| > EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 > 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25| > CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com > commsmundi.com 100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25|| > EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 > 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25| > CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com > commsmundi.com 100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25|| > EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 > 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25| > CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com > commsmundi.com 100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25|| > EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 > 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25| > CS_EXECUTE|101|101| > > > 192.168.10.75|133|voicemail|check commsmundi.com > commsmundi.com 100|XML| > > > local|PCMU|8000|PCMU|8000||marces|101 at 192.168.10.25|| > EARLY|||| > > > > > > in the navigation menu: > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 > 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25| > CS_EXECUTE|101|101| > > > 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU| > 8000||marces| > > > 101 at 192.168.10.25||EARLY|||| > > > > > > sqlite> select * from channels; > > > 58de6c5f-1d0a-45f6-9d0c-4665b32812eb|inbound|2010-08-19 > 10:01:46| > > > 1282204906|sofia/192.168.10.25/101 at 192.168.10.25| > CS_EXECUTE|101|101| > > > 192.168.10.75|133|sleep|100|XML|local|PCMU|8000|PCMU| > 8000||marces| > > > 101 at 192.168.10.25||EARLY|||| > > > > > > > > > > > > Thanks, > > > Ant?nio > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > -- > > > Un cordial saludo / Best regards, > > _________________________ > > Ant?nio Silva > > E-mail:asilva at wirelessmundi.com > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Un cordial saludo / Best regards, _________________________ Ant?nio Silva E-mail:asilva at wirelessmundi.com From steveayre at gmail.com Tue Aug 24 01:58:33 2010 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 24 Aug 2010 09:58:33 +0100 Subject: [Freeswitch-users] FreeSwitch - uuid find list dump In-Reply-To: References: Message-ID: The nonce is not the UUID... it's a value that's generated and only ever used once as part of the authentication process, and isn't related to the call UUID. As you've already heard show calls/channels is the way to go. -Steve On 24 August 2010 08:17, Shamun toha md wrote: > Hi, i have few questions, don't know why its not working. > > Q. How do i get the uuid? To make a test from the CLI. > > freeswitch at example> sofia profile internal siptrace on > > Enabled sip debugging on internal > freeswitch at example> uuid_exists 3f7f5dc2-af4f-11df-b704-69c5ea8291c5 > > false > freeswitch at example> > > > Q. When i register SIP phone, nonce = uuid? > Authorization: Digest username="2001", realm="78.23.89.55", > nonce="3f7f5dc2-af4f-11df-b704-69c5ea8291c5", uri="sip:78.23.89.55", > algorithm=MD5, qop=auth, cnonce="339774495957948623487dc37177be3f", > nc=00000001, response="8fa80654a0c7bfba71c81f5e9cbafee6" > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/0d281778/attachment.html From shamun.toha at gmail.com Tue Aug 24 02:47:11 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Tue, 24 Aug 2010 11:47:11 +0200 Subject: [Freeswitch-users] FreeSwitch - uuid find list dump In-Reply-To: References: Message-ID: Yes its true. Only way to find is. ; to list whole show channels ; to list specific target query show channels like 2000 at my.freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/ee5847ec/attachment.html From tculjaga at gmail.com Tue Aug 24 03:52:50 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 24 Aug 2010 12:52:50 +0200 Subject: [Freeswitch-users] long DTMF Message-ID: hello, i just got a question. Is it possible somehow to achieve a DTMF collect cancel feature. quite hard to explain :)) lets suppose you have an extension that collects some digits ... e.g. 123456789 .. and in the middle you realized you entered a wrong digit and u want to correct it. Is there any way you can assign a digit e-g. * or # to interupt te current input and to start over again by lets say transfering back to the extension collecting digits? Any hint ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/003bc64c/attachment.html From fdelawarde at wirelessmundi.com Tue Aug 24 04:41:50 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 24 Aug 2010 13:41:50 +0200 Subject: [Freeswitch-users] long DTMF In-Reply-To: References: Message-ID: <1282650110.2970.52.camel@luna.tc.commsmundi.com> On Tue, 2010-08-24 at 12:52 +0200, Tihomir Culjaga wrote: > hello, i just got a question. > > Is it possible somehow to achieve a DTMF collect cancel feature. > quite hard to explain :)) > > > lets suppose you have an extension that collects some digits ... e.g. > 123456789 .. and in the middle you realized you entered a wrong digit > and u want to correct it. > Is there any way you can assign a digit e-g. * or # to interupt te > current input and to start over again by lets say transfering back to > the extension collecting digits? It shouldn't be hard to do with a small lua script with an input callback handling DTMFs. You could even create a small DTMF editor with a digit for "backspace" character for example, that would erase the last digit, or a digit that would say the current sequence.... Check out: http://wiki.freeswitch.org/wiki/Lua#session:setInputCallback Fran?ois. From jim at k4gvo.com Tue Aug 24 05:30:53 2010 From: jim at k4gvo.com (Jim) Date: Tue, 24 Aug 2010 08:30:53 -0400 Subject: [Freeswitch-users] Does anyone have a POTS phone connected to an openZAP supported card working? Message-ID: <4C73BB7D.9070106@k4gvo.com> The phone can connect to other phones in the system but not to an outside line. Thanks, Jim. From fs-list at communicatefreely.net Tue Aug 24 06:10:07 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 24 Aug 2010 09:10:07 -0400 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: Message-ID: <4C73C4AF.6000100@communicatefreely.net> So, what are your thoughts? > I think I would be interested in helping if I can, as I'm more or less in the same boat, with an existing user base on Asterisk. I would like to make things a little bit more configurable (not that they aren't already). I don't have any experience working with C, but I am getting to know LUA, PHP, SQL, and a few others. I'm hoping I can do something helpful. I have moved a lot of logic outside of the application and into a lua script to do what I want. Here's some of the features I have added. Let me know if this is something we could roll into a new module. -Configurable cause code to greeting mappings This lets you define an alternate greeting that will play if your phone is busy, in do not disturb, not registered, etc. It gives similar functionality to the busy/unavail greetings in Asterisk, but with a lot more flexibility. I also defined a mapping for "vm-direct", as on our PBX, you can transfer a call to **EXTEN to go straight to voicemail. On a NO_ANSWER cause, the selected greeting is played, which allows for vacation messages. If the mappings are set in the database, but the alternate greeting hasn't been recorded, it will use the selected main greeting, or play "the person at extension ... is unavailable". -Configurable 0 IVR routing I have put a bit more functionality in the 0 routing. In our Asterisk implementation, we had 0 go to reception, and # go to the users cell phone. A lot of people use this. There isn't a good way to do this in Freeswitch, so we came up with a different option. In the VM config database that we use, I have an IVR enable flag, as well as an extension slot for digits 0-9. If the IVR is disabled, pressing 0 transfers to the extension in the 0 slot. If the IVR is enabled, it uses greeting 9 as the menu, and will let the caller choose an option, transferring them to the extension set in the database. This lets a user do something like this: "You have reached Tim, but I'm away from my desk at the moment. Please leave a message after the tone or press 0 for immediate assistance". ->"For technical support, press 1. To reach my assistant, press 2. If you would like to try my cell phone, press 3". I would also like to see: An option NOT to delete the voice message when you e-mail it from the vm menu. A mechanism to record your name that the dial by name directory module can use. An option to skip the message details before you play them. I'm sure there are others. Let me know if you think any of these options are useful. From ali.stgt at gmail.com Tue Aug 24 06:14:33 2010 From: ali.stgt at gmail.com (=?UTF-8?B?RHVybXXFnyBBbGkgw5Z6dMO8cms=?=) Date: Tue, 24 Aug 2010 16:14:33 +0300 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session Message-ID: Brian, we are not bridging calls. FreeSWITCH (Leg A) as the caller can not be in a 'not answered' state. Also if I see 200 then it must be come from Leg B (the callee) which means that other party has answered the call. Also, both legs was active but FS had reported us the opposite. And this happened more than 40.000 times between 8 hours. I would like to know what you are expecting from the sip trace, because the log is detailed enough to be verified that FS has catched the 200 SIP - but it seems to be ignored (maybe hangup timer started or something else) therefor the session is closed with the wrong state. It is also clear for me, that FS is *over-challenged* to set the correct hangup cause in bulk calls over 200 cps. @Milena We must to re-call all numbers again to give you a SIP trace but this is to expensive for us :-) > > ---------- Weitergeleitete Nachricht ---------- > From: Milena > To: FreeSWITCH Users Help > Date: Mon, 23 Aug 2010 10:17:37 -0500 > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > answered session > type "version" on the CLI > > If the "sip trace wasnt active" then activate it and pastebin it, that's > the only way you can get help with this, you know the developers are experts > not soothsayers :) > > > -Mile > > 2010/8/23 Durmu? Ali ?zt?rk > >> I have setup freeswitch by using git and followed the instructions on the >> wiki page. How can I retrieve the exact git version? >> > > > ---------- Weitergeleitete Nachricht ---------- > From: Brian West > To: FreeSWITCH Users Help > Date: Mon, 23 Aug 2010 10:18:30 -0500 > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > answered session > Again without a SIP trace I can't say... You have to remember that > FreeSWITCH is a B2BUA and a call thru it results in TWO legs. If you were > to ANSWER the first but then bridge to the second then it didn't answer you > might still get a NO_ANSWER even thou you see the 200ok for the one leg of > the call. > > /b > > On Aug 23, 2010, at 10:08 AM, Durmu? Ali ?zt?rk wrote: > > > I have setup freeswitch by using git and followed the instructions on the > wiki page. How can I retrieve the exact git version? > > > > There is no override of the hangup cause in our code. The algorithm is > very simple; we call the originate function with the playback action and use > a wav file as argument. Hangup is done automatically by FreeSWITCH, after > eof of the wav file is reached. Or the other part hangs up before the file > ends. > > > > We dont traced the communication but in other hand, the sip codes are > traced out into the log file (183-->180-->200=call established). This show > to me, that there is no problem with the SIP transactions. Please assume, > that SIP messages are OK. > > > > What else could be happen? > > > > > > ---------- Weitergeleitete Nachricht ---------- > From: Sergey Okhapkin > To: FreeSWITCH Users Help > Date: Mon, 23 Aug 2010 11:22:37 -0400 > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > answered session > You're running concurrent calls, hangup cause NO_ANSWER and log line with > 200 > OK belong to different calls. > > On Monday 23 August 2010, Durmu? Ali ?zt?rk wrote: > > I have setup freeswitch by using git and followed the instructions on the > > wiki page. How can I retrieve the exact git version? > > > > There is no override of the hangup cause in our code. The algorithm is > very > > simple; we call the originate function with the playback action and use a > > wav file as argument. Hangup is done automatically by FreeSWITCH, after > eof > > of the wav file is reached. Or the other part hangs up before the file > > ends. > > > > We dont traced the communication but in other hand, the sip codes are > > traced out into the log file (183-->180-->200=call established). This > show > > to me, that there is no problem with the SIP transactions. Please > assume, > > that SIP messages are OK. > > > > What else could be happen? > > > > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > > string 0 = [sip_from_uri=sip:xxxx at xxxxx] > > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > > string 1 = [ignore_early_media=true] > > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > > string 2 = [sip_cid_type=none] > > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > > string 3 = [originate_timeout=40] > > 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel > > sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8- > > 227e99f3f0ef] > > 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 > > (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT > > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal > > sofia/internal/9055599XXXXX [BREAK] > > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 > > sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 > > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 > > (sofia/internal/9055599XXXXX) Running State Change CS_INIT > > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 > > (sofia/internal/9055599XXXXX) State INIT > > 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 > > sofia/internal/9055599XXXXX SOFIA INIT > > 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 > > (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING > > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal > > sofia/internal/9055599XXXXX [BREAK] > > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 > > (sofia/internal/9055599XXXXX) State INIT going to sleep > > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 > > (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING > > 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 > > (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING > > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > > (sofia/internal/9055599XXXXX) State ROUTING > > 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 > > sofia/internal/9055599XXXXX SOFIA ROUTING > > 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 > > (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal > > sofia/internal/9055599XXXXX [BREAK] > > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > > (sofia/internal/9055599XXXXX) State ROUTING going to sleep > > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 > > (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA > > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 > > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA > > 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 > > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep > > 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel > > sofia/internal/9055599XXXXX entering state [calling][0] > > 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX > > Update Callee ID to "9055599XXXXX" <9055599XXXXX> > > 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel > > sofia/internal/9055599XXXXX entering state [proceeding][183] > > 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready > > sofia/internal/9055599XXXXX! > > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel > > sofia/internal/9055599XXXXX entering state [proceeding][183] > > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: > > 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec > > sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples > > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send > > payload to 101 > > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP > > [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 > > port 18620 codec: 8 ms: 20 > > 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer > [soft] > > 160 bytes per 20ms > > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send > > payload to 101 > > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf > receive > > payload to 101 > > 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer > > sofia/internal/9055599XXXXX! > > 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 > > (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY > > 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel > > sofia/internal/9055599XXXXX skipping state [proceeding][180] > > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel > > sofia/internal/9055599XXXXX entering state [completing][200] > > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP > > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 > > (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP > > 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup > > sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] > > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal > > sofia/internal/9055599XXXXX [KILL] > > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 > > (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP > > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 > > (sofia/internal/9055599XXXXX) State HANGUP > > 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel > > sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER > > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal > > sofia/internal/9055599XXXXX [BREAK] > > 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to > > sofia/internal/9055599XXXXX > > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 > > sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER > > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 > > (sofia/internal/9055599XXXXX) State HANGUP going to sleep > > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 > > (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING > > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal > > sofia/internal/9055599XXXXX [BREAK] > > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 > > (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING > > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > > (sofia/internal/9055599XXXXX) State REPORTING > > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 > > sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER > > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > > (sofia/internal/9055599XXXXX) State REPORTING going to sleep > > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 > > (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY > > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal > > sofia/internal/9055599XXXXX [BREAK] > > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session > 18770 > > (sofia/internal/9055599XXXXX) Locked, Waiting on external entities > > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session > > 18770 (sofia/internal/9055599XXXXX) Ended > > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close > > Channel sofia/internal/9055599XXXXX [CS_DESTROY] > > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 > > (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN > > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 > > (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY > > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > > (sofia/internal/9055599XXXXX) State DESTROY > > 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 > > sofia/internal/9055599XXXXX SOFIA DESTROY > > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 > > sofia/internal/9055599XXXXX Standard DESTROY > > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > > (sofia/internal/9055599XXXXX) State DESTROY going to sleep > > > > > > > > > > ---------- Weitergeleitete Nachricht ---------- > > > > > From: Brian West > > > To: FreeSWITCH Users Help > > > Date: Mon, 23 Aug 2010 09:29:32 -0500 > > > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > > > answered session > > > Without that we can't help. You also need to make sure you're on the > > > very latest code. The one thing I'm sure we get right are the hangup > > > causes unless you're doing something to override them. > > > > > > /b > > > > > > On Aug 23, 2010, at 9:24 AM, Durmu? Ali ?zt?rk wrote: > > > > I'm afraid, sip trace wasnt active. > > > > > > ---------- Weitergeleitete Nachricht ---------- > > > From: Steven Ayre > > > To: FreeSWITCH Users Help > > > Date: Mon, 23 Aug 2010 11:53:33 +0100 > > > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while > > > answered session > > > Do you have a sip trace for those calls? > > > > > > > > > On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: > > > Hello, > > > > > > we had some trouble while executing a bulk call process with > originating > > > a parallel call of 200. Because in many cases, FreeSWITCH has notified > > > hangups (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong > > > hangup-cause. Instead of notifying the successful state, we got the > > > NO_ANSWER hangup cause. BUSY and NO_ANSWER states are candidates for > > > retries, therefor many numbers are called/payed twice. See log below. > > > > > > Some other questions: > > > > > > B) Is the originate_timeout value an overall timer or a timer for the > > > ringing (starts if SIP code 180 incomes?) stage. > > > > > > C) We are originating each number in a separate thread and listen to > the > > > channel events for updating the call result. Should we change this > > > implementation or is this a good scenario/standard way. Related to the > > > call result, if it is busy or not answered, the call is retried after > 30 > > > min. What are the recommends on this side to be ensured, the correct > > > hangup case be got and the number is not called twice.. > > > > > > D) What do I have to bear in mind for bulk calls with parallel calls > over > > > 200. > > > > > > Thanks for your answer in advance. > > > > > > > > > Ali > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/3052243f/attachment-0001.html From jaybinks at gmail.com Tue Aug 24 06:15:16 2010 From: jaybinks at gmail.com (jay binks) Date: Tue, 24 Aug 2010 23:15:16 +1000 Subject: [Freeswitch-users] mod_conference Message-ID: is there a way ( ESL or API ) to list active conferences on an FS box. im playing with dynamic conferences, so they aren't pre-defined.. I COULD keep track using ESL but it would be nice to avoid tracking state if possible. -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/b6c8466a/attachment.html From jcasale at activenetwerx.com Tue Aug 24 06:24:52 2010 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 24 Aug 2010 13:24:52 +0000 Subject: [Freeswitch-users] Does anyone have a POTS phone connected to an openZAP supported card working? In-Reply-To: <4C73BB7D.9070106@k4gvo.com> References: <4C73BB7D.9070106@k4gvo.com> Message-ID: >The phone can connect to other phones in the system but not to an >outside line. That's a dialplan issue. I assume if you pastebin your zap conf files and your dialplan etc it wouldn't be hard to figure out why... From dujinfang at gmail.com Tue Aug 24 06:36:37 2010 From: dujinfang at gmail.com (Seven Du) Date: Tue, 24 Aug 2010 21:36:37 +0800 Subject: [Freeswitch-users] mod_conference In-Reply-To: References: Message-ID: conference list On Tue, Aug 24, 2010 at 9:15 PM, jay binks wrote: > is there a way ( ESL or API ) to list active conferences on an FS box. > im playing with dynamic conferences, so they?aren't?pre-defined.. > I COULD keep track using ESL but it would be nice to avoid tracking state if > possible. > -- > Sincerely > > Jay > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From jaybinks at gmail.com Tue Aug 24 06:49:43 2010 From: jaybinks at gmail.com (jay binks) Date: Tue, 24 Aug 2010 23:49:43 +1000 Subject: [Freeswitch-users] mod_conference In-Reply-To: References: Message-ID: geez that will teach me to post at 11:30pm... :( I Tried that and it didnt work... ( maybe a mistype I guess ) however, it dosnt seem that the tab completion on "conference" has this listed.. ( they all seem to require a conference name ) and im not sure i saw it on the wiki... hmm but maybe I should head off to bed before I make more of an ass of myself :P as a side note... awww.. no "as xml" on this one ( like show channels / calls ) would be nice for parsing J On Tue, Aug 24, 2010 at 11:36 PM, Seven Du wrote: > conference list > > On Tue, Aug 24, 2010 at 9:15 PM, jay binks wrote: > > is there a way ( ESL or API ) to list active conferences on an FS box. > > im playing with dynamic conferences, so they aren't pre-defined.. > > I COULD keep track using ESL but it would be nice to avoid tracking state > if > > possible. > > -- > > Sincerely > > > > Jay > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/7bb6d575/attachment.html From helmut.kuper at ewetel.de Tue Aug 24 06:52:11 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 24 Aug 2010 15:52:11 +0200 Subject: [Freeswitch-users] Predefined hangup cause possible for originate? Message-ID: <4C73CE8B.6060805@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, is it possible, to load originate command with a predefined hangup cause e.g. "completed_elsewhere" when originate-call-timeout occured? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMc86L4tZeNddg3dwRAuWEAKCzxADKkpv6x/gc8Xkmgy8QaGhXUACfSr8R Gh+h7TEqV8KR7Xe+GdrS7wM= =pLvI -----END PGP SIGNATURE----- From devel at thom.fr.eu.org Tue Aug 24 06:54:21 2010 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Tue, 24 Aug 2010 15:54:21 +0200 Subject: [Freeswitch-users] =?utf-8?q?Does_anyone_have_a_POTS_phone_connec?= =?utf-8?q?ted_to_an_openZAP_supported_card_working=3F?= In-Reply-To: <4C73BB7D.9070106@k4gvo.com> References: <4C73BB7D.9070106@k4gvo.com> Message-ID: On Tue, 24 Aug 2010 08:30:53 -0400, Jim wrote: > The phone can connect to other phones in the system but not to an > outside line. > > Thanks, > Jim. > If this is a new setup, you should be using freetdm instead of openzap. Fran?ois > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Tue Aug 24 07:03:01 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 24 Aug 2010 16:03:01 +0200 Subject: [Freeswitch-users] mod_conference In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC4655BC@cooper> conference xml_list :) type "conference" for a complete list of commands. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r jay binks Skickat: den 24 augusti 2010 15:50 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] mod_conference geez that will teach me to post at 11:30pm... :( I Tried that and it didnt work... ( maybe a mistype I guess ) however, it dosnt seem that the tab completion on "conference" has this listed.. ( they all seem to require a conference name ) and im not sure i saw it on the wiki... hmm but maybe I should head off to bed before I make more of an ass of myself :P as a side note... awww.. no "as xml" on this one ( like show channels / calls ) would be nice for parsing J On Tue, Aug 24, 2010 at 11:36 PM, Seven Du > wrote: conference list On Tue, Aug 24, 2010 at 9:15 PM, jay binks > wrote: > is there a way ( ESL or API ) to list active conferences on an FS box. > im playing with dynamic conferences, so they aren't pre-defined.. > I COULD keep track using ESL but it would be nice to avoid tracking state if > possible. > -- > Sincerely > > Jay > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely Jay !DSPAM:4c73cfee32939223221216! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/a8aeae75/attachment-0001.html From fs-list at communicatefreely.net Tue Aug 24 07:26:53 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 24 Aug 2010 10:26:53 -0400 Subject: [Freeswitch-users] Account selection In-Reply-To: <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> Message-ID: <4C73D6AD.8000802@communicatefreely.net> > This can be very important when each SIP account represents a different company/business. Although one person is dealing with all those businesses, when an outgoing call is made it is imperative that the correct SIP account is used to make that call so that the recipient is correctly informed who is making the call. > So really, your issue is with presented identity in terms of caller ID name and number then? Can you set outgoing caller ID name nad number on the 6 provider accounts? If you can, you may have some better options. On our platform, we use the dialplan to route all calls to the most appropriate provider, based on the number that was dialed, and what the rates are for each carrier in a given area (least cost routing), although reliability in certain areas is also factored. Each extension registers with a single registration, and has it's own internal caller ID name and number (the user's name and extension). We use the Aastra phones, and built a little XML app that lets the user pick from a list of possible caller ID name and number combinations. This tool updates the database value that will be used for effective_caller_id name and number. With this setup, one user (one SIP registration) can have an unlimited number "businesses". For incoming calls, we prefix the caller ID name with a short string that identifies the incoming number or "business". Sometimes, a combination of registrations and the selector tool is best. If you don't have XML browsers on the phone, you could just as easily do this with an IVR tool, a web page, or with prefixes. Whatever is easiest. You can have more than one option. If your upstream providers can give you DID numbers, you can have more than one business on the same provider account, which is a lot easier to manage (one gateway entry, but lots of "lines"). This is getting into serious PBX stuff though, and I get the impression you don't really want a PBX. Or do you? -Tim From mthakershi at gmail.com Tue Aug 24 07:39:07 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 24 Aug 2010 09:39:07 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: <4C731F60.6030801@isptelecom.net> References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> <54AB9D1D-BFE7-4685-B098-ECB3806C3666@ipeva.fr> <4C72FD79.5030401@isptelecom.net> <4C731F60.6030801@isptelecom.net> Message-ID: Not a problem. Thanks for help. Will test as you've suggested. On Aug 23, 2010 8:33 PM, "Victor Chukalovskiy" wrote: Malay, I believe full configuration is too big to be posted here, however my sip profile has following relevant lines: My acl.conf.xml has following lines: As you see, this creates a simple ACL that will "deny" everything but address range defined by A.B.C.D/24 My sip profile makes use of this ACL. In your configuration you seem to miss a couple things: 1) You use reserved ACL name acl.auto. Please use something else and non-default. 2) If your ACL is "deny" by default, all nodes should be "allow". And vice verse, if your ACL is "allow" by default then nodes only make sence if they are set to "deny". This depends on what your want to do :-) In your case try: And then point apply-inbound-acl to this new "Malay_ACL". It could be useful to consult following pages: http://wiki.freeswitch.org/wiki/Acl http://en.wikipedia.org/wiki/Classless_Inter-Domain_Routing -Victor On -10/01/37 02:59 PM, Malay Thakershi wrote: > > Thank you. Could you please share your configura... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/022b4fda/attachment.html From jim at k4gvo.com Tue Aug 24 08:29:52 2010 From: jim at k4gvo.com (Jim) Date: Tue, 24 Aug 2010 11:29:52 -0400 Subject: [Freeswitch-users] Does anyone have a POTS phone connected to an openZAP supported card working? In-Reply-To: References: <4C73BB7D.9070106@k4gvo.com> Message-ID: <4C73E570.20301@k4gvo.com> On 08/24/2010 09:24 AM, Joseph L. Casale wrote: >> The phone can connect to other phones in the system but not to an >> outside line. >> > That's a dialplan issue. I assume if you pastebin your zap conf files > and your dialplan etc it wouldn't be hard to figure out why... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > It may go a little beyond that. It appears that when defining the extension number which is done in the openzap.conf file, there is no way to set a variable, like . Without that FS fails the test for ${toll_allow}(domestic,international,local) and doesn't connect. An attempt to use a special context (not default) to set that variable and transfer to the default context fails when the Didn't get set. Because I don't quite know what I am doing I decided to try to set the default_gateway via a And it worked! I have no idea if it's the right way and it doesn't give me any flexibility to use other gateways in case that one is busy, but it worked. Thanks, Jim. From anthony.minessale at gmail.com Tue Aug 24 08:31:54 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Aug 2010 10:31:54 -0500 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: Message-ID: Maybe you should start with an extensive list of what you can't do with mod_voicemail that you are missing. The thought of making a separate module to pretend to be Asterisk makes me sad. I consider that a step backwards. If you want to make mod_voicemail better so you can personally do anything you choose with it (such as make it emulate other systems) that's a *bit* better. On Mon, Aug 23, 2010 at 11:52 PM, Kristian Kielhofner wrote: > Hello everyone, > > ?While extensive tweaks to Mod_voicemail and phrases can get "pretty > close" to emulating existing voicemail systems like Asterisk, "pretty > close" doesn't cut it when you have many discerning users that have > become accustomed to the behavior of a given system. ?I want to move > to FreeSWITCH voicemail but I don't want to have to reprint quick > reference cards and retrain users and support staff. > > ?I want to make FreeSWITCH mod_voicemail even better by expanding its > already impressive customization. ?A few tweaks here, and few new > features there and we should be able to do drop in replacements for > Asterisk and presumably other voicemail systems. > > ?Official bounty here: > > http://jira.freeswitch.org/browse/BOUNTY-22 > > ?You'll notice I haven't included a dollar/euro/pound amount... ?Well > it's because I'm not even sure of the scope of this work. ?I'm hoping > to get some discussion going first (of course any work we pay for will > be open source and given back). ?So, what are your thoughts? > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Tue Aug 24 08:35:48 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Aug 2010 10:35:48 -0500 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: You did not provide a sip trace you provided the freeswitch console log without a sip trace. you need to also enable "sofia profile internal siptrace on" also listen to sergey. On Tue, Aug 24, 2010 at 8:14 AM, Durmu? Ali ?zt?rk wrote: > Brian, we are not bridging calls. FreeSWITCH (Leg A) as the caller can not > be in a 'not answered' state. Also if I see 200 then it must be come from > Leg B (the callee) which means that other party has answered the call. Also, > both legs was active but FS had reported us the opposite. And this happened > more than 40.000 times between 8 hours. > > I would like to know what you are expecting from the sip trace, because the > log is detailed enough to be verified that FS has catched the 200 SIP - but > it seems to be ignored (maybe hangup timer started or something else) > therefor the session is closed with the wrong state. It is also clear for > me, that FS is over-challenged to set the correct hangup cause in bulk calls > over 200 cps. > > @Milena > We must to re-call all numbers again to give you a SIP trace but this is to > expensive for us :-) > >> >> ---------- Weitergeleitete Nachricht ---------- >> From:?Milena >> To:?FreeSWITCH Users Help >> Date:?Mon, 23 Aug 2010 10:17:37 -0500 >> Subject:?Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> answered session >> type "version" on the CLI >> If the "sip trace wasnt active" then activate it and pastebin it, that's >> the only way you can get help with this, you know the developers are experts >> not?soothsayers :) >> >> >> -Mile >> 2010/8/23 Durmu? Ali ?zt?rk >>> >>> I have setup freeswitch by using git and followed the instructions on the >>> wiki page. How can I retrieve the exact git version? >> >> >> ---------- Weitergeleitete Nachricht ---------- >> From:?Brian West >> To:?FreeSWITCH Users Help >> Date:?Mon, 23 Aug 2010 10:18:30 -0500 >> Subject:?Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> answered session >> Again without a SIP trace I can't say... You have to remember that >> FreeSWITCH is a B2BUA and a call thru it results in TWO legs. ?If you were >> to ANSWER the first but then bridge to the second then it didn't answer you >> might still get a NO_ANSWER even thou you see the 200ok for the one leg of >> the call. >> >> /b >> >> On Aug 23, 2010, at 10:08 AM, Durmu? Ali ?zt?rk wrote: >> >> > I have setup freeswitch by using git and followed the instructions on >> > the wiki page. How can I retrieve the exact git version? >> > >> > There is no override of the hangup cause in our code. The algorithm is >> > very simple; we call the originate function with the playback action and use >> > a wav file as argument. Hangup is done automatically by FreeSWITCH, after >> > eof of the wav file is reached. Or the other part hangs up before the file >> > ends. >> > >> > We dont traced the communication but in other hand, the sip codes are >> > traced out into the log file (183-->180-->200=call established). This show >> > to me, that there is no problem with the SIP transactions. Please assume, >> > that SIP messages are OK. >> > >> > What else could be happen? >> >> >> >> >> >> ---------- Weitergeleitete Nachricht ---------- >> From:?Sergey Okhapkin >> To:?FreeSWITCH Users Help >> Date:?Mon, 23 Aug 2010 11:22:37 -0400 >> Subject:?Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> answered session >> You're running concurrent calls, hangup cause NO_ANSWER and log line with >> 200 >> OK belong to different calls. >> >> On Monday 23 August 2010, Durmu? Ali ?zt?rk wrote: >> > I have setup freeswitch by using git and followed the instructions on >> > the >> > wiki page. How can I retrieve the exact git version? >> > >> > There is no override of the hangup cause in our code. The algorithm is >> > very >> > simple; we call the originate function with the playback action and use >> > a >> > wav file as argument. Hangup is done automatically by FreeSWITCH, after >> > eof >> > of the wav file is reached. Or the other part hangs up before the file >> > ?ends. >> > >> > We dont traced the communication but in other hand, the sip codes are >> > ?traced out into the log file (183-->180-->200=call established). This >> > show >> > ?to me, that there is no problem with the SIP transactions. Please >> > assume, >> > ?that SIP messages are OK. >> > >> > What else could be happen? >> > >> > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> > string 0 = [sip_from_uri=sip:xxxx at xxxxx] >> > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> > string 1 = [ignore_early_media=true] >> > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> > string 2 = [sip_cid_type=none] >> > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> > string 3 = [originate_timeout=40] >> > 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel >> > sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8- >> > 227e99f3f0ef] >> > 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 >> > (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT >> > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send >> > signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 >> > sofia/internal/9055599XXXXX set >> > UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 >> > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_INIT >> > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 >> > (sofia/internal/9055599XXXXX) State INIT >> > 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 >> > sofia/internal/9055599XXXXX SOFIA INIT >> > 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 >> > (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING >> > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send >> > signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 >> > (sofia/internal/9055599XXXXX) State INIT going to sleep >> > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 >> > (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 >> > (sofia/internal/9055599XXXXX) State ROUTING >> > 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 >> > sofia/internal/9055599XXXXX SOFIA ROUTING >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 >> > (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> >> > CS_CONSUME_MEDIA >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send >> > signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 >> > (sofia/internal/9055599XXXXX) State ROUTING going to sleep >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 >> > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA >> > 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 >> > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep >> > 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel >> > sofia/internal/9055599XXXXX entering state [calling][0] >> > 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 >> > sofia/internal/9055599XXXXX >> > Update Callee ID to "9055599XXXXX" <9055599XXXXX> >> > 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel >> > sofia/internal/9055599XXXXX entering state [proceeding][183] >> > 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready >> > sofia/internal/9055599XXXXX! >> > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel >> > sofia/internal/9055599XXXXX entering state [proceeding][183] >> > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: >> > 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec >> > sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples >> > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send >> > payload to 101 >> > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP >> > [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 >> > ?port 18620 codec: 8 ms: 20 >> > 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer >> > [soft] >> > 160 bytes per 20ms >> > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send >> > payload to 101 >> > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf >> > receive >> > payload to 101 >> > 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer >> > sofia/internal/9055599XXXXX! >> > 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 >> > (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY >> > ?2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel >> > sofia/internal/9055599XXXXX skipping state [proceeding][180] >> > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel >> > sofia/internal/9055599XXXXX entering state [completing][200] >> > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 >> > (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP >> > 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup >> > sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal >> > sofia/internal/9055599XXXXX [KILL] >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 >> > (sofia/internal/9055599XXXXX) State HANGUP >> > 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel >> > sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send >> > signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to >> > sofia/internal/9055599XXXXX >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 >> > sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 >> > (sofia/internal/9055599XXXXX) State HANGUP going to sleep >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 >> > (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send >> > signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING >> > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 >> > (sofia/internal/9055599XXXXX) State REPORTING >> > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 >> > sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER >> > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 >> > (sofia/internal/9055599XXXXX) State REPORTING going to sleep >> > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 >> > (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY >> > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send >> > signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session >> > 18770 >> > (sofia/internal/9055599XXXXX) Locked, Waiting on external entities >> > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session >> > ?18770 (sofia/internal/9055599XXXXX) Ended >> > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close >> > ?Channel sofia/internal/9055599XXXXX [CS_DESTROY] >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 >> > (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 >> > (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 >> > (sofia/internal/9055599XXXXX) State DESTROY >> > 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 >> > sofia/internal/9055599XXXXX SOFIA DESTROY >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 >> > sofia/internal/9055599XXXXX Standard DESTROY >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 >> > (sofia/internal/9055599XXXXX) State DESTROY going to sleep >> > >> > >> > >> > >> > ---------- Weitergeleitete Nachricht ---------- >> > >> > > From: Brian West >> > > To: FreeSWITCH Users Help >> > > Date: Mon, 23 Aug 2010 09:29:32 -0500 >> > > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> > > answered session >> > > Without that we can't help. ?You also need to make sure you're on the >> > > very latest code. ?The one thing I'm sure we get right are the hangup >> > > causes unless you're doing something to override them. >> > > >> > > /b >> > > >> > > On Aug 23, 2010, at 9:24 AM, Durmu? Ali ?zt?rk wrote: >> > > > I'm afraid, sip trace wasnt active. >> > > >> > > ---------- Weitergeleitete Nachricht ---------- >> > > From: Steven Ayre >> > > To: FreeSWITCH Users Help >> > > Date: Mon, 23 Aug 2010 11:53:33 +0100 >> > > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> > > answered session >> > > Do you have a sip trace for those calls? >> > > >> > > >> > > On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: >> > > Hello, >> > > >> > > we had some trouble while executing a bulk call process with >> > > originating >> > > a parallel call of 200. Because in many cases, FreeSWITCH has notified >> > > hangups (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong >> > > hangup-cause. Instead of notifying the successful state, we got the >> > > NO_ANSWER hangup cause. BUSY and NO_ANSWER states are candidates for >> > > retries, therefor many numbers are called/payed twice. See log below. >> > > >> > > Some other questions: >> > > >> > > B) Is the originate_timeout value an overall timer or a timer for the >> > > ringing (starts if SIP code 180 incomes?) stage. >> > > >> > > C) We are originating each number in a separate thread and listen to >> > > the >> > > channel events for updating the call result. Should we change this >> > > implementation or is this a good scenario/standard way. Related to the >> > > call result, if it is busy or not answered, the call is retried after >> > > 30 >> > > min. What are the recommends on this side to be ensured, the correct >> > > hangup case be got and the number is not called twice.. >> > > >> > > D) What do I have to bear in mind for bulk calls with parallel calls >> > > over >> > > 200. >> > > >> > > Thanks for your answer in advance. >> > > >> > > >> > > Ali >> > >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From asilva at wirelessmundi.com Tue Aug 24 08:41:42 2010 From: asilva at wirelessmundi.com (Antonio) Date: Tue, 24 Aug 2010 17:41:42 +0200 Subject: [Freeswitch-users] ivr menu matching and execution problem Message-ID: <1282664502.13028.34.camel@marces.tc.commsmundi.com> During the configuration of an ivr menu and test i just realized that if you have multiple actions with the same "digits" it executes the both actions. The execution of the multiple actions is normal? The normal behavior should executed only the action that is match first, reading the priority of the entries in the menu. I look at the code, and in the function, switch_ivr_menu_execute, looks like the loop that searches for the match/action should need a break after a success match, to avoid the continues search for other matches. Thanks, Ant?nio /* configuration */ /* some log from fs */ 2010-08-24 17:07:47.888057 [DEBUG] switch_ivr_menu.c:777 building menu 'ivr' 2010-08-24 17:07:47.888057 [DEBUG] switch_ivr_menu.c:838 binding menu action 'menu-exec-app' to '/^(10[0-9])$/' 2010-08-24 17:07:47.888057 [DEBUG] switch_ivr_menu.c:838 binding menu action 'menu-exec-app' to '9' 2010-08-24 17:07:47.888057 [DEBUG] switch_ivr_menu.c:838 binding menu action 'menu-exec-app' to '9' 2010-08-24 17:07:47.888057 [DEBUG] switch_ivr_menu.c:414 Executing IVR menu ivr 2010-08-24 17:07:52.616160 [DEBUG] switch_rtp.c:2892 RTP RECV DTMF 9:480 2010-08-24 17:07:52.617160 [DEBUG] switch_ivr_play_say.c:1468 done playing file 2010-08-24 17:07:52.617160 [DEBUG] switch_ivr_menu.c:329 waiting for 3/4 digits t/o 2000 2010-08-24 17:07:54.635190 [DEBUG] switch_ivr_menu.c:376 digits '9' 2010-08-24 17:07:54.635190 [DEBUG] switch_ivr_menu.c:488 IVR action on menu 'ivr' matched '9' param 'transfer 8 XML ivr-[1]' 2010-08-24 17:07:54.635190 [DEBUG] switch_ivr_menu.c:492 switch_ivr_menu_execute todo=[2] EXECUTE sofia/192.168.10.25/101 at 192.168.10.25 transfer(8 XML ivr-[1]) 2010-08-24 17:07:54.635190 [DEBUG] switch_ivr.c:1470 (sofia/192.168.10.25/101 at 192.168.10.25) State Change CS_EXECUTE -> CS_ROUTING 2010-08-24 17:07:54.635190 [DEBUG] switch_core_session.c:1039 Send signal sofia/192.168.10.25/101 at 192.168.10.25 [BREAK] 2010-08-24 17:07:54.635190 [DEBUG] switch_core_session.c:658 Send signal sofia/192.168.10.25/101 at 192.168.10.25 [BREAK] 2010-08-24 17:07:54.635190 [NOTICE] switch_ivr.c:1476 Transfer sofia/192.168.10.25/101 at 192.168.10.25 to XML[8 at ivr-[1]] 2010-08-24 17:07:54.635190 [DEBUG] switch_ivr_menu.c:488 IVR action on menu 'ivr' matched '9' param 'transfer 9 XML ivr-[1]' 2010-08-24 17:07:54.635190 [DEBUG] switch_ivr_menu.c:492 switch_ivr_menu_execute todo=[2] EXECUTE sofia/192.168.10.25/101 at 192.168.10.25 transfer(9 XML ivr-[1]) 2010-08-24 17:07:54.635190 [DEBUG] switch_core_session.c:658 Send signal sofia/192.168.10.25/101 at 192.168.10.25 [BREAK] 2010-08-24 17:07:54.635190 [NOTICE] switch_ivr.c:1476 Transfer sofia/192.168.10.25/101 at 192.168.10.25 to XML[9 at ivr-[1]] 2010-08-24 17:07:54.635190 [DEBUG] switch_ivr_menu.c:470 action regex [9] [/^(10[0-9])$/] [0] 2010-08-24 17:07:54.635190 [DEBUG] switch_ivr_menu.c:580 exit-sound '/home/system/telephony/sounds/prompt/en/conf-invalid.gsm' 2010-08-24 17:07:54.635190 [DEBUG] switch_core_state_machine.c:348 (sofia/192.168.10.25/101 at 192.168.10.25) State EXECUTE going to sleep 2010-08-24 17:07:54.635190 [DEBUG] switch_core_state_machine.c:314 (sofia/192.168.10.25/101 at 192.168.10.25) Running State Change CS_ROUTING 2010-08-24 17:07:54.635190 [DEBUG] switch_channel.c:1512 (sofia/192.168.10.25/101 at 192.168.10.25) Callstate Change ACTIVE -> RINGING 2010-08-24 17:07:54.635190 [DEBUG] switch_core_state_machine.c:341 (sofia/192.168.10.25/101 at 192.168.10.25) State ROUTING 2010-08-24 17:07:54.635190 [DEBUG] mod_sofia.c:142 sofia/192.168.10.25/101 at 192.168.10.25 SOFIA ROUTING 2010-08-24 17:07:54.635190 [DEBUG] switch_core_state_machine.c:77 sofia/192.168.10.25/101 at 192.168.10.25 Standard ROUTING 2010-08-24 17:07:54.635190 [INFO] mod_dialplan_xml.c:331 Processing 101->9 in context ivr-[1] From fdelawarde at wirelessmundi.com Tue Aug 24 08:42:26 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 24 Aug 2010 17:42:26 +0200 Subject: [Freeswitch-users] fax ident and header variables Message-ID: <1282664546.2970.63.camel@luna.tc.commsmundi.com> Anyone knows the meaning of the fax_ident and fax_header variables? Are those mandatory? What should go in there and how should it be formatted? The only thing I could find is that apparently in the USA, the fax header must contain the telephone number of the line where the fax is connected. Is this only in the USA? Thank you, Fran?ois. From brian at freeswitch.org Tue Aug 24 08:46:02 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Aug 2010 10:46:02 -0500 Subject: [Freeswitch-users] fax ident and header variables In-Reply-To: <1282664546.2970.63.camel@luna.tc.commsmundi.com> References: <1282664546.2970.63.camel@luna.tc.commsmundi.com> Message-ID: <38B60B0D-5D6B-4600-B16A-490D1E29F1FE@freeswitch.org> I think it SHOULD in the US... but really are the fax police going to come arrest you? NO. /b On Aug 24, 2010, at 10:42 AM, Fran?ois Delawarde wrote: > Anyone knows the meaning of the fax_ident and fax_header variables? Are > those mandatory? What should go in there and how should it be formatted? > > The only thing I could find is that apparently in the USA, the fax > header must contain the telephone number of the line where the fax is > connected. Is this only in the USA? > > Thank you, > Fran?ois. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kris at kriskinc.com Tue Aug 24 08:56:47 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 24 Aug 2010 11:56:47 -0400 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: Message-ID: Anthony, I intend to make a list but I wanted to start by: 1) Gauging interest in making this an open source project. 2) Seeing what other people have done (I'm certainly not the first to think of this). Don't focus too much on Asterisk, I'm not. The real goal of this effort is to provide a powerful, easy to use, COMPLETELY customizable voicemail interface for FreeSWITCH. FreeSWITCH is one of the most powerful pieces of software I have ever used and mod_voicemail is an excellent start. However, every user, company, and organization in the world has a different idea of what "voicemail" should be like. Asterisk, Mitel, Avaya, legacy Nortel, NEC, Shoretel, cell carriers like T-Mobile, AT&T, Verizon, etc. All slightly different prompt structures and feature sets. In our case we're coming from Asterisk so at first it will need to emulate Asterisk. Eventually we're going to run it in two modes "legacy voicemail" and "enhanced voicemail" (with our own menu design and feature set). Enhanced voicemail should also have advanced unified messaging capabilities (I feel so slimy for using that buzzword) and other features integrated with our architecture that just aren't practical for us to implement in C. Having to modify mod_voicemail and recompile it to meet some of these needs is less than ideal. If all goes to plan FreeSWITCH will have the best (in my opinion) "voicemail" application in the industry, making it easier for admins/users of other systems to migrate to FreeSWITCH and further FreeSWITCH in its quest to take over the world. What's wrong with that? ;) On Tue, Aug 24, 2010 at 11:31 AM, Anthony Minessale wrote: > Maybe you should start with an extensive list of what you can't do > with mod_voicemail that you are missing. > > The thought of making a separate module to pretend to be Asterisk > makes me sad. ?I consider that a step backwards. > If you want to make mod_voicemail better so you can personally do > anything you choose with it (such as make it emulate other systems) > that's a *bit* better. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kris at kriskinc.com Tue Aug 24 08:59:55 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 24 Aug 2010 11:59:55 -0400 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: <4C73C4AF.6000100@communicatefreely.net> References: <4C73C4AF.6000100@communicatefreely.net> Message-ID: Tim, Thanks for replying. I'm in contact with someone who has a fairly complete voicemail implementation written in LUA. Hopefully we can work out an arrangement to open source that code so we can all work on it and adapt it for our needs. If everything goes as planned I'd love to have you help out. On Tue, Aug 24, 2010 at 9:10 AM, Tim St. Pierre wrote: > ?So, what are your thoughts? >> > I think I would be interested in helping if I can, as I'm more or less in the same boat, with an > existing user base on Asterisk. ?I would like to make things a little bit more configurable (not > that they aren't already). > > I don't have any experience working with C, but I am getting to know LUA, PHP, SQL, and a few > others. I'm hoping I can do something helpful. > > I have moved a lot of logic outside of the application and into a lua script to do what I want. > Here's some of the features I have added. ?Let me know if this is something we could roll into a new > module. > > -Configurable cause code to greeting mappings > > This lets you define an alternate greeting that will play if your phone is busy, in do not disturb, > not registered, etc. ?It gives similar functionality to the busy/unavail greetings in Asterisk, but > with a lot more flexibility. ?I also defined a mapping for "vm-direct", as on our PBX, you can > transfer a call to **EXTEN to go straight to voicemail. ?On a NO_ANSWER cause, the selected greeting > is played, which allows for vacation messages. ?If the mappings are set in the database, but the > alternate greeting hasn't been recorded, it will use the selected main greeting, or play "the person > at extension ... is unavailable". > > -Configurable 0 IVR routing > > I have put a bit more functionality in the 0 routing. ?In our Asterisk implementation, we had 0 go > to reception, and # go to the users cell phone. ?A lot of people use this. ?There isn't a good way > to do this in Freeswitch, so we came up with a different option. ?In the VM config database that we > use, I have an IVR enable flag, as well as an extension slot for digits 0-9. ?If the IVR is > disabled, pressing 0 transfers to the extension in the 0 slot. ?If the IVR is enabled, it uses > greeting 9 as the menu, and will let the caller choose an option, transferring them to the extension > set in the database. ?This lets a user do something like this: > > "You have reached Tim, but I'm away from my desk at the moment. ?Please leave a message after the > tone or press 0 for immediate assistance". > ->"For technical support, press 1. ?To reach my assistant, press 2. ?If you would like to try my > cell phone, press 3". > > > I would also like to see: > An option NOT to delete the voice message when you e-mail it from the vm menu. > A mechanism to record your name that the dial by name directory module can use. > An option to skip the message details before you play them. > > I'm sure there are others. > > Let me know if you think any of these options are useful. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From fdelawarde at wirelessmundi.com Tue Aug 24 09:02:35 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 24 Aug 2010 18:02:35 +0200 Subject: [Freeswitch-users] fax ident and header variables In-Reply-To: <38B60B0D-5D6B-4600-B16A-490D1E29F1FE@freeswitch.org> References: <1282664546.2970.63.camel@luna.tc.commsmundi.com> <38B60B0D-5D6B-4600-B16A-490D1E29F1FE@freeswitch.org> Message-ID: <1282665755.2970.71.camel@luna.tc.commsmundi.com> Are you sure?... Any knowledge about the other one (fax ident)? Fran?ois. On Tue, 2010-08-24 at 10:46 -0500, Brian West wrote: > I think it SHOULD in the US... but really are the fax police going to come arrest you? NO. > > /b > > On Aug 24, 2010, at 10:42 AM, Fran?ois Delawarde wrote: > > > Anyone knows the meaning of the fax_ident and fax_header variables? Are > > those mandatory? What should go in there and how should it be formatted? > > > > The only thing I could find is that apparently in the USA, the fax > > header must contain the telephone number of the line where the fax is > > connected. Is this only in the USA? > > > > Thank you, > > Fran?ois. > > From mnhassan at usa.net Tue Aug 24 09:16:38 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Tue, 24 Aug 2010 22:16:38 +0600 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: <4C73C4AF.6000100@communicatefreely.net> Message-ID: Voicemail is also of high interest to us. Can you be more specific on what you find as "not there" in the current implementation of mod_voicemail? Regards HASSAN On Tue, Aug 24, 2010 at 21:59, Kristian Kielhofner wrote: > Tim, > > Thanks for replying. > > I'm in contact with someone who has a fairly complete voicemail > implementation written in LUA. Hopefully we can work out an > arrangement to open source that code so we can all work on it and > adapt it for our needs. > > If everything goes as planned I'd love to have you help out. > > On Tue, Aug 24, 2010 at 9:10 AM, Tim St. Pierre > wrote: > > So, what are your thoughts? > >> > > I think I would be interested in helping if I can, as I'm more or less in > the same boat, with an > > existing user base on Asterisk. I would like to make things a little bit > more configurable (not > > that they aren't already). > > > > I don't have any experience working with C, but I am getting to know LUA, > PHP, SQL, and a few > > others. I'm hoping I can do something helpful. > > > > I have moved a lot of logic outside of the application and into a lua > script to do what I want. > > Here's some of the features I have added. Let me know if this is > something we could roll into a new > > module. > > > > -Configurable cause code to greeting mappings > > > > This lets you define an alternate greeting that will play if your phone > is busy, in do not disturb, > > not registered, etc. It gives similar functionality to the busy/unavail > greetings in Asterisk, but > > with a lot more flexibility. I also defined a mapping for "vm-direct", > as on our PBX, you can > > transfer a call to **EXTEN to go straight to voicemail. On a NO_ANSWER > cause, the selected greeting > > is played, which allows for vacation messages. If the mappings are set > in the database, but the > > alternate greeting hasn't been recorded, it will use the selected main > greeting, or play "the person > > at extension ... is unavailable". > > > > -Configurable 0 IVR routing > > > > I have put a bit more functionality in the 0 routing. In our Asterisk > implementation, we had 0 go > > to reception, and # go to the users cell phone. A lot of people use > this. There isn't a good way > > to do this in Freeswitch, so we came up with a different option. In the > VM config database that we > > use, I have an IVR enable flag, as well as an extension slot for digits > 0-9. If the IVR is > > disabled, pressing 0 transfers to the extension in the 0 slot. If the > IVR is enabled, it uses > > greeting 9 as the menu, and will let the caller choose an option, > transferring them to the extension > > set in the database. This lets a user do something like this: > > > > "You have reached Tim, but I'm away from my desk at the moment. Please > leave a message after the > > tone or press 0 for immediate assistance". > > ->"For technical support, press 1. To reach my assistant, press 2. If > you would like to try my > > cell phone, press 3". > > > > > > I would also like to see: > > An option NOT to delete the voice message when you e-mail it from the vm > menu. > > A mechanism to record your name that the dial by name directory module > can use. > > An option to skip the message details before you play them. > > > > I'm sure there are others. > > > > Let me know if you think any of these options are useful. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/93c58c26/attachment-0001.html From jim at k4gvo.com Tue Aug 24 09:23:53 2010 From: jim at k4gvo.com (Jim) Date: Tue, 24 Aug 2010 12:23:53 -0400 Subject: [Freeswitch-users] Does anyone have a POTS phone connected to an openZAP supported card working? In-Reply-To: References: <4C73BB7D.9070106@k4gvo.com> Message-ID: <4C73F219.8040801@k4gvo.com> On 08/24/2010 09:54 AM, Fran?ois Legal wrote: > > On Tue, 24 Aug 2010 08:30:53 -0400, Jim wrote: > >> The phone can connect to other phones in the system but not to an >> outside line. >> >> Thanks, >> Jim. >> >> > If this is a new setup, you should be using freetdm instead of openzap. > > Fran?ois > > I'm using the instructions from the Sangoma web site which kind of indicated I should be using oz. I've never heard of freetdm. I'll check it out. Jim. From steveayre at gmail.com Tue Aug 24 09:32:14 2010 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 24 Aug 2010 17:32:14 +0100 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: Message-ID: <39C8C9F7-1CB1-4A73-92A6-5F7E029A6C5F@gmail.com> > 1) Gauging interest in making this an open source project. mod_voicemail is already open source. If you want a feature not in it, specify what it is so that it can be added, or someone can chime in with a tip on how you can already do it. Or you could even supply a patch which adds the feature. So the list of what's missing is really what's useful. Steve on iPhone On 24 Aug 2010, at 16:56, Kristian Kielhofner wrote: > Anthony, > > I intend to make a list but I wanted to start by: > 1) Gauging interest in making this an open source project. > 2) Seeing what other people have done (I'm certainly not the first to > think of this). > > Don't focus too much on Asterisk, I'm not. The real goal of this > effort is to provide a powerful, easy to use, COMPLETELY customizable > voicemail interface for FreeSWITCH. FreeSWITCH is one of the most > powerful pieces of software I have ever used and mod_voicemail is an > excellent start. However, every user, company, and organization in > the world has a different idea of what "voicemail" should be like. > Asterisk, Mitel, Avaya, legacy Nortel, NEC, Shoretel, cell carriers > like T-Mobile, AT&T, Verizon, etc. All slightly different prompt > structures and feature sets. > > In our case we're coming from Asterisk so at first it will need to > emulate Asterisk. Eventually we're going to run it in two modes > "legacy voicemail" and "enhanced voicemail" (with our own menu design > and feature set). Enhanced voicemail should also have advanced > unified messaging capabilities (I feel so slimy for using that > buzzword) and other features integrated with our architecture that > just aren't practical for us to implement in C. > > Having to modify mod_voicemail and recompile it to meet some of > these needs is less than ideal. If all goes to plan FreeSWITCH will > have the best (in my opinion) "voicemail" application in the industry, > making it easier for admins/users of other systems to migrate to > FreeSWITCH and further FreeSWITCH in its quest to take over the world. > What's wrong with that? ;) > > On Tue, Aug 24, 2010 at 11:31 AM, Anthony Minessale > wrote: >> Maybe you should start with an extensive list of what you can't do >> with mod_voicemail that you are missing. >> >> The thought of making a separate module to pretend to be Asterisk >> makes me sad. I consider that a step backwards. >> If you want to make mod_voicemail better so you can personally do >> anything you choose with it (such as make it emulate other systems) >> that's a *bit* better. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveu at coppice.org Tue Aug 24 09:39:46 2010 From: steveu at coppice.org (Steve Underwood) Date: Wed, 25 Aug 2010 00:39:46 +0800 Subject: [Freeswitch-users] fax ident and header variables In-Reply-To: <1282664546.2970.63.camel@luna.tc.commsmundi.com> References: <1282664546.2970.63.camel@luna.tc.commsmundi.com> Message-ID: <4C73F5D2.3040300@coppice.org> On 08/24/2010 11:42 PM, Fran?ois Delawarde wrote: > Anyone knows the meaning of the fax_ident and fax_header variables? Are > those mandatory? What should go in there and how should it be formatted? > > The only thing I could find is that apparently in the USA, the fax > header must contain the telephone number of the line where the fax is > connected. Is this only in the USA? fax_ident should be set to the telephone number to be used within the fax exchange. This will typically appear on an LCD display at the far end. In theory it should be limited to digits, spaces, + and one or two other characters appropriate to telephone numbers. In practice FAX machines are usually happy with any text, up to 20 characters. This string may also play a part in page headers. If fax_header is set to a non-null string, a header line will be inserted at the start of each page, just like a typical FAX machine does. The fax_ident, fax_header and the page number will be used to form the text of this line. If you are forwarding FAXes, you probably don't want to add a header line, as there will already be one that was inserted by the original source. If you are sending a locally generated FAX, you probably do want to add header lines to each page. Steve From fdelawarde at wirelessmundi.com Tue Aug 24 10:03:34 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 24 Aug 2010 19:03:34 +0200 Subject: [Freeswitch-users] fax ident and header variables In-Reply-To: <4C73F5D2.3040300@coppice.org> References: <1282664546.2970.63.camel@luna.tc.commsmundi.com> <4C73F5D2.3040300@coppice.org> Message-ID: <1282669414.2970.83.camel@luna.tc.commsmundi.com> On Wed, 2010-08-25 at 00:39 +0800, Steve Underwood wrote: > fax_ident should be set to the telephone number to be used within the > fax exchange. This will typically appear on an LCD display at the far > end. In theory it should be limited to digits, spaces, + and one or two > other characters appropriate to telephone numbers. In practice FAX > machines are usually happy with any text, up to 20 characters. This > string may also play a part in page headers. > > If fax_header is set to a non-null string, a header line will be > inserted at the start of each page, just like a typical FAX machine > does. The fax_ident, fax_header and the page number will be used to form > the text of this line. If you are forwarding FAXes, you probably don't > want to add a header line, as there will already be one that was > inserted by the original source. If you are sending a locally generated > FAX, you probably do want to add header lines to each page. Thanks Steve, I was looking at that in spandsp, and it's just as you say (for some reason i guess why...). I also looked at US law, specifically the Telephone Consumer Protection Act of 1991 (47 USC 227), or TCPA. As Brian said, the fax police is not going to come, arrest you and rape your wife, but the receiving party has the right to complain to the FCC which will fine you with $500 if this header doesn't exist on each and every page (I love US law)!! "The Commission shall revise the regulations setting technical and procedural standards for telephone facsimile machines to require that any such machine which is manufactured after one year after December 20, 1991, clearly marks, in a margin at the top or bottom of each transmitted page or on the first page of each transmission, the date and time sent, an identification of the business, other entity, or individual sending the message, and the telephone number of the sending machine or of such business, other entity, or individual." So at least in the US, one should place both fax_ident (telephone number of the sending machine or of such business, ...) and fax_header (identification of the business, ...). SpanDSP will automatically generate a header with date and time, fax_header and fax_ident. Fran?ois. From msc at freeswitch.org Tue Aug 24 10:20:29 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Aug 2010 10:20:29 -0700 Subject: [Freeswitch-users] Still can't dial gateway from ZAP phone. In-Reply-To: <4C730A7F.6030408@k4gvo.com> References: <4C730A7F.6030408@k4gvo.com> Message-ID: > EXECUTE sofia/internal/1002 at 192.168.2.51 bridge(sofia/gateway/ > gw4.telasip.com/17705550068) > > Somehow the information in the directory/default/default.xml file never got > included and I'm not sure how to fix it. > When dialing from an FXS port you are not an "authenticated user" so those variables from user directory don't get populated. You have two choices as I see it: #1 - Manually set the ${default_gateway} variable by inserting this line before the bridge: #2 - Use the set_user app ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user) to make the call act like it was from an auth'd user: Give it a whirl and let me know how it goes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/7b63c18f/attachment.html From brian at freeswitch.org Tue Aug 24 10:21:16 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Aug 2010 12:21:16 -0500 Subject: [Freeswitch-users] fax ident and header variables In-Reply-To: <1282669414.2970.83.camel@luna.tc.commsmundi.com> References: <1282664546.2970.63.camel@luna.tc.commsmundi.com> <4C73F5D2.3040300@coppice.org> <1282669414.2970.83.camel@luna.tc.commsmundi.com> Message-ID: First they have to find you... Have you ever tried to fill out a complaint on the FCC's website? OMG its pages and pages and pages and pages of utter bullshit to even get to the point where you can actually complain. :P /b On Aug 24, 2010, at 12:03 PM, Fran?ois Delawarde wrote: > As Brian said, the fax police is not going to come, arrest you and rape > your wife, but the receiving party has the right to complain to the FCC > which will fine you with $500 if this header doesn't exist on each and > every page (I love US law)!! From msc at freeswitch.org Tue Aug 24 10:21:42 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Aug 2010 10:21:42 -0700 Subject: [Freeswitch-users] mod_conference In-Reply-To: References: Message-ID: > > > and im not sure i saw it on the wiki... hmm > but maybe I should head off to bed before I make more of an ass of myself > :P > Awww, don't do that! You entertain us when you make an ass of yourself! ;) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/eacaf1cd/attachment.html From mustafa.pk at gmail.com Tue Aug 24 10:24:41 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Tue, 24 Aug 2010 22:24:41 +0500 Subject: [Freeswitch-users] mod_conference In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57DC4655BC@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C57DC4655BC@cooper> Message-ID: go to bed Jay :) On Tue, Aug 24, 2010 at 7:03 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > conference xml_list :) > > > > type ?conference? for a complete list of commands. > > > > /Peter > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *jay binks > *Skickat:* den 24 augusti 2010 15:50 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] mod_conference > > > > geez that will teach me to post at 11:30pm... :( > > > > I Tried that and it didnt work... ( maybe a mistype I guess ) > > > > however, it dosnt seem that the tab completion on "conference" has this > listed.. > > ( they all seem to require a conference name ) > > > > and im not sure i saw it on the wiki... hmm > > but maybe I should head off to bed before I make more of an ass of myself > :P > > > > as a side note... awww.. > > no "as xml" on this one ( like show channels / calls ) > > would be nice for parsing > > > > J > > > > > > On Tue, Aug 24, 2010 at 11:36 PM, Seven Du wrote: > > conference list > > > On Tue, Aug 24, 2010 at 9:15 PM, jay binks wrote: > > is there a way ( ESL or API ) to list active conferences on an FS box. > > im playing with dynamic conferences, so they aren't pre-defined.. > > I COULD keep track using ESL but it would be nice to avoid tracking state > if > > possible. > > -- > > Sincerely > > > > Jay > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Sincerely > > Jay > > !DSPAM:4c73cfee32939223221216! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/37e35d36/attachment-0001.html From msc at freeswitch.org Tue Aug 24 10:47:22 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Aug 2010 10:47:22 -0700 Subject: [Freeswitch-users] ivr menu matching and execution problem In-Reply-To: <1282664502.13028.34.camel@marces.tc.commsmundi.com> References: <1282664502.13028.34.camel@marces.tc.commsmundi.com> Message-ID: Sounds to me like a possible solution would be adding a "break" option for "entry" so that the user could decide whether or not to break after the first match. However, I wonder if that's even necessary. My question to you is why would you have two different actions for keypress "9" and yet not want the second action ever to be executed? If the menu only ever executed the first occurrence of a given digit then what would be the value of having another occurrence of that same digit? In your example, under what conditions would pressing "9" execute "transfer 8 XML ..." ? There might be a less intrusive solution we could suggest if we knew more about why you have multiple different actions for a single keypress. -MC On Tue, Aug 24, 2010 at 8:41 AM, Antonio wrote: > During the configuration of an ivr menu and test i just realized that if > you have multiple actions with the same "digits" it executes the both > actions. > > The execution of the multiple actions is normal? > > The normal behavior should executed only the action that is match first, > reading the priority of the entries in the menu. > I look at the code, and in the function, switch_ivr_menu_execute, looks > like the loop that searches for the match/action should need a break > after a success match, to avoid the continues search for other matches. > > Thanks, > Ant?nio > > > /* configuration */ > > greet-long="/home/system/telephony/sounds/buraka.mp3" > exit-sound="/home/system/telephony/sounds/prompt/en/conf-invalid.gsm" > timeout ="2000" > inter-digit-timeout="2000" > max-failures="10" > digit-len="4" > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/acbe901e/attachment.html From juanito1982 at gmail.com Tue Aug 24 11:00:47 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Tue, 24 Aug 2010 20:00:47 +0200 Subject: [Freeswitch-users] Outbound codecs preference Message-ID: I can see when I try to stablish one B-leg call FS gives more precedence to remote codecs order. This way it is difficult to avoid transcoding in some situations where it could be possible. Is there any way to get a behaviour similar inbound-codec-negotiation=greedy for outbounds calls? What would be the best way to avoid transcoding? May be to use inbound-late-negotiation and rewrite codecs strings? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/645aa87d/attachment.html From tculjaga at gmail.com Tue Aug 24 11:16:36 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 24 Aug 2010 20:16:36 +0200 Subject: [Freeswitch-users] long DTMF In-Reply-To: <1282650110.2970.52.camel@luna.tc.commsmundi.com> References: <1282650110.2970.52.camel@luna.tc.commsmundi.com> Message-ID: It shouldn't be hard to do with a small lua script with an input > callback handling DTMFs. > > You could even create a small DTMF editor with a digit for "backspace" > character for example, that would erase the last digit, or a digit that > would say the current sequence.... > > Check out: > http://wiki.freeswitch.org/wiki/Lua#session:setInputCallback > > Fran?ois. > > Nice! i wrote a small script: function onInput(s, type, obj) if (type == "dtmf") then freeswitch.consoleLog("info", "DTMF Digit: " .. obj.digit .. "\n"); freeswitch.consoleLog("info", "DTMF Duration: " .. obj.duration .. "\n"); if (obj.duration > 2000) then freeswitch.consoleLog("info", "we got long DTMF \n"); end end end session:answer(); session:setInputCallback("onInput"); session:sleep(200); while (session:ready() == true) do session:sleep(100); end i run it in dialplan as: but it seems the lua scripts is blocking .... i never get to playback part. I need to make it run in background for the entire duration of the session how can i do it? T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/0ca10085/attachment.html From jeff at jefflenk.com Tue Aug 24 11:41:18 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 24 Aug 2010 11:41:18 -0700 (PDT) Subject: [Freeswitch-users] long DTMF In-Reply-To: References: <1282650110.2970.52.camel@luna.tc.commsmundi.com> Message-ID: <1282675278265-5458183.post@n2.nabble.com> Move the playback inside the script -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/long-DTMF-tp5456386p5458183.html Sent from the freeswitch-users mailing list archive at Nabble.com. From Victor at isptelecom.net Tue Aug 24 11:44:01 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Tue, 24 Aug 2010 14:44:01 -0400 Subject: [Freeswitch-users] How to stop FS from replying to OPTIONS? Message-ID: <4C7412F1.9000509@isptelecom.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/f2002b50/attachment.html From brian at freeswitch.org Tue Aug 24 11:57:54 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Aug 2010 13:57:54 -0500 Subject: [Freeswitch-users] How to stop FS from replying to OPTIONS? In-Reply-To: <4C7412F1.9000509@isptelecom.net> References: <4C7412F1.9000509@isptelecom.net> Message-ID: I don't think thats legal to ignore them. /b On Aug 24, 2010, at 1:44 PM, Victor Chukalovskiy wrote: > Is there a way to stop FS from replying to "OPTIONS" that other non-registered devices send? > That is, I'd like to make my FS server less visible to other people who ping me with "OPTIONS". > > Thank you, > Victor From mthakershi at gmail.com Tue Aug 24 11:58:21 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 24 Aug 2010 13:58:21 -0500 Subject: [Freeswitch-users] Inbound and then outbound call? Message-ID: Hello, It would be a great help if someone can guide me. 1. I would like to first receive a call, perform certain validations. (Able to do this via mod_managed application that handles call from dialplan). 2. Now, I would like to dial out to a PSTN number so that received call is connected to this new outbound number. How can this be done? Do I use Originate from within my .NET (mod_managed) code? Do I get charged for both incoming and outbound call until the entire session ends? Is there a way to receive call, validate and then sort of transfer and then terminate the received call so I do not get charged for both? Please help. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/19d14932/attachment.html From tculjaga at gmail.com Tue Aug 24 12:05:32 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 24 Aug 2010 21:05:32 +0200 Subject: [Freeswitch-users] long DTMF In-Reply-To: <1282675278265-5458183.post@n2.nabble.com> References: <1282650110.2970.52.camel@luna.tc.commsmundi.com> <1282675278265-5458183.post@n2.nabble.com> Message-ID: On Tue, Aug 24, 2010 at 8:41 PM, Jeff Lenk wrote: > > Move the playback inside the script > > not really feasible, all the logic is already done in DP... Im missing just this feature. what is the bgapi equivalent of: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/c213f9c7/attachment.html From kris at kriskinc.com Tue Aug 24 12:14:14 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 24 Aug 2010 15:14:14 -0400 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: <39C8C9F7-1CB1-4A73-92A6-5F7E029A6C5F@gmail.com> References: <39C8C9F7-1CB1-4A73-92A6-5F7E029A6C5F@gmail.com> Message-ID: On Tue, Aug 24, 2010 at 12:32 PM, Steven Ayre wrote: > > mod_voicemail is already open source. Yep. > If you want a feature not in it, specify what it is so that it can be added, or someone can chime in with a tip on how you can already do it. Or you could even supply a patch which adds the feature. I know I could. However, this development and patch would only meet *my* needs *right now*. What about my future needs? What about some other guy that loves FreeSWITCH but can't leave Avaya-style voicemail? When it really comes down to it an implementation in LUA would be far easier to customize. To be honest I've never really understood why Asterisk voicemail is a C module and why FreeSWITCH voicemail is a C module when both applications (especially FreeSWITCH) provide so many ways to accomplish the same goals without being tied into a C module. Mod_conference is muxing different audio streams at different sample rates, doing speaker detection, etc. That makes perfect sense to implement in C. Voicemail is basically an IVR app. How many people develop IVR apps in C? How many people use LUA, Perl, ESL, etc? The FreeSWITCH wiki, docs, etc encourage quick and easy application development using these existing interfaces (Lua, JS, ESL, etc). All I'm trying to do is build an application. This application just happens to be what most people usually refer to as "voicemail". > So the list of what's missing is really what's useful. I could provide a list of what is missing for Asterisk voicemail but my list today would never be the end-all-be-all of voicemail. An implementation in LUA would make it significantly easier for almost anyone to endlessly customize voicemail to meet their needs, now and in the future. Then again maybe I'm completely wrong about LUA for voicemail. I still think it's worth investigating. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kris at kriskinc.com Tue Aug 24 12:17:46 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 24 Aug 2010 15:17:46 -0400 Subject: [Freeswitch-users] How to stop FS from replying to OPTIONS? In-Reply-To: References: <4C7412F1.9000509@isptelecom.net> Message-ID: I think it's legal but you really should remove OPTIONS from Supported: headers returned by FreeSWITCH. I have mixed feelings about what the OP is trying to do and whether or not he really should but I suppose it kind of makes sense... Perhaps he could look at something like my sipdos script: http://blog.krisk.org/2008/07/sip-dosddos-mitigation.html It will do iptables string matching on SIP packets by method (OPTIONS, INVITE, REGISTER, etc) and allow you to rate limit and/or drop them. That would probably work. On Tue, Aug 24, 2010 at 2:57 PM, Brian West wrote: > I don't think thats legal to ignore them. > > /b > > On Aug 24, 2010, at 1:44 PM, Victor Chukalovskiy wrote: > >> Is there a way to stop FS from replying to "OPTIONS" that other non-registered devices send? >> That is, I'd like to make my FS server less visible to other people who ping me with "OPTIONS". >> >> Thank you, >> Victor > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Tue Aug 24 12:24:05 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Aug 2010 14:24:05 -0500 Subject: [Freeswitch-users] How to stop FS from replying to OPTIONS? In-Reply-To: References: <4C7412F1.9000509@isptelecom.net> Message-ID: <1D82C453-5039-4EAE-81B0-46DD511EE579@freeswitch.org> But still removing it from supported will result in an unsupported method response and won't ignore the OPTIONS packet totally. /b On Aug 24, 2010, at 2:17 PM, Kristian Kielhofner wrote: > I think it's legal but you really should remove OPTIONS from > Supported: headers returned by FreeSWITCH. > > I have mixed feelings about what the OP is trying to do and whether or > not he really should but I suppose it kind of makes sense... Perhaps > he could look at something like my sipdos script: > > http://blog.krisk.org/2008/07/sip-dosddos-mitigation.html > > It will do iptables string matching on SIP packets by method (OPTIONS, > INVITE, REGISTER, etc) and allow you to rate limit and/or drop them. > That would probably work. > > On Tue, Aug 24, 2010 at 2:57 PM, Brian West wrote: >> I don't think thats legal to ignore them. >> >> /b From kris at kriskinc.com Tue Aug 24 12:37:01 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 24 Aug 2010 15:37:01 -0400 Subject: [Freeswitch-users] How to stop FS from replying to OPTIONS? In-Reply-To: <1D82C453-5039-4EAE-81B0-46DD511EE579@freeswitch.org> References: <4C7412F1.9000509@isptelecom.net> <1D82C453-5039-4EAE-81B0-46DD511EE579@freeswitch.org> Message-ID: Brian, It sure will but I was saying that if he is going to black packets my METHOD he shouldn't leave the blocked METHODs in any Supported headers that may be sent in subsequent dialogs. That would be really confusing! On Tue, Aug 24, 2010 at 3:24 PM, Brian West wrote: > But still removing it from supported will result in an unsupported method response and won't ignore the OPTIONS packet totally. > > /b > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Tue Aug 24 12:56:24 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Aug 2010 14:56:24 -0500 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: <39C8C9F7-1CB1-4A73-92A6-5F7E029A6C5F@gmail.com> Message-ID: They are C because they are expected to load and work and there is 0 tolerance for missteps typically. a LUA one is ok but that's in the realm of the users. I would not ever ship an external app as part of the main distribution but naturally making your own external systems is why we made FS modular. If you want to do it in LUA it would probably end up in contrib. On Tue, Aug 24, 2010 at 2:14 PM, Kristian Kielhofner wrote: > On Tue, Aug 24, 2010 at 12:32 PM, Steven Ayre wrote: >> >> mod_voicemail is already open source. > > ?Yep. > >> If you want a feature not in it, specify what it is so that it can be added, or someone can chime in with a tip on how you can already do it. Or you could even supply a patch which adds the feature. > > ?I know I could. ?However, this development and patch would only meet > *my* needs *right now*. ?What about my future needs? ?What about some > other guy that loves FreeSWITCH but can't leave Avaya-style voicemail? > ?When it really comes down to it an implementation in LUA would be far > easier to customize. ?To be honest I've never really understood why > Asterisk voicemail is a C module and why FreeSWITCH voicemail is a C > module when both applications (especially FreeSWITCH) provide so many > ways to accomplish the same goals without being tied into a C module. > Mod_conference is muxing different audio streams at different sample > rates, doing speaker detection, etc. ?That makes perfect sense to > implement in C. ?Voicemail is basically an IVR app. ?How many people > develop IVR apps in C? ?How many people use LUA, Perl, ESL, etc? > > ?The FreeSWITCH wiki, docs, etc encourage quick and easy application > development using these existing interfaces (Lua, JS, ESL, etc). ?All > I'm trying to do is build an application. ?This application just > happens to be what most people usually refer to as "voicemail". > >> So the list of what's missing is really what's useful. > > ?I could provide a list of what is missing for Asterisk voicemail but > my list today would never be the end-all-be-all of voicemail. ?An > implementation in LUA would make it significantly easier for almost > anyone to endlessly customize voicemail to meet their needs, now and > in the future. > > ?Then again maybe I'm completely wrong about LUA for voicemail. ?I > still think it's worth investigating. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From lloydie.t at gmail.com Tue Aug 24 09:38:48 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Tue, 24 Aug 2010 17:38:48 +0100 Subject: [Freeswitch-users] Why not Ubuntu Message-ID: I have seen a few posts where it has been recommended not to use ubuntu to run freeswitch on. I'll admit I did not pay that much attention. Is there a really pressing reason not to or is it a preference of most users not to? I would prefer to use Ubuntu as that is what I am used to. Regards Lloydie T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/0944efe8/attachment.html From kris at kriskinc.com Tue Aug 24 13:08:00 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 24 Aug 2010 16:08:00 -0400 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: <39C8C9F7-1CB1-4A73-92A6-5F7E029A6C5F@gmail.com> Message-ID: Anthony, Understood and your logic makes perfect sense (as always). I think contrib would be a great place for it :). On Tue, Aug 24, 2010 at 3:56 PM, Anthony Minessale wrote: > They are C because they are expected to load and work and there is 0 > tolerance for missteps typically. > > a LUA one is ok but that's in the realm of the users. ?I would not > ever ship an external app as part of the main distribution but > naturally making your own external systems is why we made FS modular. > > If you want to do it in LUA it would probably end up in contrib. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Tue Aug 24 13:07:48 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Aug 2010 15:07:48 -0500 Subject: [Freeswitch-users] Why not Ubuntu In-Reply-To: References: Message-ID: Basically it's because Ubuntu updates itself constantly with newer kernels and newer system libs where FreeSWITCH is not heavily tested. The recommendation is only for those expecting to ask our community for help. We only have limited resources and we have found several instabilities with bleeding distros so it's a use at your own risk policy. There are some ubuntu users who package it and provide them for download. So if you run into some problems you will have a smaller pool from which to draw support. On Tue, Aug 24, 2010 at 11:38 AM, lloyd thomas wrote: > I have seen a few posts where it has been recommended not to use ubuntu to > run freeswitch on. I'll admit I did not pay that much attention. Is there a > really pressing reason not to or is it a preference of most users not to? > I would prefer to use Ubuntu as that is what I am used to. > > > Regards > Lloydie T > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Tue Aug 24 13:37:22 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Aug 2010 15:37:22 -0500 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: <39C8C9F7-1CB1-4A73-92A6-5F7E029A6C5F@gmail.com> Message-ID: I personally prefer the idea of making the existing one do more things. Other reasons its C is: It makes heavy use of the eventing system. Reads and writes files to the disk and merges files possibly of varying codecs. Seamlessly uses the DB Interops with SIP over messaging for MWI and message count. Most of the things you want to do from afar is exposed via API commands that automatically handle things like if you use the api to delete a VM it turns off the MWI yada yada. just $0.02 On Tue, Aug 24, 2010 at 3:08 PM, Kristian Kielhofner wrote: > Anthony, > > ?Understood and your logic makes perfect sense (as always). ?I think > contrib would be a great place for it :). > > On Tue, Aug 24, 2010 at 3:56 PM, Anthony Minessale > wrote: >> They are C because they are expected to load and work and there is 0 >> tolerance for missteps typically. >> >> a LUA one is ok but that's in the realm of the users. ?I would not >> ever ship an external app as part of the main distribution but >> naturally making your own external systems is why we made FS modular. >> >> If you want to do it in LUA it would probably end up in contrib. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From robert.hadley at teotech.com Tue Aug 24 14:03:04 2010 From: robert.hadley at teotech.com (Robert Hadley) Date: Tue, 24 Aug 2010 14:03:04 -0700 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemailEmulation In-Reply-To: References: <4C73C4AF.6000100@communicatefreely.net> Message-ID: Voicemail is of high interest to us too. We have a few ideas about improvements to mod_voicemail. I have entered a couple JIRA requests so you could look there for some more ideas for improvements to make. http://jira.freeswitch.org/secure/IssueNavigator.jspa?reset=true&mode=hide&p id=10021&sorter/order=DESC&sorter/field=priority&resolution=-1&component=100 77 I have submitted a patch to add a per user operator extension. I have a patch to add main menu access from other menus. There are open requests to fix the repeated ?Saved? message and to reset the greeting back to the default. Please feel free to contact me off list for more ideas. Also, I know C pretty good :) Thanks, Robert -----Original Message----- From: Kristian Kielhofner [mailto:kris at kriskinc.com] Sent: Tuesday, August 24, 2010 9:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] New Bounty (again): Improved Mod_voicemailEmulation Tim, Thanks for replying. I'm in contact with someone who has a fairly complete voicemail implementation written in LUA. Hopefully we can work out an arrangement to open source that code so we can all work on it and adapt it for our needs. If everything goes as planned I'd love to have you help out. On Tue, Aug 24, 2010 at 9:10 AM, Tim St. Pierre wrote: > ?So, what are your thoughts? >> > I think I would be interested in helping if I can, as I'm more or less in the same boat, with an > existing user base on Asterisk. ?I would like to make things a little bit more configurable (not > that they aren't already). > > I don't have any experience working with C, but I am getting to know LUA, PHP, SQL, and a few > others. I'm hoping I can do something helpful. > > I have moved a lot of logic outside of the application and into a lua script to do what I want. > Here's some of the features I have added. ?Let me know if this is something we could roll into a new > module. > > -Configurable cause code to greeting mappings > > This lets you define an alternate greeting that will play if your phone is busy, in do not disturb, > not registered, etc. ?It gives similar functionality to the busy/unavail greetings in Asterisk, but > with a lot more flexibility. ?I also defined a mapping for "vm-direct", as on our PBX, you can > transfer a call to **EXTEN to go straight to voicemail. ?On a NO_ANSWER cause, the selected greeting > is played, which allows for vacation messages. ?If the mappings are set in the database, but the > alternate greeting hasn't been recorded, it will use the selected main greeting, or play "the person > at extension ... is unavailable". > > -Configurable 0 IVR routing > > I have put a bit more functionality in the 0 routing. ?In our Asterisk implementation, we had 0 go > to reception, and # go to the users cell phone. ?A lot of people use this. ?There isn't a good way > to do this in Freeswitch, so we came up with a different option. ?In the VM config database that we > use, I have an IVR enable flag, as well as an extension slot for digits 0-9. ?If the IVR is > disabled, pressing 0 transfers to the extension in the 0 slot. ?If the IVR is enabled, it uses > greeting 9 as the menu, and will let the caller choose an option, transferring them to the extension > set in the database. ?This lets a user do something like this: > > "You have reached Tim, but I'm away from my desk at the moment. ?Please leave a message after the > tone or press 0 for immediate assistance". > ->"For technical support, press 1. ?To reach my assistant, press 2. ?If you would like to try my > cell phone, press 3". > > > I would also like to see: > An option NOT to delete the voice message when you e-mail it from the vm menu. > A mechanism to record your name that the dial by name directory module can use. > An option to skip the message details before you play them. > > I'm sure there are others. > > Let me know if you think any of these options are useful. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From mrene_lists at avgs.ca Tue Aug 24 15:21:56 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 24 Aug 2010 18:21:56 -0400 Subject: [Freeswitch-users] Predefined hangup cause possible for originate? In-Reply-To: <4C73CE8B.6060805@ewetel.de> References: <4C73CE8B.6060805@ewetel.de> Message-ID: <8D2E7E31-342B-41D4-954A-3A97D4AF772A@avgs.ca> Hi, There is already a cause for timeout. You can decide to hangup yourself after that with the cause you want. Gr?sse, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-08-24, at 9:52 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > is it possible, to load originate command with a predefined hangup cause > e.g. "completed_elsewhere" when originate-call-timeout occured? > > regards > helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFMc86L4tZeNddg3dwRAuWEAKCzxADKkpv6x/gc8Xkmgy8QaGhXUACfSr8R > Gh+h7TEqV8KR7Xe+GdrS7wM= > =pLvI > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From david.ponzone at ipeva.fr Tue Aug 24 16:03:03 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 01:03:03 +0200 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: References: Message-ID: <989CA9B5-B6AD-43B0-89E1-595F37E9DF0E@ipeva.fr> Malay, in most countries, you are not charged for a call you receive. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 24/08/2010 ? 20:58, Malay Thakershi a ?crit : > Hello, > > It would be a great help if someone can guide me. > > 1. I would like to first receive a call, perform certain > validations. (Able to do this via mod_managed application that > handles call from dialplan). > > 2. Now, I would like to dial out to a PSTN number so that received > call is connected to this new outbound number. > > How can this be done? Do I use Originate from within my .NET > (mod_managed) code? > > Do I get charged for both incoming and outbound call until the > entire session ends? Is there a way to receive call, validate and > then sort of transfer and then terminate the received call so I do > not get charged for both? > > Please help. > > Thank you. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/af1dade6/attachment.html From acosgrov at gmail.com Tue Aug 24 16:06:40 2010 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Tue, 24 Aug 2010 19:06:40 -0400 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: References: Message-ID: <1282691200.4420.23.camel@anthony-desktop> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: > Hello, > > > It would be a great help if someone can guide me. > > > 1. I would like to first receive a call, perform certain validations. > (Able to do this via mod_managed application that handles call from > dialplan). > That should not be a problem, I don't know your requirements so can't provide a full answer. > > 2. Now, I would like to dial out to a PSTN number so that received > call is connected to this new outbound number. > This can be done and is called hairpinning. > > How can this be done? Do I use Originate from within my .NET > (mod_managed) code? > Yes, you would be bridging the two legs like any normal call. Instead of going to an endpoint you're going back out over the PSTN. > > Do I get charged for both incoming and outbound call until the entire > session ends? Is there a way to receive call, validate and then sort > of transfer and then terminate the received call so I do not get > charged for both? That would depend on your provider but most likely yes. As for terminating one end after validation that is not going to happen. A leg terminates to you on an agreed fee schedule. No carrier that I know of supports that. Now if you kept everything SIP... you could do a transfer after the validation. Anthony C. From david.ponzone at ipeva.fr Tue Aug 24 16:08:41 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 01:08:41 +0200 Subject: [Freeswitch-users] Outbound codecs preference In-Reply-To: References: Message-ID: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> The best way is probably to use inbound-late-negotiation, and in the dialplan before bridging to add: David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 24/08/2010 ? 20:00, Juan Antonio Iba?ez Santorum a ?crit : > I can see when I try to stablish one B-leg call FS gives more > precedence to remote codecs order. This way it is difficult to avoid > transcoding in some situations where it could be possible. Is there > any way to get a behaviour similar inbound-codec-negotiation=greedy > for outbounds calls? What would be the best way to avoid > transcoding? May be to use inbound-late-negotiation and rewrite > codecs strings? > > Regards > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/9d4fe496/attachment.html From brian at freeswitch.org Tue Aug 24 16:11:27 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Aug 2010 18:11:27 -0500 Subject: [Freeswitch-users] Outbound codecs preference In-Reply-To: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> References: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> Message-ID: Is this documented on the wiki anywhere? /b On Aug 24, 2010, at 6:08 PM, David Ponzone wrote: > The best way is probably to use inbound-late-negotiation, and in the dialplan before bridging to add: > > > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com From david.ponzone at ipeva.fr Tue Aug 24 16:31:29 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 01:31:29 +0200 Subject: [Freeswitch-users] Outbound codecs preference In-Reply-To: References: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> Message-ID: <396C2AA2-9F51-46CA-AE37-685EA43B2B4B@ipeva.fr> Brian, is this your very polite way to remind me that I never finished rewriting the codec negotiation page ? :) If it is, you're absolutely true! I must admit I needed to be really confident with late-neg before rewriting it, and it's been only a few days since I "get" it. So I hope to be able to rewrite the section on late-neg ASAP, in order to cascade back what I learned during my "dark journey in understanding FreeSWITCH's codec negotiation". David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 01:11, Brian West a ?crit : > Is this documented on the wiki anywhere? > > /b > > On Aug 24, 2010, at 6:08 PM, David Ponzone wrote: > >> The best way is probably to use inbound-late-negotiation, and in >> the dialplan before bridging to add: >> >> >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/921f3aff/attachment.html From chat2jesse at gmail.com Tue Aug 24 16:42:00 2010 From: chat2jesse at gmail.com (jesse) Date: Tue, 24 Aug 2010 16:42:00 -0700 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : At some point before July 8, 2009, mod_php was removed from the trunk. There don't appear to be any concrete plans to reintroduce mod_php, so if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket or the PHP ESL library. On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale wrote: > We never have had PHP in our code distribution. > > > On Mon, Aug 23, 2010 at 4:39 PM, jesse wrote: >> The current FS version doesn't include source code for PHP. >> What are the steps to install it on top of my current version? ?where >> to get the source? how to build and install? >> >> Iif you guys want to keep the new system slim by get rid of PHP, >> please at least keep a doc about how to add it in case of need. >> >> thanks! >> >> -jesse >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at microcomaustralia.com.au Tue Aug 24 16:47:40 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Wed, 25 Aug 2010 09:47:40 +1000 Subject: [Freeswitch-users] incoming pots ring Message-ID: Hello, Something that has been bugging me for a while is the behaviour when receiving an incoming ring from a pots connection. If the incoming connection stops ringing, it takes Freeswitch ages for it to realize. In fact, I have found even if the phone stops ringing after the first ring, Freeswitch doesn't realize until after it activates voicemail (IIRC 20 second delay). I can understand that Freeswitch can't work out the instant the phone has stopped ringing, but the delay seems to be excessively long - is it possible to tweak it? If so how? I am using freetdm with a TDM400 card. Thanks -- Brian May From chat2jesse at gmail.com Tue Aug 24 16:56:58 2010 From: chat2jesse at gmail.com (jesse) Date: Tue, 24 Aug 2010 16:56:58 -0700 Subject: [Freeswitch-users] Lua script issue. Message-ID: I have a bridge.lua script: phone1 = argv[1]; phone2 = argv[2]; dialstring1 = "sofia/gateway/xyz.com/" .. phone1; dialstring2 = "sofia/gateway/xyz.com7" .. phone2; session1 = freeswitch.Session(dialstring1); session2 = freeswitch.Session(dialstring2, session1); freeswitch.bridge(session1, session2); freeswitch at xyz.com> luarun bridge.lua 16502222222 16503333333 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] +OK freeswitch at xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer sofia/external/16502222222! 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 sofia/external/16503333333 Update Callee ID to "16503333333" <16503333333> 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer sofia/external/16503333333! 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not ready 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session 25 (sofia/external/16503333333) Ended 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/16503333333 [CS_DESTROY] 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session 24 (sofia/external/16502222222) Ended 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/16502222222 [CS_DESTROY] As you can see the call gets dropped immediately after couple rings. however, it works well if i do like this: originate sofia/gateway/xyz.com/16502222222 &bridge(sofia/gateway/xyz.com/16503333333) What is the reason Lua script will fail? any difference between the two approaches? thanks! jesse From mnhassan at usa.net Tue Aug 24 16:56:05 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Wed, 25 Aug 2010 05:56:05 +0600 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: I got my phpmod installed and working great! Did you follow: http://wiki.freeswitch.org/wiki/Esl#Installation? Please mention at which steps you're seeing an error. Regards HASSAN On Wed, Aug 25, 2010 at 05:42, jesse wrote: > the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : > > At some point before July 8, 2009, mod_php was removed from the trunk. > There don't appear to be any concrete plans to reintroduce mod_php, so > if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket or > the PHP ESL library. > > > On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale > wrote: > > We never have had PHP in our code distribution. > > > > > > On Mon, Aug 23, 2010 at 4:39 PM, jesse wrote: > >> The current FS version doesn't include source code for PHP. > >> What are the steps to install it on top of my current version? where > >> to get the source? how to build and install? > >> > >> Iif you guys want to keep the new system slim by get rid of PHP, > >> please at least keep a doc about how to add it in case of need. > >> > >> thanks! > >> > >> -jesse > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/1b15de31/attachment-0001.html From chat2jesse at gmail.com Tue Aug 24 17:07:34 2010 From: chat2jesse at gmail.com (jesse) Date: Tue, 24 Aug 2010 17:07:34 -0700 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: during installation: make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' make[1]: Nothing to be done for `all'. make[1]: Leaving directory `/tmp/freeswitch/libs/esl/php' make -C php install PHP Warning: PHP Startup: ESL: Unable to initialize module Module compiled with module API=20090626, debug=0, thread-safety=0 PHP compiled with module API=20060613, debug=0, thread-safety=0 These options need to match in Unknown on line 0 make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' cp ESL.so /usr/lib/php5/20060613 cp ESL.php /usr/share/php Then when I open the PHP script from browser, I always get Unable to initialize module Module compiled with module API=20090626, debug=0, thread-safety=0 PHP compiled with module API=20060613, debug=0, thread-safety=0 These options need to match in /usr/share/php/ESL.php on line 23 any one is good at PHP can fix this issue? I guess the root cause is that I have two PHP installed and my system is totally messed. I tried to remove the new php installation, but still the same result. On Tue, Aug 24, 2010 at 4:56 PM, Nyamul Hassan wrote: > I got my phpmod installed and working great! ?Did you follow: > http://wiki.freeswitch.org/wiki/Esl#Installation? > Please mention at which steps you're seeing an error. > Regards > HASSAN > > > On Wed, Aug 25, 2010 at 05:42, jesse wrote: >> >> the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : >> >> At some point before July 8, 2009, mod_php was removed from the trunk. >> There don't appear to be any concrete plans to reintroduce mod_php, so >> if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket or >> the PHP ESL library. >> >> >> On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale >> wrote: >> > We never have had PHP in our code distribution. >> > >> > >> > On Mon, Aug 23, 2010 at 4:39 PM, jesse wrote: >> >> The current FS version doesn't include source code for PHP. >> >> What are the steps to install it on top of my current version? ?where >> >> to get the source? how to build and install? >> >> >> >> Iif you guys want to keep the new system slim by get rid of PHP, >> >> please at least keep a doc about how to add it in case of need. >> >> >> >> thanks! >> >> >> >> -jesse >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From michael.scheidell at secnap.com Tue Aug 24 17:17:48 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Tue, 24 Aug 2010 20:17:48 -0400 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 Message-ID: <4C74612C.8010009@secnap.com> I am in need of putting a sbc or sip proxy in front of our existing sip solution to help solve a very specific problem. The SIP Trunk provider won't send to anything but port 5060, and the existing sip solution listens to port 5080 ONLY. (background: sipxecs. yes, they are using freeswitch inside for a couple of things) I have tried simple port redirection with iptables, and while it SEEMS to work, the sip packets don't. so, does anyone know of a simple solution? I don't really want to replicate all the users, dial plans, sip trunk accounts on freeswitch. Just basically, anything coming in from a certain ip to the freeswitch port 5060, 'sipproxy' it to port 5080 on the internal system. then, anything sent out from the one certain internal ip, sourced on port 5080 to port 5060, send it to the external sip trunk provider. Using ip based authentication all around. I assume using an ip based condition with a FWD, (but the FWD would need to be generic) if call comes in from siptrunkprovider to 1000 at freeswitch.secnap.com:5060 , I want it transparently forwarded to 1000 at sipx.secnap.com:5080 if an internal call goes out to 5619995000 at freeswitch.com:5060, I want it transparently forwarded to 5619995000 at siptrunkprovider.com:5060 (yes, I tried simple port mapping. worked with tcp on http, and ALMOST worked with udp on tftp, but audio was mangled) sipx suggested purchasing an ingate siperaor. hard to justify $4 or $5K when I could replace sipx with a commercial system for that price. (yes, I could move everything to freeswitch, and I might, but I am under a deadline ) -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100824/73412f0f/attachment.html From anthony.minessale at gmail.com Tue Aug 24 17:43:51 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Aug 2010 19:43:51 -0500 Subject: [Freeswitch-users] incoming pots ring In-Reply-To: References: Message-ID: it's a lack of hangup detection on the line itself. you can try using tone_detect on your inbound dialplan This will listen for busy signal in the background and hangup if it encounters it. On Tue, Aug 24, 2010 at 6:47 PM, Brian May wrote: > Hello, > > Something that has been bugging me for a while is the behaviour when > receiving an incoming ring from a pots connection. > > If the incoming connection stops ringing, it takes Freeswitch ages for > it to realize. In fact, I have found even if the phone stops ringing > after the first ring, Freeswitch doesn't realize until after it > activates voicemail (IIRC 20 second delay). > > I can understand that Freeswitch can't work out the instant the phone > has stopped ringing, but the delay seems to be excessively long - is > it possible to tweak it? If so how? > > I am using freetdm with a TDM400 card. > > Thanks > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From brian at microcomaustralia.com.au Tue Aug 24 17:57:24 2010 From: brian at microcomaustralia.com.au (Brian May) Date: Wed, 25 Aug 2010 10:57:24 +1000 Subject: [Freeswitch-users] incoming pots ring In-Reply-To: References: Message-ID: On 25 August 2010 10:43, Anthony Minessale wrote: > it's a lack of hangup detection on the line itself. > you can try using tone_detect on your inbound dialplan > > > > This will listen for busy signal in the background and hangup if it > encounters it. No, that is only after the call is answered. I am talking about a caller that hangs up before it is answered. -- Brian May From mnhassan at usa.net Tue Aug 24 18:12:18 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Wed, 25 Aug 2010 07:12:18 +0600 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: * enter the /libs/esl directory * run "make" * run "make phpmod" If you face error in any step above, then please output the "full output" into http://pastebin.freeswitch.org and let us know the pastebin link. Regards HASSAN On Wed, Aug 25, 2010 at 06:07, jesse wrote: > during installation: > > make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' > make[1]: Nothing to be done for `all'. > make[1]: Leaving directory `/tmp/freeswitch/libs/esl/php' > make -C php install > PHP Warning: PHP Startup: ESL: Unable to initialize module > Module compiled with module API=20090626, debug=0, thread-safety=0 > PHP compiled with module API=20060613, debug=0, thread-safety=0 > These options need to match > in Unknown on line 0 > make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' > cp ESL.so /usr/lib/php5/20060613 > cp ESL.php /usr/share/php > > Then when I open the PHP script from browser, I always get > > Unable to initialize module Module compiled with module API=20090626, > debug=0, thread-safety=0 PHP compiled with module API=20060613, > debug=0, thread-safety=0 These options need to match in > /usr/share/php/ESL.php on line 23 > > any one is good at PHP can fix this issue? I guess the root cause is > that I have two PHP installed and my system is totally messed. I > tried to remove the new php installation, but still the same result. > > > > On Tue, Aug 24, 2010 at 4:56 PM, Nyamul Hassan wrote: > > I got my phpmod installed and working great! Did you follow: > > http://wiki.freeswitch.org/wiki/Esl#Installation? > > Please mention at which steps you're seeing an error. > > Regards > > HASSAN > > > > > > On Wed, Aug 25, 2010 at 05:42, jesse wrote: > >> > >> the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : > >> > >> At some point before July 8, 2009, mod_php was removed from the trunk. > >> There don't appear to be any concrete plans to reintroduce mod_php, so > >> if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket or > >> the PHP ESL library. > >> > >> > >> On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale > >> wrote: > >> > We never have had PHP in our code distribution. > >> > > >> > > >> > On Mon, Aug 23, 2010 at 4:39 PM, jesse wrote: > >> >> The current FS version doesn't include source code for PHP. > >> >> What are the steps to install it on top of my current version? where > >> >> to get the source? how to build and install? > >> >> > >> >> Iif you guys want to keep the new system slim by get rid of PHP, > >> >> please at least keep a doc about how to add it in case of need. > >> >> > >> >> thanks! > >> >> > >> >> -jesse > >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/8082f8fd/attachment.html From david.ponzone at ipeva.fr Tue Aug 24 18:43:25 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 03:43:25 +0200 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <4C74612C.8010009@secnap.com> References: <4C74612C.8010009@secnap.com> Message-ID: <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> Michael, what you want to achieve is pretty simple to do with FreeSWITCH, and is quite close from what I do. To get you started, here are the major changes you need to apply to the default conf: -keep external SIP profile but change its port to 5060 -modify internal SIP profile to be identical to external profile and bind it to 5080 (you could also do everything with external profile, but I think it's simpler and easier to understand to have 2 profiles) - > this is required because the default internal profile expects registration Then you end up with 2 profiles using the public context. Then if you prefer the simple way, you can change internal to use another context "younameit" (could be "internal"). Or you do like me: you split calls coming into public to a specific contex, based on source IP with such extensions in public.xml: Then you can add 2 dialplans, provider and sipxecs (I add them in specific files I put in dialplan/, I find it more readable). And you just have to add the right extensions in those dialplans to bridge calls to the other side. That's the easy part. In other words, an Ingate Siparator for this would be quite overkill. Remember that FS is not a proxy, so if your SipXecs sends some specific headers, you may need to deal with them and put them back on the B-leg to the provider. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 02:17, Michael Scheidell a ?crit : > I am in need of putting a sbc or sip proxy in front of our existing > sip solution to help solve a very specific problem. > The SIP Trunk provider won't send to anything but port 5060, and the > existing sip solution listens to port 5080 ONLY. > (background: sipxecs. yes, they are using freeswitch inside for a > couple of things) > > I have tried simple port redirection with iptables, and while it > SEEMS to work, the sip packets don't. > > so, does anyone know of a simple solution? I don't really want to > replicate all the users, dial plans, sip trunk accounts on freeswitch. > > Just basically, anything coming in from a certain ip to the > freeswitch port 5060, 'sipproxy' it to port 5080 on the internal > system. > then, anything sent out from the one certain internal ip, sourced on > port 5080 to port 5060, send it to the external sip trunk provider. > > Using ip based authentication all around. I assume using an ip based > condition with a FWD, (but the FWD would need to be generic) > if call comes in from siptrunkprovider to > 1000 at freeswitch.secnap.com:5060 , I want it transparently forwarded > to 1000 at sipx.secnap.com:5080 > > if an internal call goes out to 5619995000 at freeswitch.com:5060, I > want it transparently forwarded to 5619995000 at siptrunkprovider.com: > 5060 > > (yes, I tried simple port mapping. worked with tcp on http, and > ALMOST worked with udp on tftp, but audio was mangled) > > sipx suggested purchasing an ingate siperaor. hard to justify $4 or > $5K when I could replace sipx with a commercial system for that price. > (yes, I could move everything to freeswitch, and I might, but I am > under a deadline ) > > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > > | SECNAP Network Security Corporation > Certified SNORT Integrator > 2008-9 Hot Company Award Winner, World Executive Alliance > Five-Star Partner Program 2009, VARBusiness > Best in Email Security,2010: Network Products Guide > King of Spam Filters, SC Magazine 2008 > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see http://www.secnap.com/products/spammertrap/ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/ff7a53a4/attachment-0001.html From 12ukwn at gmail.com Tue Aug 24 19:11:52 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 25 Aug 2010 04:11:52 +0200 Subject: [Freeswitch-users] php esl compilation PBs Message-ID: <20100825041152.0c5a17c4@anubis.defcon1> Debian sid ========== Hi list, I'm trying to figure out a way to compile ESL for PHP on my system. I was obliged to copy ../src/include/* into .../esl/php and modify them a bit, but I still have an error: g++ -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:2583: attention : deprecated conversion from string constant to ?char*? g++ esl_wrap.o -lcrypt -lcrypt -lonig -ldb -lqdbm -lrt -lm -ldl -lnsl -lcrypt -lcrypt -lpthread -o ESL.so -L. /usr/bin/ld: cannot find -lonig collect2: ld a retourn? 1 code d'?tat d'ex?cution make: *** [ESL.so] Error 1 Apparently it is a missing library (onig), but which one? JY -- On a clear disk you can seek forever. From fs-list at communicatefreely.net Tue Aug 24 19:15:39 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Tue, 24 Aug 2010 22:15:39 -0400 Subject: [Freeswitch-users] incoming pots ring In-Reply-To: References: Message-ID: <4C747CCB.6080903@communicatefreely.net> The problem is in the drivers or the config for the analog interface. If the CO switch supports it, a CPC signal gets sent down the line to tell your interface to cancel. This was called fxs_ks in zaptel. It should be possible to implement a timer that considers it canceled if it goes beyond a preset time between rings and nobody has answered. It's 6 seconds in North America. This goes back to the driver though, and isn't really a freeswitch problem I don't imagine. -Tim Brian May wrote: > On 25 August 2010 10:43, Anthony Minessale wrote: >> it's a lack of hangup detection on the line itself. >> you can try using tone_detect on your inbound dialplan >> >> >> >> This will listen for busy signal in the background and hangup if it >> encounters it. > > No, that is only after the call is answered. > > I am talking about a caller that hangs up before it is answered. From mike at jerris.com Tue Aug 24 19:46:18 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Aug 2010 22:46:18 -0400 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: <680732BE-748F-4CB5-AD86-24A0D51664F2@jerris.com> mod_php != php. PHP was never in tree. Patches to improve the configure detection and installation of all the esl language mods are needed and gladly accepted. Mike On Aug 24, 2010, at 7:42 PM, jesse wrote: > the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : > > At some point before July 8, 2009, mod_php was removed from the trunk. > There don't appear to be any concrete plans to reintroduce mod_php, so > if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket or > the PHP ESL library. > > > On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale > wrote: >> We never have had PHP in our code distribution. >> >> >> On Mon, Aug 23, 2010 at 4:39 PM, jesse wrote: >>> The current FS version doesn't include source code for PHP. >>> What are the steps to install it on top of my current version? where >>> to get the source? how to build and install? >>> >>> Iif you guys want to keep the new system slim by get rid of PHP, >>> please at least keep a doc about how to add it in case of need. From juanbackson at gmail.com Tue Aug 24 20:45:24 2010 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 25 Aug 2010 11:45:24 +0800 Subject: [Freeswitch-users] mod_callcenter vs mod_fifo Message-ID: Hi, I am doing an investigation of using freeswitch for a call center app. It seems like both mod_fifo and mod_callcenter can pretty much do the same thing. Can someone share a comparison analysis between the two? Which one is better in terms of functionality and efficiency? Thanks jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/cdb2c791/attachment.html From mnhassan at usa.net Tue Aug 24 20:45:36 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Wed, 25 Aug 2010 09:45:36 +0600 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: <680732BE-748F-4CB5-AD86-24A0D51664F2@jerris.com> References: <680732BE-748F-4CB5-AD86-24A0D51664F2@jerris.com> Message-ID: Yes, but the original question was about PHP & ESL, so I tried to help him the way I got it working. Regards, HASSAN On 2010-08-25, Michael Jerris wrote: > mod_php != php. PHP was never in tree. Patches to improve the configure > detection and installation of all the esl language mods are needed and > gladly accepted. > > Mike > > > On Aug 24, 2010, at 7:42 PM, jesse wrote: > >> the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : >> >> At some point before July 8, 2009, mod_php was removed from the trunk. >> There don't appear to be any concrete plans to reintroduce mod_php, so >> if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket or >> the PHP ESL library. >> >> >> On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale >> wrote: >>> We never have had PHP in our code distribution. >>> >>> >>> On Mon, Aug 23, 2010 at 4:39 PM, jesse wrote: >>>> The current FS version doesn't include source code for PHP. >>>> What are the steps to install it on top of my current version? where >>>> to get the source? how to build and install? >>>> >>>> Iif you guys want to keep the new system slim by get rid of PHP, >>>> please at least keep a doc about how to add it in case of need. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From mthakershi at gmail.com Tue Aug 24 22:22:04 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Wed, 25 Aug 2010 00:22:04 -0500 Subject: [Freeswitch-users] How to stop SPAM calls? In-Reply-To: References: <8D99EAE7-10F4-48A3-8C9F-54902C718451@freeswitch.org> <5DFEE95D-9535-4F66-BB70-AEF6803273BB@ipeva.fr> <2FBF0F98-23E9-416E-BC83-7EBD9F257CD0@freeswitch.org> <89D52435-BB9F-4AC8-8D61-AF7F9BE3BAE2@ipeva.fr> <54AB9D1D-BFE7-4685-B098-ECB3806C3666@ipeva.fr> <4C72FD79.5030401@isptelecom.net> <4C731F60.6030801@isptelecom.net> Message-ID: Victor, your solution seems to be working. Thank you and thank you all for help. I did the following: acl.conf.xml In my sip profile file: Thank you. On Tue, Aug 24, 2010 at 9:39 AM, Malay Thakershi wrote: > Not a problem. Thanks for help. > > Will test as you've suggested. > > On Aug 23, 2010 8:33 PM, "Victor Chukalovskiy" > wrote: > > Malay, > > I believe full configuration is too big to be posted here, > however my sip profile has following relevant lines: > > > > > My acl.conf.xml has following lines: > > > > > > As you see, this creates a simple ACL that will "deny" everything but > address range defined by A.B.C.D/24 > My sip profile makes use of this ACL. > > In your configuration you seem to miss a couple things: > 1) You use reserved ACL name acl.auto. Please use something else and > non-default. > 2) If your ACL is "deny" by default, all nodes should be "allow". And vice > verse, if your ACL is "allow" by default then nodes only make sence if they > are set to "deny". This depends on what your want to do :-) > > In your case try: > > > > > > And then point apply-inbound-acl to this new "Malay_ACL". > > It could be useful to consult following pages: > http://wiki.freeswitch.org/wiki/Acl > http://en.wikipedia.org/wiki/Classless_Inter-Domain_Routing > > -Victor > > > > On -10/01/37 02:59 PM, Malay Thakershi wrote: > > > > Thank you. Could you please share your configura... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/c8ad262b/attachment.html From mthakershi at gmail.com Tue Aug 24 22:33:20 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Wed, 25 Aug 2010 00:33:20 -0500 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: <1282691200.4420.23.camel@anthony-desktop> References: <1282691200.4420.23.camel@anthony-desktop> Message-ID: I am currently looking at Vitelity. They have 1.44 cents per minute. They do charge for incoming/outgoing both. I am not literally dialing out to a phone. I want to dial a mobile number or any other US number for that matter. Here is what I intent to do: 1. A US phone dials to Vitelity number -- comes to my FS box 2. I validate few things 3. Dial another US number and connect received call to that one So as you said I will be charged 2 * 1.44 (since I can't terminate the arrived call after validation). Is there any other way to the sequence I have specified above? Are there providers similar to Vitelity but cheaper (with relatively same features)? Thank you. On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove wrote: > On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: > > Hello, > > > > > > It would be a great help if someone can guide me. > > > > > > 1. I would like to first receive a call, perform certain validations. > > (Able to do this via mod_managed application that handles call from > > dialplan). > > > > That should not be a problem, I don't know your requirements so can't > provide a full answer. > > > > > 2. Now, I would like to dial out to a PSTN number so that received > > call is connected to this new outbound number. > > > > This can be done and is called hairpinning. > > > > > How can this be done? Do I use Originate from within my .NET > > (mod_managed) code? > > > > Yes, you would be bridging the two legs like any normal call. Instead of > going to an endpoint you're going back out over the PSTN. > > > > > Do I get charged for both incoming and outbound call until the entire > > session ends? Is there a way to receive call, validate and then sort > > of transfer and then terminate the received call so I do not get > > charged for both? > > That would depend on your provider but most likely yes. As for > terminating one end after validation that is not going to happen. A leg > terminates to you on an agreed fee schedule. No carrier that I know of > supports that. Now if you kept everything SIP... you could do a transfer > after the validation. > > > > Anthony C. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/cba91a27/attachment-0001.html From mustafa.pk at gmail.com Tue Aug 24 22:44:17 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Wed, 25 Aug 2010 10:44:17 +0500 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: References: <1282691200.4420.23.camel@anthony-desktop> Message-ID: Hi, i am wondering what will happen if you send a SIP REFER (transfer) after validating leg-a without actually answering the call; will it still cost you 2 * 1.44 :/ -mustafa On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi wrote: > I am currently looking at Vitelity. They have 1.44 cents per minute. They > do charge for incoming/outgoing both. > > I am not literally dialing out to a phone. I want to dial a mobile number > or any other US number for that matter. > > Here is what I intent to do: > 1. A US phone dials to Vitelity number -- comes to my FS box > 2. I validate few things > 3. Dial another US number and connect received call to that one > > So as you said I will be charged 2 * 1.44 (since I can't terminate the > arrived call after validation). > > Is there any other way to the sequence I have specified above? > > Are there providers similar to Vitelity but cheaper (with relatively same > features)? > > Thank you. > > On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove wrote: > >> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: >> > Hello, >> > >> > >> > It would be a great help if someone can guide me. >> > >> > >> > 1. I would like to first receive a call, perform certain validations. >> > (Able to do this via mod_managed application that handles call from >> > dialplan). >> > >> >> That should not be a problem, I don't know your requirements so can't >> provide a full answer. >> >> > >> > 2. Now, I would like to dial out to a PSTN number so that received >> > call is connected to this new outbound number. >> > >> >> This can be done and is called hairpinning. >> >> > >> > How can this be done? Do I use Originate from within my .NET >> > (mod_managed) code? >> > >> >> Yes, you would be bridging the two legs like any normal call. Instead of >> going to an endpoint you're going back out over the PSTN. >> >> > >> > Do I get charged for both incoming and outbound call until the entire >> > session ends? Is there a way to receive call, validate and then sort >> > of transfer and then terminate the received call so I do not get >> > charged for both? >> >> That would depend on your provider but most likely yes. As for >> terminating one end after validation that is not going to happen. A leg >> terminates to you on an agreed fee schedule. No carrier that I know of >> supports that. Now if you kept everything SIP... you could do a transfer >> after the validation. >> >> >> >> Anthony C. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/2dddbf42/attachment.html From chat2jesse at gmail.com Tue Aug 24 23:39:50 2010 From: chat2jesse at gmail.com (jesse) Date: Tue, 24 Aug 2010 23:39:50 -0700 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: HASSAN: my pasted error message in the previous e-mail was generated from : make phpmod-install. -jesse On Tue, Aug 24, 2010 at 6:12 PM, Nyamul Hassan wrote: > * ?enter the /libs/esl directory > * ?run "make" > * ?run "make phpmod" > If you face error in any step above, then please output the "full output" > into > http://pastebin.freeswitch.org > and let us know the pastebin link. > Regards > HASSAN > > On Wed, Aug 25, 2010 at 06:07, jesse wrote: >> >> during installation: >> >> make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' >> make[1]: Nothing to be done for `all'. >> make[1]: Leaving directory `/tmp/freeswitch/libs/esl/php' >> make -C php install >> PHP Warning: ?PHP Startup: ESL: Unable to initialize module >> Module compiled with module API=20090626, debug=0, thread-safety=0 >> PHP ? ?compiled with module API=20060613, debug=0, thread-safety=0 >> These options need to match >> ?in Unknown on line 0 >> make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' >> cp ESL.so /usr/lib/php5/20060613 >> cp ESL.php /usr/share/php >> >> Then when I open the PHP script from browser, I always get >> >> Unable to initialize module Module compiled with module API=20090626, >> debug=0, thread-safety=0 PHP compiled with module API=20060613, >> debug=0, thread-safety=0 These options need to match in >> /usr/share/php/ESL.php on line 23 >> >> any one is good at PHP can fix this issue? I guess the root cause is >> that I have two PHP installed and my system is totally messed. ?I >> tried to remove the new php installation, but still the same result. >> >> >> >> On Tue, Aug 24, 2010 at 4:56 PM, Nyamul Hassan wrote: >> > I got my phpmod installed and working great! ?Did you follow: >> > http://wiki.freeswitch.org/wiki/Esl#Installation? >> > Please mention at which steps you're seeing an error. >> > Regards >> > HASSAN >> > >> > >> > On Wed, Aug 25, 2010 at 05:42, jesse wrote: >> >> >> >> the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : >> >> >> >> At some point before July 8, 2009, mod_php was removed from the trunk. >> >> There don't appear to be any concrete plans to reintroduce mod_php, so >> >> if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket or >> >> the PHP ESL library. >> >> >> >> >> >> On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale >> >> wrote: >> >> > We never have had PHP in our code distribution. >> >> > >> >> > >> >> > On Mon, Aug 23, 2010 at 4:39 PM, jesse wrote: >> >> >> The current FS version doesn't include source code for PHP. >> >> >> What are the steps to install it on top of my current version? >> >> >> ?where >> >> >> to get the source? how to build and install? >> >> >> >> >> >> Iif you guys want to keep the new system slim by get rid of PHP, >> >> >> please at least keep a doc about how to add it in case of need. >> >> >> >> >> >> thanks! >> >> >> >> >> >> -jesse >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From chat2jesse at gmail.com Wed Aug 25 00:10:25 2010 From: chat2jesse at gmail.com (jesse) Date: Wed, 25 Aug 2010 00:10:25 -0700 Subject: [Freeswitch-users] Lua script help Message-ID: I have a bridge.lua script: phone1 = argv[1]; phone2 = argv[2]; dialstring1 = "sofia/gateway/xyz.com/" .. phone1; dialstring2 = "sofia/gateway/xyz.com7" .. phone2; session1 = freeswitch.Session(dialstring1); session2 = freeswitch.Session(dialstring2, session1); freeswitch.bridge(session1, session2); freeswitch at xyz.com> luarun bridge.lua 16502222222 16503333333 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] +OK freeswitch at xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer sofia/external/16502222222! 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 sofia/external/16503333333 Update Callee ID to "16503333333" <16503333333> 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer sofia/external/16503333333! 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not ready 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session 25 (sofia/external/16503333333) Ended 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/16503333333 [CS_DESTROY] 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session 24 (sofia/external/16502222222) Ended 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/16502222222 [CS_DESTROY] As you can see the call gets dropped immediately after couple rings. however, it works well if i do like this: originate sofia/gateway/xyz.com/16502222222 &bridge(sofia/gateway/xyz.com/16503333333) What is the reason Lua script will fail? any difference between the two approaches? thanks! jesse From math.parent at gmail.com Wed Aug 25 01:13:00 2010 From: math.parent at gmail.com (Mathieu Parent) Date: Wed, 25 Aug 2010 10:13:00 +0200 Subject: [Freeswitch-users] New Bounty (again): Improved Mod_voicemail Emulation In-Reply-To: References: <39C8C9F7-1CB1-4A73-92A6-5F7E029A6C5F@gmail.com> Message-ID: On Tue, Aug 24, 2010 at 10:37 PM, Anthony Minessale wrote: > Interops with SIP over messaging for MWI and message count. And mod_skinny ;) Mathieu Parent From asilva at wirelessmundi.com Wed Aug 25 01:26:20 2010 From: asilva at wirelessmundi.com (Antonio) Date: Wed, 25 Aug 2010 10:26:20 +0200 Subject: [Freeswitch-users] ivr menu matching and execution problem In-Reply-To: References: <1282664502.13028.34.camel@marces.tc.commsmundi.com> Message-ID: <1282724780.6596.4.camel@marces.tc.commsmundi.com> Michael thanks for the reply. I try the same digit, "9", to see what was happen. You are right, in a normal and conventional menu, i will never had two actions for the same single key. Actually i found the problem when i add rules that overlap, when i had something like, "105" => "action 1", "1\d+" => "action 2", so i need that the rules to be execute in the right prio. The behavior that i would like to expect is the same like for the dialplan, tow different states, matching and executing, for the ivr you have only one state, that is matching and executing a the same time. For the IVR we could have more problems, in my case the action transfer is done twice, because of the duplicate match. Thanks, Ant?nio On Tue, 2010-08-24 at 10:47 -0700, Michael Collins wrote: > Sounds to me like a possible solution would be adding a "break" option > for "entry" so that the user could decide whether or not to break > after the first match. However, I wonder if that's even necessary. > > My question to you is why would you have two different actions for > keypress "9" and yet not want the second action ever to be executed? > If the menu only ever executed the first occurrence of a given digit > then what would be the value of having another occurrence of that same > digit? In your example, under what conditions would pressing "9" > execute "transfer 8 XML ..." ? > > There might be a less intrusive solution we could suggest if we knew > more about why you have multiple different actions for a single > keypress. > > -MC > > > On Tue, Aug 24, 2010 at 8:41 AM, Antonio > wrote: > During the configuration of an ivr menu and test i just > realized that if > you have multiple actions with the same "digits" it executes > the both > actions. > > The execution of the multiple actions is normal? > > The normal behavior should executed only the action that is > match first, > reading the priority of the entries in the menu. > I look at the code, and in the function, > switch_ivr_menu_execute, looks > like the loop that searches for the match/action should need a > break > after a success match, to avoid the continues search for other > matches. > > Thanks, > Ant?nio > > > /* configuration */ > > greet-long="/home/system/telephony/sounds/buraka.mp3" > exit-sound="/home/system/telephony/sounds/prompt/en/conf-invalid.gsm" > timeout ="2000" > inter-digit-timeout="2000" > max-failures="10" > digit-len="4" > > > param="transfer $1 > XML local"/> > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Un cordial saludo / Best regards, _________________________ Ant?nio Silva E-mail:asilva at wirelessmundi.com From david.ponzone at ipeva.fr Wed Aug 25 01:30:09 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 10:30:09 +0200 Subject: [Freeswitch-users] php esl compilation PBs In-Reply-To: <20100825041152.0c5a17c4@anubis.defcon1> References: <20100825041152.0c5a17c4@anubis.defcon1> Message-ID: <9AD8C995-BDDA-4D65-ACCB-AF6559CB425D@ipeva.fr> Jean-Yves, you may describe exactly all the steps you are taking, the command you use to compile ESL, and how you compiled FreeSWITCH before. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 04:11, Jean-Yves F. Barbier a ?crit : > Debian sid > ========== > > Hi list, > > I'm trying to figure out a way to compile ESL for PHP on my system. > > I was obliged to copy ../src/include/* into .../esl/php and modify > them a bit, but I still have an error: > > g++ -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/ > php5/TSRM > -I/usr/include/php5/Zend -I/usr/include/php5/ext > -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE - > D_FILE_OFFSET_BITS=64 > -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > > esl_wrap.cpp:2583: attention : deprecated conversion from string > constant > to ?char*? > > g++ esl_wrap.o -lcrypt -lcrypt -lonig -ldb -lqdbm -lrt -lm > -ldl -lnsl -lcrypt -lcrypt -lpthread -o ESL.so -L. > > /usr/bin/ld: cannot find -lonig collect2: ld a retourn? 1 code > d'?tat > d'ex?cution make: *** [ESL.so] > > Error 1 > > Apparently it is a missing library (onig), but which one? > > JY > -- > On a clear disk you can seek forever. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/a967a150/attachment-0001.html From helmut.kuper at ewetel.de Wed Aug 25 01:29:47 2010 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 25 Aug 2010 10:29:47 +0200 Subject: [Freeswitch-users] Predefined hangup cause possible for originate? In-Reply-To: <8D2E7E31-342B-41D4-954A-3A97D4AF772A@avgs.ca> References: <4C73CE8B.6060805@ewetel.de> <8D2E7E31-342B-41D4-954A-3A97D4AF772A@avgs.ca> Message-ID: <4C74D47B.6050700@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Mathieu, hm this is not clear to me. After originate timeout I would like to send a SIP cancel with this header: Reason: SIP;cause=200;text="Call completed elsewhere" so that the phone doesn't record the missed call in its local call list. I'm not sure if this is done with executing hangup + cause after originate timeout. PS: Your answer closed with the word "Gr?sse", so do you talk german over there in canada? Am 25.08.2010 00:21, schrieb Mathieu Rene: > Hi, > > There is already a cause for timeout. You can decide to hangup yourself after that with the cause you want. > > Gr?sse, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMdNR74tZeNddg3dwRAvnbAKCH5/z62QdN7lcwVIammu4FbKT1hQCggvXz HK0kUbPoX/nMMclrR6xHs0U= =LrZE -----END PGP SIGNATURE----- From david.ponzone at ipeva.fr Wed Aug 25 01:33:43 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 10:33:43 +0200 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: References: <1282691200.4420.23.camel@anthony-desktop> Message-ID: AFAIK, most (all ?) providers wont accept the transfer. They are not allowed to change the rate of the call from the caller perspective after the call was initiated. Imagine you transfer the call to an Iridium number... David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 07:44, Ghulam Mustafa a ?crit : > Hi, > > i am wondering what will happen if you send a SIP REFER (transfer) > after validating leg-a without actually answering the call; will it > still cost you 2 * 1.44 > > :/ > > -mustafa > > On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi > wrote: > I am currently looking at Vitelity. They have 1.44 cents per minute. > They do charge for incoming/outgoing both. > > I am not literally dialing out to a phone. I want to dial a mobile > number or any other US number for that matter. > > Here is what I intent to do: > 1. A US phone dials to Vitelity number -- comes to my FS box > 2. I validate few things > 3. Dial another US number and connect received call to that one > > So as you said I will be charged 2 * 1.44 (since I can't terminate > the arrived call after validation). > > Is there any other way to the sequence I have specified above? > > Are there providers similar to Vitelity but cheaper (with relatively > same features)? > > Thank you. > > On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove > wrote: > On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: > > Hello, > > > > > > It would be a great help if someone can guide me. > > > > > > 1. I would like to first receive a call, perform certain > validations. > > (Able to do this via mod_managed application that handles call from > > dialplan). > > > > That should not be a problem, I don't know your requirements so can't > provide a full answer. > > > > > 2. Now, I would like to dial out to a PSTN number so that received > > call is connected to this new outbound number. > > > > This can be done and is called hairpinning. > > > > > How can this be done? Do I use Originate from within my .NET > > (mod_managed) code? > > > > Yes, you would be bridging the two legs like any normal call. > Instead of > going to an endpoint you're going back out over the PSTN. > > > > > Do I get charged for both incoming and outbound call until the > entire > > session ends? Is there a way to receive call, validate and then sort > > of transfer and then terminate the received call so I do not get > > charged for both? > > That would depend on your provider but most likely yes. As for > terminating one end after validation that is not going to happen. A > leg > terminates to you on an agreed fee schedule. No carrier that I know of > supports that. Now if you kept everything SIP... you could do a > transfer > after the validation. > > > > Anthony C. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/c44b83c5/attachment.html From david.ponzone at ipeva.fr Wed Aug 25 01:37:38 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 10:37:38 +0200 Subject: [Freeswitch-users] Lua script help In-Reply-To: References: Message-ID: <8C671BAE-612E-4A85-B404-833BDA61A2A3@ipeva.fr> Jesse, it's unlikely you will get a quicker answer by sending the same a second time after 7 hours. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 09:10, jesse a ?crit : > I have a bridge.lua script: > > phone1 = argv[1]; > phone2 = argv[2]; > > dialstring1 = "sofia/gateway/xyz.com/" .. phone1; > dialstring2 = "sofia/gateway/xyz.com7" .. phone2; > > session1 = freeswitch.Session(dialstring1); > session2 = freeswitch.Session(dialstring2, session1); > freeswitch.bridge(session1, session2); > > > > freeswitch at xyz.com> luarun bridge.lua 16502222222 16503333333 > 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel > sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] > > +OK > > freeswitch at xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 > sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> > 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer > sofia/external/16502222222! > 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel > sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] > 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 > sofia/external/16503333333 Update Callee ID to "16503333333" > <16503333333> > 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer > sofia/external/16503333333! > 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not > ready > 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup > sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] > 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup > sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session > 25 (sofia/external/16503333333) Ended > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/16503333333 [CS_DESTROY] > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session > 24 (sofia/external/16502222222) Ended > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/16502222222 [CS_DESTROY] > > As you can see the call gets dropped immediately after couple rings. > > however, it works well if i do like this: > > originate sofia/gateway/xyz.com/16502222222 > &bridge(sofia/gateway/xyz.com/16503333333) > > What is the reason Lua script will fail? any difference between the > two approaches? > > thanks! > > jesse > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/0e76fbc0/attachment-0001.html From ken at ukgb.net Wed Aug 25 02:46:39 2010 From: ken at ukgb.net (Ken Gillett) Date: Wed, 25 Aug 2010 10:46:39 +0100 Subject: [Freeswitch-users] Account selection In-Reply-To: <4C73D6AD.8000802@communicatefreely.net> References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> <4C73D6AD.8000802@communicatefreely.net> Message-ID: On 24 Aug 2010, at 15:26, Tim St. Pierre wrote: >> >> This can be very important when each SIP account represents a different company/business. Although one person is dealing with all those businesses, when an outgoing call is made it is imperative that the correct SIP account is used to make that call so that the recipient is correctly informed who is making the call. >> > > So really, your issue is with presented identity in terms of caller ID name and number then? Yes, that's about it. > Can you set outgoing caller ID name nad number on the 6 provider accounts? Only the name. Each account has the outgoing number set by the provider (not unreasonable). > If you can, you may have some better options. > > On our platform, we use the dialplan to route all calls to the most appropriate provider, based on > the number that was dialed, and what the rates are for each carrier in a given area (least cost > routing), although reliability in certain areas is also factored. > > Each extension registers with a single registration, and has it's own internal caller ID name and > number (the user's name and extension). > > We use the Aastra phones, and built a little XML app that lets the user pick from a list of possible > caller ID name and number combinations. This tool updates the database value that will be used for > effective_caller_id name and number. With this setup, one user (one SIP registration) can have an > unlimited number "businesses". For incoming calls, we prefix the caller ID name with a short string > that identifies the incoming number or "business". Sometimes, a combination of registrations and > the selector tool is best. If you don't have XML browsers on the phone, you could just as easily do > this with an IVR tool, a web page, or with prefixes. Whatever is easiest. You can have more than > one option. This is all helpful info, thanks. > If your upstream providers can give you DID numbers, you can have more than one business on the same > provider account, which is a lot easier to manage (one gateway entry, but lots of "lines"). I still don't fully grasp DID with VOIP. Some years ago I had a company with a call centre that used a PBX over ISDN30 and I did a lot of the configuration myself so I am familiar with most PBX concepts. But not all of it is directly applicable to VOIP which itself can be handled in many different ways. So what I am trying to do is apply my previous PBX knowledge to what I now know about SIP and VOIP in general and in particular how I can best make use of FreeSwitch to do what I want. So where does DID fit into the VOIP world? Having separate SIP accounts, one for each 'business', each with its own number seems to provide all that DID did (I had to say that:-), but I know my (and other) provider offers 'aliases' which can be used for DID, but I don't fully grasp how this is different from simply different accounts. Maybe in SIP terms it isn't. But if someone cares to enlighten me on this issue, I'm all ears. > This is getting into serious PBX stuff though, and I get the impression you don't really want a PBX. > Or do you? Oh yes indeed. But I am still in a learning process and asking questions is how I learn. My own current requirements are relatively simple, but I need to understand the greater picture and be able to make it work for more complex scenarios. First step though is to configure it for my own use. Ken G i l l e t t _/_/_/_/_/_/_/_/ From dujinfang at gmail.com Wed Aug 25 03:15:00 2010 From: dujinfang at gmail.com (Seven Du) Date: Wed, 25 Aug 2010 18:15:00 +0800 Subject: [Freeswitch-users] Lua script issue. In-Reply-To: References: Message-ID: On Wed, Aug 25, 2010 at 7:56 AM, jesse wrote: > I have a bridge.lua script: > > phone1 = argv[1]; > phone2 = argv[2]; > > dialstring1 = "sofia/gateway/xyz.com/" .. phone1; > dialstring2 = "sofia/gateway/xyz.com7" .. phone2; > > session1 = freeswitch.Session(dialstring1); > session2 = freeswitch.Session(dialstring2, session1); According to http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.bridge , you may try session2 = freeswitch.Session(dialstring2); > freeswitch.bridge(session1, session2); > > > > freeswitch at xyz.com> luarun bridge.lua 16502222222 16503333333 > 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel > sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] > > +OK > > freeswitch at xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 > sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> > 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer > sofia/external/16502222222! > 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel > sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] > 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 > sofia/external/16503333333 Update Callee ID to "16503333333" > <16503333333> > 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer > sofia/external/16503333333! > 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not ready > 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup > sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] > 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup > sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session > 25 (sofia/external/16503333333) Ended > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/16503333333 [CS_DESTROY] > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session > 24 (sofia/external/16502222222) Ended > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/16502222222 [CS_DESTROY] > > As you can see the call gets dropped immediately after couple rings. > however, it works well if i do like this: > > originate sofia/gateway/xyz.com/16502222222 > &bridge(sofia/gateway/xyz.com/16503333333) > > What is the reason Lua script will fail? ?any difference between the > two approaches? > > thanks! > > jesse > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From david.ponzone at ipeva.fr Wed Aug 25 03:47:29 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 12:47:29 +0200 Subject: [Freeswitch-users] Account selection In-Reply-To: References: <91288C84-DB29-45D7-B9A8-0EF08907D0E5@ukgb.net> <755FE4AE-E775-43CE-8D35-3F59D5A19632@ukgb.net> <4C73D6AD.8000802@communicatefreely.net> Message-ID: Ken, Well basically, in the TDM world, you usually order one or several lines. One analog line can't support DIDs (except in the US where they have/ used to have a such product). Nowadays, as you know, ISDN is the right choice when you need DIDs. Physically, a DID on ISDN is just that a different number (DNIS) is sent in the called number field of the Q931 packet, as 4 digits or the whole number, depending on the specific ISDN protocol used and on the telco. In the SIP world, there is only one protocol: SIP. Even if sometimes you feel that there are different type of SIP accounts, they are actually all the same. You can have a SIP account allowing 1 call with one number (generally, they don't allow to send a custom outbound caller-id). But a SIP account can also be used to provide a SIP trunk allowing 200 calls and 2000 DIDs (on those, you can obviously send a custom outbound caller-id, but it can still be restricted to be one of your 2000 DIDs, or not restricted. It depends on the kind of contract you have with the carrier, wholesale vs business). Account, trunk, what's the difference ? Technically, there is none, but let's say a trunk is generally a SIP account (but it can also be a raw SIP trunk without authentication) with several DIDs. If you only have one DID on the SIP account, it's not really meaningful to call that a trunk. On a SIP trunk, a DID is just sent as the dialed number, generally in the INVITE To and/or Request-Line fields. It's pretty much the same than ISDN. The main differences are: -in SIP, a DID is far more decorrelated from the physical media (you can have a SIP account without any DID, so you won't be reachable) -in SIP, a DID could be anything, not just a phone number, but because most of the traffic comes from the PSTN and regular phones, we mainly use digits at the moment. So in SIP, when you have 6 SIP accounts, each other with a different DID, you may replace that with one SIP account (SIP trunk) with the 6 DIDs routed on it by the carrier. I think it's easier to manage. Then in FreeSWITCH, as you won't be able to distinguish by the SIP account receiving the call (you can bind a SIP account to a specific context/dialplan in FS), you'll have to distinguish by the called- number in a common dialplan, which is very easy. I hope I did not confuse you more :) David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 11:46, Ken Gillett a ?crit : > > On 24 Aug 2010, at 15:26, Tim St. Pierre wrote: > >>> >>> This can be very important when each SIP account represents a >>> different company/business. Although one person is dealing with >>> all those businesses, when an outgoing call is made it is >>> imperative that the correct SIP account is used to make that call >>> so that the recipient is correctly informed who is making the call. >>> >> >> So really, your issue is with presented identity in terms of caller >> ID name and number then? > > Yes, that's about it. > >> Can you set outgoing caller ID name nad number on the 6 provider >> accounts? > > Only the name. Each account has the outgoing number set by the > provider (not unreasonable). > >> If you can, you may have some better options. >> >> On our platform, we use the dialplan to route all calls to the most >> appropriate provider, based on >> the number that was dialed, and what the rates are for each carrier >> in a given area (least cost >> routing), although reliability in certain areas is also factored. >> >> Each extension registers with a single registration, and has it's >> own internal caller ID name and >> number (the user's name and extension). >> >> We use the Aastra phones, and built a little XML app that lets the >> user pick from a list of possible >> caller ID name and number combinations. This tool updates the >> database value that will be used for >> effective_caller_id name and number. With this setup, one user >> (one SIP registration) can have an >> unlimited number "businesses". For incoming calls, we prefix the >> caller ID name with a short string >> that identifies the incoming number or "business". Sometimes, a >> combination of registrations and >> the selector tool is best. If you don't have XML browsers on the >> phone, you could just as easily do >> this with an IVR tool, a web page, or with prefixes. Whatever is >> easiest. You can have more than >> one option. > > This is all helpful info, thanks. > >> If your upstream providers can give you DID numbers, you can have >> more than one business on the same >> provider account, which is a lot easier to manage (one gateway >> entry, but lots of "lines"). > > I still don't fully grasp DID with VOIP. Some years ago I had a > company with a call centre that used a PBX over ISDN30 and I did a > lot of the configuration myself so I am familiar with most PBX > concepts. But not all of it is directly applicable to VOIP which > itself can be handled in many different ways. So what I am trying to > do is apply my previous PBX knowledge to what I now know about SIP > and VOIP in general and in particular how I can best make use of > FreeSwitch to do what I want. > > So where does DID fit into the VOIP world? Having separate SIP > accounts, one for each 'business', each with its own number seems to > provide all that DID did (I had to say that:-), but I know my (and > other) provider offers 'aliases' which can be used for DID, but I > don't fully grasp how this is different from simply different > accounts. Maybe in SIP terms it isn't. But if someone cares to > enlighten me on this issue, I'm all ears. > >> This is getting into serious PBX stuff though, and I get the >> impression you don't really want a PBX. >> Or do you? > > > Oh yes indeed. But I am still in a learning process and asking > questions is how I learn. My own current requirements are relatively > simple, but I need to understand the greater picture and be able to > make it work for more complex scenarios. First step though is to > configure it for my own use. > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/fd0c6b5d/attachment-0001.html From steveayre at gmail.com Wed Aug 25 04:55:10 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 25 Aug 2010 12:55:10 +0100 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: "Brian, we are not bridging calls. FreeSWITCH (Leg A) as the caller can not be in a 'not answered' state. Also if I see 200 then it must be come from Leg B (the callee) which means that other party has answered the call." You mean you're doing an originate from ESL? Then there is no 'Leg A', only the 'Leg B' - saying otherwise confuses matters. "And this happened more than 40.000 times between 8 hours" Then you should have no problem providing a sip trace. The log does not contain enough detail to see the full information. For instance the 200 OK could be in a different SIP dialog, or there may be something unusual in the packet that makes FS interpret it incorrectly. If the developers ask for a sip trace saying it's neccesary to debug the problem, don't argue, just provide it. "It is also clear for me, that FS is *over-challenged* to set the correct hangup cause in bulk calls over 200 cps" The code executed will be the same, so hangup causes shouldn't be affected by being at a higher call rate. People have had FS running at >200cps without issue. The only reason I can think of that this might be an issue is if your server is getting overloaded causing a race condition between the hangup timer and the 200 OK being processed - how high is your cpu usage and average load when NO_ANSWER occurs? "We must to re-call all numbers again to give you a SIP trace but this is to expensive for us :-)" Appreciate this, so I would suggest that you either only make calls until you see a call with the issue. Since you say you saw 40000 in 8 hours it should be easy to reproduce, so that would be far cheaper than calling all the numbers; or you can wait until you need to legitimately call them again so there's no additional cost to you. -Steve On 24 August 2010 14:14, Durmu? Ali ?zt?rk wrote: > Brian, we are not bridging calls. FreeSWITCH (Leg A) as the caller can not > be in a 'not answered' state. Also if I see 200 then it must be come from > Leg B (the callee) which means that other party has answered the call. Also, > both legs was active but FS had reported us the opposite. And this happened > more than 40.000 times between 8 hours. > > I would like to know what you are expecting from the sip trace, because the > log is detailed enough to be verified that FS has catched the 200 SIP - but > it seems to be ignored (maybe hangup timer started or something else) > therefor the session is closed with the wrong state. It is also clear for > me, that FS is *over-challenged* to set the correct hangup cause in bulk > calls over 200 cps. > > @Milena > We must to re-call all numbers again to give you a SIP trace but this is to > expensive for us :-) > > >> >> ---------- Weitergeleitete Nachricht ---------- >> From: Milena >> >> To: FreeSWITCH Users Help >> Date: Mon, 23 Aug 2010 10:17:37 -0500 >> >> Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> answered session >> type "version" on the CLI >> >> If the "sip trace wasnt active" then activate it and pastebin it, that's >> the only way you can get help with this, you know the developers are experts >> not soothsayers :) >> >> >> -Mile >> >> 2010/8/23 Durmu? Ali ?zt?rk >> >> I have setup freeswitch by using git and followed the instructions on the >>> wiki page. How can I retrieve the exact git version? >>> >> >> >> ---------- Weitergeleitete Nachricht ---------- >> From: Brian West >> To: FreeSWITCH Users Help >> Date: Mon, 23 Aug 2010 10:18:30 -0500 >> >> Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> answered session >> Again without a SIP trace I can't say... You have to remember that >> FreeSWITCH is a B2BUA and a call thru it results in TWO legs. If you were >> to ANSWER the first but then bridge to the second then it didn't answer you >> might still get a NO_ANSWER even thou you see the 200ok for the one leg of >> the call. >> >> /b >> >> >> On Aug 23, 2010, at 10:08 AM, Durmu? Ali ?zt?rk wrote: >> >> > I have setup freeswitch by using git and followed the instructions on >> the wiki page. How can I retrieve the exact git version? >> > >> > There is no override of the hangup cause in our code. The algorithm is >> very simple; we call the originate function with the playback action and use >> a wav file as argument. Hangup is done automatically by FreeSWITCH, after >> eof of the wav file is reached. Or the other part hangs up before the file >> ends. >> > >> > We dont traced the communication but in other hand, the sip codes are >> traced out into the log file (183-->180-->200=call established). This show >> to me, that there is no problem with the SIP transactions. Please assume, >> that SIP messages are OK. >> > >> > What else could be happen? >> >> >> >> >> >> ---------- Weitergeleitete Nachricht ---------- >> From: Sergey Okhapkin >> >> To: FreeSWITCH Users Help >> Date: Mon, 23 Aug 2010 11:22:37 -0400 >> >> Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> answered session >> You're running concurrent calls, hangup cause NO_ANSWER and log line with >> 200 >> OK belong to different calls. >> >> On Monday 23 August 2010, Durmu? Ali ?zt?rk wrote: >> > I have setup freeswitch by using git and followed the instructions on >> the >> > wiki page. How can I retrieve the exact git version? >> > >> > There is no override of the hangup cause in our code. The algorithm is >> very >> > simple; we call the originate function with the playback action and use >> a >> > wav file as argument. Hangup is done automatically by FreeSWITCH, after >> eof >> > of the wav file is reached. Or the other part hangs up before the file >> > ends. >> > >> > We dont traced the communication but in other hand, the sip codes are >> > traced out into the log file (183-->180-->200=call established). This >> show >> > to me, that there is no problem with the SIP transactions. Please >> assume, >> > that SIP messages are OK. >> > >> > What else could be happen? >> > >> > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> > string 0 = [sip_from_uri=sip:xxxx at xxxxx] >> > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> > string 1 = [ignore_early_media=true] >> > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> > string 2 = [sip_cid_type=none] >> > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> > string 3 = [originate_timeout=40] >> > 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel >> > sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8- >> > 227e99f3f0ef] >> > 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 >> > (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT >> > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send >> signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 >> > sofia/internal/9055599XXXXX set >> UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 >> > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_INIT >> > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 >> > (sofia/internal/9055599XXXXX) State INIT >> > 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 >> > sofia/internal/9055599XXXXX SOFIA INIT >> > 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 >> > (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING >> > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send >> signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 >> > (sofia/internal/9055599XXXXX) State INIT going to sleep >> > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 >> > (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 >> > (sofia/internal/9055599XXXXX) State ROUTING >> > 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 >> > sofia/internal/9055599XXXXX SOFIA ROUTING >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 >> > (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send >> signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 >> > (sofia/internal/9055599XXXXX) State ROUTING going to sleep >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA >> > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 >> > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA >> > 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 >> > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep >> > 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel >> > sofia/internal/9055599XXXXX entering state [calling][0] >> > 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 >> sofia/internal/9055599XXXXX >> > Update Callee ID to "9055599XXXXX" <9055599XXXXX> >> > 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel >> > sofia/internal/9055599XXXXX entering state [proceeding][183] >> > 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready >> > sofia/internal/9055599XXXXX! >> > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel >> > sofia/internal/9055599XXXXX entering state [proceeding][183] >> > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: >> > 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec >> > sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples >> > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send >> > payload to 101 >> > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP >> > [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 >> > port 18620 codec: 8 ms: 20 >> > 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer >> [soft] >> > 160 bytes per 20ms >> > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send >> > payload to 101 >> > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf >> receive >> > payload to 101 >> > 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer >> > sofia/internal/9055599XXXXX! >> > 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 >> > (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY >> > 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel >> > sofia/internal/9055599XXXXX skipping state [proceeding][180] >> > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel >> > sofia/internal/9055599XXXXX entering state [completing][200] >> > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 >> > (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP >> > 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup >> > sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal >> > sofia/internal/9055599XXXXX [KILL] >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 >> > (sofia/internal/9055599XXXXX) State HANGUP >> > 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel >> > sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER >> > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send >> signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to >> > sofia/internal/9055599XXXXX >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 >> > sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 >> > (sofia/internal/9055599XXXXX) State HANGUP going to sleep >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 >> > (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send >> signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 >> > (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING >> > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 >> > (sofia/internal/9055599XXXXX) State REPORTING >> > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 >> > sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER >> > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 >> > (sofia/internal/9055599XXXXX) State REPORTING going to sleep >> > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 >> > (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY >> > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send >> signal >> > sofia/internal/9055599XXXXX [BREAK] >> > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session >> 18770 >> > (sofia/internal/9055599XXXXX) Locked, Waiting on external entities >> > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session >> > 18770 (sofia/internal/9055599XXXXX) Ended >> > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close >> > Channel sofia/internal/9055599XXXXX [CS_DESTROY] >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 >> > (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 >> > (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 >> > (sofia/internal/9055599XXXXX) State DESTROY >> > 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 >> > sofia/internal/9055599XXXXX SOFIA DESTROY >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 >> > sofia/internal/9055599XXXXX Standard DESTROY >> > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 >> > (sofia/internal/9055599XXXXX) State DESTROY going to sleep >> > >> > >> > >> > >> > ---------- Weitergeleitete Nachricht ---------- >> > >> > > From: Brian West >> > > To: FreeSWITCH Users Help >> > > Date: Mon, 23 Aug 2010 09:29:32 -0500 >> > > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> > > answered session >> > > Without that we can't help. You also need to make sure you're on the >> > > very latest code. The one thing I'm sure we get right are the hangup >> > > causes unless you're doing something to override them. >> > > >> > > /b >> > > >> > > On Aug 23, 2010, at 9:24 AM, Durmu? Ali ?zt?rk wrote: >> > > > I'm afraid, sip trace wasnt active. >> > > >> > > ---------- Weitergeleitete Nachricht ---------- >> > > From: Steven Ayre >> > > To: FreeSWITCH Users Help >> > > Date: Mon, 23 Aug 2010 11:53:33 +0100 >> > > Subject: Re: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while >> > > answered session >> > > Do you have a sip trace for those calls? >> > > >> > > >> > > On 23 August 2010 11:36, Durmu? Ali ?zt?rk >> wrote: >> > > Hello, >> > > >> > > we had some trouble while executing a bulk call process with >> originating >> > > a parallel call of 200. Because in many cases, FreeSWITCH has notified >> > > hangups (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong >> > > hangup-cause. Instead of notifying the successful state, we got the >> > > NO_ANSWER hangup cause. BUSY and NO_ANSWER states are candidates for >> > > retries, therefor many numbers are called/payed twice. See log below. >> > > >> > > Some other questions: >> > > >> > > B) Is the originate_timeout value an overall timer or a timer for the >> > > ringing (starts if SIP code 180 incomes?) stage. >> > > >> > > C) We are originating each number in a separate thread and listen to >> the >> > > channel events for updating the call result. Should we change this >> > > implementation or is this a good scenario/standard way. Related to the >> > > call result, if it is busy or not answered, the call is retried after >> 30 >> > > min. What are the recommends on this side to be ensured, the correct >> > > hangup case be got and the number is not called twice.. >> > > >> > > D) What do I have to bear in mind for bulk calls with parallel calls >> over >> > > 200. >> > > >> > > Thanks for your answer in advance. >> > > >> > > >> > > Ali >> > >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/d96e163c/attachment-0001.html From steveayre at gmail.com Wed Aug 25 04:59:04 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 25 Aug 2010 12:59:04 +0100 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: Your log: 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel sofia/internal/9055599XXXXX entering state [completing][200] Latest git: sofia.c line 4318: switch_channel_t *channel = NULL; The line number doesn't match with the latest version. You should upgrade and see whether the issue still occurs with the latest version. -Steve On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: > Hello, > > we had some trouble while executing a bulk call process with originating a > parallel call of 200. Because in many cases, FreeSWITCH has notified hangups > (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong hangup-cause. Instead > of notifying the successful state, we got the NO_ANSWER hangup cause. BUSY > and NO_ANSWER states are candidates for retries, therefor many numbers are > called/payed twice. See log below. > > Some other questions: > > B) Is the originate_timeout value an overall timer or a timer for the > ringing (starts if SIP code 180 incomes?) stage. > > C) We are originating each number in a separate thread and listen to the > channel events for updating the call result. Should we change this > implementation or is this a good scenario/standard way. Related to the call > result, if it is busy or not answered, the call is retried after 30 min. > What are the recommends on this side to be ensured, the correct hangup case > be got and the number is not called twice.. > > D) What do I have to bear in mind for bulk calls with parallel calls over > 200. > > Thanks for your answer in advance. > > Ali > > > An extraction of the log (regard to the first issue): > > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 0 = [sip_from_uri=sip:xxxx at xxxxx] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 1 = [ignore_early_media=true] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 2 = [sip_cid_type=none] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 3 = [originate_timeout=40] > 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel > sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8-227e99f3f0ef] > 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 > (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 > sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_INIT > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9055599XXXXX) State INIT > 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 > sofia/internal/9055599XXXXX SOFIA INIT > 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 > (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9055599XXXXX) State INIT going to sleep > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 > (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9055599XXXXX) State ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 > sofia/internal/9055599XXXXX SOFIA ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9055599XXXXX) State ROUTING going to sleep > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA > 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep > 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [calling][0] > 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX > Update Callee ID to "9055599XXXXX" <9055599XXXXX> > 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [proceeding][183] > 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready > sofia/internal/9055599XXXXX! > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [proceeding][183] > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: > 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec > sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send > payload to 101 > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP > [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 port > 18620 codec: 8 ms: 20 > 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer [soft] > 160 bytes per 20ms > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send > payload to 101 > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive > payload to 101 > 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer > sofia/internal/9055599XXXXX! > 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 > (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY > 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel > sofia/internal/9055599XXXXX skipping state [proceeding][180] > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [completing][200] > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 > (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP > 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup > sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal > sofia/internal/9055599XXXXX [KILL] > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9055599XXXXX) State HANGUP > 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to > sofia/internal/9055599XXXXX > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9055599XXXXX) State HANGUP going to sleep > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/9055599XXXXX) State REPORTING > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/9055599XXXXX) State REPORTING going to sleep > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session 18770 > (sofia/internal/9055599XXXXX) Locked, Waiting on external entities > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session > 18770 (sofia/internal/9055599XXXXX) Ended > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/internal/9055599XXXXX [CS_DESTROY] > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 > (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 > (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/9055599XXXXX) State DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 > sofia/internal/9055599XXXXX SOFIA DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/9055599XXXXX Standard DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/9055599XXXXX) State DESTROY going to sleep > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/55c592ae/attachment.html From steveayre at gmail.com Wed Aug 25 05:02:34 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 25 Aug 2010 13:02:34 +0100 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: How are you retrieving the log that you posted? It looks like some information about the call is missing. This line: 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: Would normally be followed by a dump of the SDP. If some of the log is missing, then we don't know what else might be missing. -Steve On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: > Hello, > > we had some trouble while executing a bulk call process with originating a > parallel call of 200. Because in many cases, FreeSWITCH has notified hangups > (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong hangup-cause. Instead > of notifying the successful state, we got the NO_ANSWER hangup cause. BUSY > and NO_ANSWER states are candidates for retries, therefor many numbers are > called/payed twice. See log below. > > Some other questions: > > B) Is the originate_timeout value an overall timer or a timer for the > ringing (starts if SIP code 180 incomes?) stage. > > C) We are originating each number in a separate thread and listen to the > channel events for updating the call result. Should we change this > implementation or is this a good scenario/standard way. Related to the call > result, if it is busy or not answered, the call is retried after 30 min. > What are the recommends on this side to be ensured, the correct hangup case > be got and the number is not called twice.. > > D) What do I have to bear in mind for bulk calls with parallel calls over > 200. > > Thanks for your answer in advance. > > Ali > > > An extraction of the log (regard to the first issue): > > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 0 = [sip_from_uri=sip:xxxx at xxxxx] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 1 = [ignore_early_media=true] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 2 = [sip_cid_type=none] > 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable > string 3 = [originate_timeout=40] > 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel > sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8-227e99f3f0ef] > 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 > (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 > sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_INIT > 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9055599XXXXX) State INIT > 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 > sofia/internal/9055599XXXXX SOFIA INIT > 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 > (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/9055599XXXXX) State INIT going to sleep > 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 > (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9055599XXXXX) State ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 > sofia/internal/9055599XXXXX SOFIA ROUTING > 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/9055599XXXXX) State ROUTING going to sleep > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA > 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA > 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 > (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep > 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [calling][0] > 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX > Update Callee ID to "9055599XXXXX" <9055599XXXXX> > 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [proceeding][183] > 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready > sofia/internal/9055599XXXXX! > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [proceeding][183] > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: > 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec > sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send > payload to 101 > 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP > [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 port > 18620 codec: 8 ms: 20 > 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer [soft] > 160 bytes per 20ms > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send > payload to 101 > 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive > payload to 101 > 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer > sofia/internal/9055599XXXXX! > 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 > (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY > 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel > sofia/internal/9055599XXXXX skipping state [proceeding][180] > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel > sofia/internal/9055599XXXXX entering state [completing][200] > 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 > (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP > 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup > sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] > 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal > sofia/internal/9055599XXXXX [KILL] > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9055599XXXXX) State HANGUP > 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER > 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to > sofia/internal/9055599XXXXX > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 > (sofia/internal/9055599XXXXX) State HANGUP going to sleep > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/9055599XXXXX) State REPORTING > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER > 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 > (sofia/internal/9055599XXXXX) State REPORTING going to sleep > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal > sofia/internal/9055599XXXXX [BREAK] > 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session 18770 > (sofia/internal/9055599XXXXX) Locked, Waiting on external entities > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session > 18770 (sofia/internal/9055599XXXXX) Ended > 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/internal/9055599XXXXX [CS_DESTROY] > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 > (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 > (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/9055599XXXXX) State DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 > sofia/internal/9055599XXXXX SOFIA DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/9055599XXXXX Standard DESTROY > 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/9055599XXXXX) State DESTROY going to sleep > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/1854fa12/attachment-0001.html From steveayre at gmail.com Wed Aug 25 05:16:34 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 25 Aug 2010 13:16:34 +0100 Subject: [Freeswitch-users] Wrong hangup cause (NO_ANSWER) while answered session In-Reply-To: References: Message-ID: It would also help to see your dialstring / originate command. In answer to your other questions: B) originate_timeout is until it gets answered from the time the invite is sent. C) Can't really comment since I don't use ESL in that way, but it's entirely up to you really. D) Can your hardware handle it. There is also some optimisation that can be done for high call loads. See http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations -Steve On 25 August 2010 13:02, Steven Ayre wrote: > How are you retrieving the log that you posted? It looks like some > information about the call is missing. This line: > > 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: > Would normally be followed by a dump of the SDP. > > If some of the log is missing, then we don't know what else might be > missing. > > -Steve > > > On 23 August 2010 11:36, Durmu? Ali ?zt?rk wrote: > >> Hello, >> >> we had some trouble while executing a bulk call process with originating a >> parallel call of 200. Because in many cases, FreeSWITCH has notified hangups >> (SWITCH_EVENT_CHANNEL_HANGUP_COMPLETE) with the wrong hangup-cause. Instead >> of notifying the successful state, we got the NO_ANSWER hangup cause. BUSY >> and NO_ANSWER states are candidates for retries, therefor many numbers are >> called/payed twice. See log below. >> >> Some other questions: >> >> B) Is the originate_timeout value an overall timer or a timer for the >> ringing (starts if SIP code 180 incomes?) stage. >> >> C) We are originating each number in a separate thread and listen to the >> channel events for updating the call result. Should we change this >> implementation or is this a good scenario/standard way. Related to the call >> result, if it is busy or not answered, the call is retried after 30 min. >> What are the recommends on this side to be ensured, the correct hangup case >> be got and the number is not called twice.. >> >> D) What do I have to bear in mind for bulk calls with parallel calls over >> 200. >> >> Thanks for your answer in advance. >> >> Ali >> >> >> An extraction of the log (regard to the first issue): >> >> 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> string 0 = [sip_from_uri=sip:xxxx at xxxxx] >> 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> string 1 = [ignore_early_media=true] >> 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> string 2 = [sip_cid_type=none] >> 2010-08-17 22:40:51.087402 [DEBUG] switch_ivr_originate.c:1979 variable >> string 3 = [originate_timeout=40] >> 2010-08-17 22:41:09.904378 [NOTICE] switch_channel.c:779 New Channel >> sofia/internal/9055599XXXXX [9e4b396e-3d99-445f-a2e8-227e99f3f0ef] >> 2010-08-17 22:41:12.641362 [DEBUG] mod_sofia.c:3892 >> (sofia/internal/9055599XXXXX) State Change CS_NEW -> CS_INIT >> 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:12.641362 [DEBUG] switch_core_session.c:454 >> sofia/internal/9055599XXXXX set UUID=9dc8a739-f439-4815-981a-347b7a90a1a8 >> 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_INIT >> 2010-08-17 22:41:12.642441 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/9055599XXXXX) State INIT >> 2010-08-17 22:41:12.642441 [DEBUG] mod_sofia.c:83 >> sofia/internal/9055599XXXXX SOFIA INIT >> 2010-08-17 22:41:12.643399 [DEBUG] mod_sofia.c:119 >> (sofia/internal/9055599XXXXX) State Change CS_INIT -> CS_ROUTING >> 2010-08-17 22:41:12.643399 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:338 >> (sofia/internal/9055599XXXXX) State INIT going to sleep >> 2010-08-17 22:41:12.643399 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_ROUTING >> 2010-08-17 22:41:12.644456 [DEBUG] switch_channel.c:1512 >> (sofia/internal/9055599XXXXX) Callstate Change DOWN -> RINGING >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/9055599XXXXX) State ROUTING >> 2010-08-17 22:41:12.644456 [DEBUG] mod_sofia.c:142 >> sofia/internal/9055599XXXXX SOFIA ROUTING >> 2010-08-17 22:41:12.644456 [DEBUG] switch_ivr_originate.c:66 >> (sofia/internal/9055599XXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:341 >> (sofia/internal/9055599XXXXX) State ROUTING going to sleep >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_CONSUME_MEDIA >> 2010-08-17 22:41:12.644456 [DEBUG] switch_core_state_machine.c:360 >> (sofia/internal/9055599XXXXX) State CONSUME_MEDIA >> 2010-08-17 22:41:12.645373 [DEBUG] switch_core_state_machine.c:360 >> (sofia/internal/9055599XXXXX) State CONSUME_MEDIA going to sleep >> 2010-08-17 22:41:42.806367 [DEBUG] sofia.c:4318 Channel >> sofia/internal/9055599XXXXX entering state [calling][0] >> 2010-08-17 22:41:42.890442 [INFO] sofia.c:662 sofia/internal/9055599XXXXX >> Update Callee ID to "9055599XXXXX" <9055599XXXXX> >> 2010-08-17 22:41:42.894355 [DEBUG] sofia.c:4318 Channel >> sofia/internal/9055599XXXXX entering state [proceeding][183] >> 2010-08-17 22:41:42.894355 [NOTICE] sofia.c:4390 Ring-Ready >> sofia/internal/9055599XXXXX! >> 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4318 Channel >> sofia/internal/9055599XXXXX entering state [proceeding][183] >> 2010-08-17 22:41:42.895440 [DEBUG] sofia.c:4329 Remote SDP: >> 2010-08-17 22:41:42.895440 [DEBUG] sofia_glue.c:2444 Set Codec >> sofia/internal/9055599XXXXX PCMA/8000 20 ms 160 samples >> 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:3937 Set 2833 dtmf send >> payload to 101 >> 2010-08-17 22:41:42.896357 [DEBUG] sofia_glue.c:2684 AUDIO RTP >> [sofia/internal/9055599XXXXX] 10.100.224.10 port 31416 -> 10.100.199.19 port >> 18620 codec: 8 ms: 20 >> 2010-08-17 22:41:42.896357 [DEBUG] switch_rtp.c:1413 Starting timer [soft] >> 160 bytes per 20ms >> 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2894 Set 2833 dtmf send >> payload to 101 >> 2010-08-17 22:41:42.897431 [DEBUG] sofia_glue.c:2899 Set 2833 dtmf receive >> payload to 101 >> 2010-08-17 22:41:42.897431 [NOTICE] sofia_glue.c:3292 Pre-Answer >> sofia/internal/9055599XXXXX! >> 2010-08-17 22:41:42.897431 [DEBUG] switch_channel.c:2397 >> (sofia/internal/9055599XXXXX) Callstate Change RINGING -> EARLY >> 2010-08-17 22:41:42.916461 [DEBUG] sofia.c:4313 Channel >> sofia/internal/9055599XXXXX skipping state [proceeding][180] >> 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4318 Channel >> sofia/internal/9055599XXXXX entering state [completing][200] >> 2010-08-17 22:41:43.434412 [DEBUG] sofia.c:4326 Duplicate SDP >> 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2309 >> (sofia/internal/9055599XXXXX) Callstate Change EARLY -> HANGUP >> 2010-08-17 22:41:52.000488 [NOTICE] switch_ivr_originate.c:3282 Hangup >> sofia/internal/9055599XXXXX [CS_CONSUME_MEDIA] [NO_ANSWER] >> 2010-08-17 22:41:52.000488 [DEBUG] switch_channel.c:2325 Send signal >> sofia/internal/9055599XXXXX [KILL] >> 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_HANGUP >> 2010-08-17 22:41:52.000488 [DEBUG] switch_core_state_machine.c:535 >> (sofia/internal/9055599XXXXX) State HANGUP >> 2010-08-17 22:41:52.000488 [DEBUG] mod_sofia.c:453 Channel >> sofia/internal/9055599XXXXX hanging up, cause: NO_ANSWER >> 2010-08-17 22:41:52.000488 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:55.204396 [DEBUG] mod_sofia.c:506 Sending CANCEL to >> sofia/internal/9055599XXXXX >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/9055599XXXXX Standard HANGUP, cause: NO_ANSWER >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:535 >> (sofia/internal/9055599XXXXX) State HANGUP going to sleep >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/9055599XXXXX) State Change CS_HANGUP -> CS_REPORTING >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:55.204396 [DEBUG] switch_core_state_machine.c:314 >> (sofia/internal/9055599XXXXX) Running State Change CS_REPORTING >> 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 >> (sofia/internal/9055599XXXXX) State REPORTING >> 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:53 >> sofia/internal/9055599XXXXX Standard REPORTING, cause: NO_ANSWER >> 2010-08-17 22:41:55.205401 [DEBUG] switch_core_state_machine.c:595 >> (sofia/internal/9055599XXXXX) State REPORTING going to sleep >> 2010-08-17 22:41:55.206428 [DEBUG] switch_core_state_machine.c:327 >> (sofia/internal/9055599XXXXX) State Change CS_REPORTING -> CS_DESTROY >> 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1039 Send signal >> sofia/internal/9055599XXXXX [BREAK] >> 2010-08-17 22:41:55.206428 [DEBUG] switch_core_session.c:1202 Session >> 18770 (sofia/internal/9055599XXXXX) Locked, Waiting on external entities >> 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1220 Session >> 18770 (sofia/internal/9055599XXXXX) Ended >> 2010-08-17 22:41:55.206428 [NOTICE] switch_core_session.c:1222 Close >> Channel sofia/internal/9055599XXXXX [CS_DESTROY] >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:427 >> (sofia/internal/9055599XXXXX) Callstate Change HANGUP -> DOWN >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:430 >> (sofia/internal/9055599XXXXX) Running State Change CS_DESTROY >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/9055599XXXXX) State DESTROY >> 2010-08-17 22:41:55.207463 [DEBUG] mod_sofia.c:358 >> sofia/internal/9055599XXXXX SOFIA DESTROY >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:60 >> sofia/internal/9055599XXXXX Standard DESTROY >> 2010-08-17 22:41:55.207463 [DEBUG] switch_core_state_machine.c:440 >> (sofia/internal/9055599XXXXX) State DESTROY going to sleep >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/0f19e908/attachment.html From 12ukwn at gmail.com Wed Aug 25 05:32:05 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 25 Aug 2010 14:32:05 +0200 Subject: [Freeswitch-users] php esl compilation PBs In-Reply-To: <9AD8C995-BDDA-4D65-ACCB-AF6559CB425D@ipeva.fr> References: <20100825041152.0c5a17c4@anubis.defcon1> <9AD8C995-BDDA-4D65-ACCB-AF6559CB425D@ipeva.fr> Message-ID: <20100825143205.325d1555@anubis.defcon1> Le Wed, 25 Aug 2010 10:30:09 +0200, David Ponzone a ?crit : Ok: I got the source from the GIT, enabled & disabled 2-3 modules, launched a modified configure: --enable-external-libs \ --with-ogg-libraries=/usr/lib \ --with-ogg-includes=/usr/lib/ogg \ --with-vorbis-libraries=/usr/lib \ --with-vorbis-includes=/usr/include \ --with-flac \ --enable-optimization \ --enable-kernel-linux \ --enable-utf8 \ --enable-unicode-properties \ --enable-pcregrep-libz \ --enable-pcregrep-libbz2 \ --enable-pcretest-libreadline \ --enable-nonportable-atomics \ --enable-threadsafe \ --enable-accross-thread-connections \ --enable-threads \ --enable-thread \ --enable-other-child \ --with-devrandom=/dev/urandom \ --with-pgsql=/usr/include/postgresql \ --enable-nonblocking \ --enable-crypto-auth \ --with-ssl=/usr/include/openssl \ --with-zlib=/usr/include \ --with-libidn=/usr/include \ --enable-cplus \ --with-editline and: make all; make install. Then I go to libs/esl/php, copy ../src/include/* into it, modify some references ("xxx.h" instead of to avoid path errors) and make. > you may describe exactly all the steps you are taking, the command you > use to compile ESL, and how you compiled FreeSWITCH before. -- "If you don't want your dog to have bad breath, do what I do: Pour a little Lavoris in the toilet." -- Jay Leno From 12ukwn at gmail.com Wed Aug 25 06:06:25 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 25 Aug 2010 15:06:25 +0200 Subject: [Freeswitch-users] php esl compilation PBs In-Reply-To: <9AD8C995-BDDA-4D65-ACCB-AF6559CB425D@ipeva.fr> References: <20100825041152.0c5a17c4@anubis.defcon1> <9AD8C995-BDDA-4D65-ACCB-AF6559CB425D@ipeva.fr> Message-ID: <20100825150625.1b48d8cb@anubis.defcon1> Le Wed, 25 Aug 2010 10:30:09 +0200, David Ponzone a ?crit : OOPS: Forgot ./bootstrap.sh before ./configure! -- Every time I lose weight, it finds me again! From dujinfang at gmail.com Wed Aug 25 06:38:02 2010 From: dujinfang at gmail.com (Seven Du) Date: Wed, 25 Aug 2010 21:38:02 +0800 Subject: [Freeswitch-users] eavesdrop question Message-ID: Hi, Playing with eavesdrop, it's very nice. And have two questions: 1) A talk to B. C eavesdrop the chanel. C can control whom he want to whisper to by DTMF. Is it possible to use an API/ESL to control ? Would like to make a patch if makes sense. uuid_send_dtmf can send to the remote party, would it make sense to implement a uuid_fake_dtmf to generate dtmf instead of from the remote party? 2) I tested another scenario with: A bridged to B. C eavesdrop the channel. And then D eavesdrop too. C can input DTMF 1/2/3, C worked as a normal eavesdrop. D has exactly the same behavior with C. (e.g. when C talk to A, D also can talk to A, when C talk to B, D can also talk to B). D cannot control anything with DTMF. Is it possible for the eavesdrop also respond to D's DTMF ? Ideally C and D has not effect to each other. Thanks. -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From spambox at haruhiism.net Tue Aug 24 23:27:59 2010 From: spambox at haruhiism.net (Kamigishi Rei) Date: Wed, 25 Aug 2010 10:27:59 +0400 Subject: [Freeswitch-users] Deployment questions: Freeswitch or Freeswitch with a SIP proxy server Message-ID: <4C74B7EF.7070102@haruhiism.net> Hello, I need to set up a relatively small (up to, probably, 5000 users at maximum) SIP service; however I'm not sure if what I need can be implemented by using Freeswitch, so I decided to consult this mailing list. First, the essential part: local calls within the SIP realm; easy to get implemented and basically works out of the box with some scripts to export data from our billing database and to convert that data into XML. The second task, implementing a PSTN gateway, looks pretty straightforward, too. However, there are a few questions about something else: external SIP servers/operators, and DNS/ports setup. Basically, I can receive inbound calls from other SIP operators to my public extensions out of the box, without setting up any gateways and such; that behaviour is welcome. However, the invite domain presented gets changed to FreeSwitch's IPv4 address before showing up on the softphone. Is overriding it with "{sip_invite_domain=${sip_from_host}}sofia/internal/$1%$${domain}" in a bridge application considered bad practice? Placing outbound calls to other SIP operators: this "sort of" works if I uncomment in dialplan/default.xml. However, this is a very weird and - to me - extremely crude solution as, first, it's a fallback rule (thus we can't call a remote extension that also exists in our system - f.ex. calling 1001 at remote will result in the call being directed to 1001 at local), and second, ${domain_exists(${sip_req_host})} is a really lousy check and doesn't account for DNS aliases, IP addresses, etc. thus resulting in possible fallback to a direct call instead of a dialplan route if the sip_invite_domain wasn't set properly and we got a call back. It's even more confusing if we use DNS NAPTR and SRV records. Currently, I have the following in my dialplan: Does it seem acceptable or is that bad practice as well? Is placing outbound calls to external SIP URIs considered bad practice overall when using Freeswitch? From what I've gathered @ the irc channel, "Freeswitch is not a proxy", so should I instead use something like OpenSIPS for such tasks? And about the DNS settings: It is established that the internal profile serves authenticated clients, and the external profile is for untrusted connections (gateways). I presume that to receive calls to my public extensions, I need the remote SIP operators/servers/clients to be able to find my "public" SIP port, which is 5080 in the default configuration. Therefore, I have the IN SRV records for "mydomain.org" pointing at port 5080, external profile. Or is it acceptable to receive external calls to public extensions to the internal profile? (Clarification needed here; is "external" profile supposed to be "for external gateways", or "for any untrusted connections, including calls from non-gateways to our public extensions". Also, is accepting calls from a non-authenticated SIP peer to the public extensions, but on the internal profile's port, not recommended?) However because of that I have to use a DNS alias (or just the IP address) in the clients' (f.ex. softphones) configuration; f.ex. "sip.mydomain.org". This results in clients using "sip.mydomain.org" as their authentication realm (which is understandable), and direct - non-proxied - outbound calls appear as "ext at sip.mydomain.org" on the remote side. Which leads to calls originating from the remote side based on their call history arriving on port 5060, internal profile. If I specify my Freeswitch server as the SIP proxy, the aforementioned external_domain_outbound_calls extension takes care of it by replacing sip_invite_domain in the headers. But since "Freeswitch is not a proxy", I guess there should be some other solution instead of having to place calls to external URIs via Freeswitch? Overall, if accepting unauthenticated inbound calls on the internal profile is fine and they still go into public context, I guess it's fine to have the DNS point at the internal profile? (Since gateways can connect to the external profile's port explicitly.) Thanks in advance. -- Kamigishi Rei KREI-RIPE From elihayunfs at gmail.com Tue Aug 24 23:57:02 2010 From: elihayunfs at gmail.com (EliFS) Date: Wed, 25 Aug 2010 09:57:02 +0300 Subject: [Freeswitch-users] Can't find mod_callcenter Message-ID: <1282719422.11822.6.camel@localhost.localdomain> Hi I can see in the wiki that there is a module called mod_callcenter but I cannot find it. I downloaded the latest snapshot and nada. Its not there. Any help? Eli -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/779d53a8/attachment.html From jim at k4gvo.com Wed Aug 25 07:05:06 2010 From: jim at k4gvo.com (Jim) Date: Wed, 25 Aug 2010 10:05:06 -0400 Subject: [Freeswitch-users] Still can't dial gateway from ZAP phone. In-Reply-To: References: <4C730A7F.6030408@k4gvo.com> Message-ID: <4C752312.3040800@k4gvo.com> On 08/24/2010 01:20 PM, Michael Collins wrote: > > EXECUTE sofia/internal/1002 at 192.168.2.51 > > bridge(sofia/gateway/gw4.telasip.com/17705550068 > ) > > Somehow the information in the directory/default/default.xml file > never got included and I'm not sure how to fix it. > > > When dialing from an FXS port you are not an "authenticated user" so > those variables from user directory don't get populated. You have two > choices as I see it: > #1 - Manually set the ${default_gateway} variable by inserting this > line before the bridge: > > > #2 - Use the set_user app > (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user) to > make the call act like it was from an auth'd user: > > > Give it a whirl and let me know how it goes. > -MC > Number 1 seems to work fine, my new file looks like: Thanks, Jim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/f1f143a3/attachment.html From brian at freeswitch.org Wed Aug 25 07:08:01 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Aug 2010 09:08:01 -0500 Subject: [Freeswitch-users] Still can't dial gateway from ZAP phone. In-Reply-To: <4C752312.3040800@k4gvo.com> References: <4C730A7F.6030408@k4gvo.com> <4C752312.3040800@k4gvo.com> Message-ID: <5B2DD198-788E-4F82-AFBC-CF67109E5330@freeswitch.org> This isn't valid there. /b On Aug 25, 2010, at 9:05 AM, Jim wrote: > /> From fdelawarde at wirelessmundi.com Wed Aug 25 07:25:07 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 25 Aug 2010 16:25:07 +0200 Subject: [Freeswitch-users] MWI doubt Message-ID: <1282746307.2970.135.camel@luna.tc.commsmundi.com> Hello, I wish to have a single domain across multiple profiles. For registrations and calls it works perfectly. Having a shared database for all profiles, using sofia_contact with "/@" gives back the full dialstring with the profile where the phone is currently registered. Now I was trying to do some MWI accross those profiles, and it appears that some phones have the NOTIFY message every time they register, and some other do not receive. Any reason? Also, when a phone is configured to SUBSCRIBE for MWI, it doesn't receive NOTIFYs when a new message is on the voicemail. The only thing that appears wrong is this console message: 2010-08-25 16:09:57.578144 [ERR] sofia_presence.c:316 Cannot find profile - Does the profile need to have an alias to a particular domain for presence/MWI subscriptions to work? - Is it possible somehow to have multiple profiles aliased to a same domain? If not, what is the recommended way to share presence or MWI across profiles? Thank you, Fran?ois. From david.ponzone at ipeva.fr Wed Aug 25 07:35:01 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 16:35:01 +0200 Subject: [Freeswitch-users] php esl compilation PBs In-Reply-To: <20100825143205.325d1555@anubis.defcon1> References: <20100825041152.0c5a17c4@anubis.defcon1> <9AD8C995-BDDA-4D65-ACCB-AF6559CB425D@ipeva.fr> <20100825143205.325d1555@anubis.defcon1> Message-ID: Isn't --with-php required too ? David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 14:32, Jean-Yves F. Barbier a ?crit : > Le Wed, 25 Aug 2010 10:30:09 +0200, > David Ponzone a ?crit : > > Ok: > I got the source from the GIT, > enabled & disabled 2-3 modules, > launched a modified configure: > > --enable-external-libs \ > --with-ogg-libraries=/usr/lib \ > --with-ogg-includes=/usr/lib/ogg \ > --with-vorbis-libraries=/usr/lib \ > --with-vorbis-includes=/usr/include \ > --with-flac \ > --enable-optimization \ > --enable-kernel-linux \ > --enable-utf8 \ > --enable-unicode-properties \ > --enable-pcregrep-libz \ > --enable-pcregrep-libbz2 \ > --enable-pcretest-libreadline \ > --enable-nonportable-atomics \ > --enable-threadsafe \ > --enable-accross-thread-connections \ > --enable-threads \ > --enable-thread \ > --enable-other-child \ > --with-devrandom=/dev/urandom \ > --with-pgsql=/usr/include/postgresql \ > --enable-nonblocking \ > --enable-crypto-auth \ > --with-ssl=/usr/include/openssl \ > --with-zlib=/usr/include \ > --with-libidn=/usr/include \ > --enable-cplus \ > --with-editline > > and: make all; make install. > > Then I go to libs/esl/php, > copy ../src/include/* into it, > modify some references ("xxx.h" instead of to avoid path > errors) > and make. > >> you may describe exactly all the steps you are taking, the command >> you >> use to compile ESL, and how you compiled FreeSWITCH before. > > -- > "If you don't want your dog to have bad breath, do what I do: Pour > a little > Lavoris in the toilet." > -- Jay Leno > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/1bf54a83/attachment-0001.html From fs-list at communicatefreely.net Wed Aug 25 07:42:43 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 25 Aug 2010 10:42:43 -0400 Subject: [Freeswitch-users] Deployment questions: Freeswitch or Freeswitch with a SIP proxy server In-Reply-To: <4C74B7EF.7070102@haruhiism.net> References: <4C74B7EF.7070102@haruhiism.net> Message-ID: <4C752BE3.1070407@communicatefreely.net> You don't have to use the profiles "as-is" if they don't suit. Here's our solution. Let me know if it might apply to your case: "public" profile: Listens on port 5080 Has it's own dialplan that can deliver a call to extensions, but doesn't allow calls to much else. The dialplan is written with security in mind. The DNS SRV record for our domain points to the public IP address for this profile, port 5080. Anyone dialing EXTEN at our.domain gets sent here, and the call is processed. "trunks" profile Listens on port 5060 (that's what our trunking providers expect). Has it's own dialplan that matches incoming calls to their respective DIDs using lua/mysql and routes the call to the correct destination within our system. An ACL authenticates our trunking peers based on IP address and blocks everyone else. We can also assume that all calls coming in on this profile are billable (they come from the PSTN via a carrier switch). DNS isn't important to us on this profile, since our carriers map straight to the switch IPs, but we could do something like trunks.our.domain for the SRV record "internal" profile. listens on port 5070 (and eventually 5071 when I get TLS working) Is used for all the SIP endpoints, and authenticates them when they register. This profile goes to our internal dialplan, that routes to extensions, features, pstn, etc. We will also setup external domain matching here. All the phones register to pbx.our.domain which has an SRV record as well. We aliased that domain name to the profile, so it's happy to register the phones with that domain name. All the phones use the same domain name to register. If a call comes in on this profile for anything other than pbx.our.domain, we can match this in the dialplan and send a call out on the "public" profile to whomever they want to call. With Freeswitch, you can make as many profiles as you want, and you can configure them any way you want. It did take a bit of figuring to make everything work right with DNS though. We wanted to use SRV records for everything, so that we can load balance / fail over the entire network. The only problem that arises is with some endpoints that insist on using the A record. You could swap the ports on your trunks and public profiles if your peers don't mind sending calls to port 5080. That would let you keep using the default port for public calls. -Tim > And about the DNS settings: > > It is established that the internal profile serves authenticated > clients, and the external profile is for untrusted connections (gateways). > I presume that to receive calls to my public extensions, I need the > remote SIP operators/servers/clients to be able to find my "public" SIP > port, which is 5080 in the default configuration. Therefore, I have the > IN SRV records for "mydomain.org" pointing at port 5080, external > profile. Or is it acceptable to receive external calls to public > extensions to the internal profile? (Clarification needed here; is > "external" profile supposed to be "for external gateways", or "for any > untrusted connections, including calls from non-gateways to our public > extensions". Also, is accepting calls from a non-authenticated SIP peer > to the public extensions, but on the internal profile's port, not > recommended?) > > However because of that I have to use a DNS alias (or just the IP > address) in the clients' (f.ex. softphones) configuration; f.ex. > "sip.mydomain.org". > This results in clients using "sip.mydomain.org" as their authentication > realm (which is understandable), and direct - non-proxied - outbound > calls appear as "ext at sip.mydomain.org" on the remote side. Which leads > to calls originating from the remote side based on their call history > arriving on port 5060, internal profile. From fs-list at communicatefreely.net Wed Aug 25 07:46:39 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 25 Aug 2010 10:46:39 -0400 Subject: [Freeswitch-users] MWI doubt In-Reply-To: <1282746307.2970.135.camel@luna.tc.commsmundi.com> References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> Message-ID: <4C752CCF.7020408@communicatefreely.net> I had this problem too, although I'm using different domains on different profiles. I had to alias each domain I was using to the profile the phones would use. When the phones subscribe to their MWI, freeswitch looks for the domain that is being subscribed to, and it has to exist. I don't know if you can have the same domain aliased to more than one profile. I haven't tried that. Anyone else? Fran?ois Delawarde wrote: > Hello, > > I wish to have a single domain across multiple profiles. For > registrations and calls it works perfectly. Having a shared database for > all profiles, using sofia_contact with "/@" > gives back the full dialstring with the profile where the phone is > currently registered. > > Now I was trying to do some MWI accross those profiles, and it appears > that some phones have the NOTIFY message every time they register, and > some other do not receive. Any reason? > > Also, when a phone is configured to SUBSCRIBE for MWI, it doesn't > receive NOTIFYs when a new message is on the voicemail. The only thing > that appears wrong is this console message: > > 2010-08-25 16:09:57.578144 [ERR] sofia_presence.c:316 Cannot find profile > > - Does the profile need to have an alias to a particular domain for > presence/MWI subscriptions to work? > > - Is it possible somehow to have multiple profiles aliased to a same > domain? If not, what is the recommended way to share presence or MWI > across profiles? > > Thank you, > Fran?ois. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs-list at communicatefreely.net Wed Aug 25 07:49:08 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 25 Aug 2010 10:49:08 -0400 Subject: [Freeswitch-users] dialplan variables in XML cdr Message-ID: <4C752D64.3080506@communicatefreely.net> Hello, Is there a way to have some custom dialplan variables appear in the XML cdr output? We want to log some things about the call, such as what option was chosen, which carrier route, etc. I want to be able to set a value in the dialplan and then figure out what it was in the CDR so we can do reports on it later. The variables I set aren't in the section. Is there any way to enable this, or am I missing something? Thanks! -Tim From brian at freeswitch.org Wed Aug 25 07:51:37 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Aug 2010 09:51:37 -0500 Subject: [Freeswitch-users] dialplan variables in XML cdr In-Reply-To: <4C752D64.3080506@communicatefreely.net> References: <4C752D64.3080506@communicatefreely.net> Message-ID: All variables set on the session appear in the CDR. /b On Aug 25, 2010, at 9:49 AM, Tim St. Pierre wrote: > Hello, > > Is there a way to have some custom dialplan variables appear in the XML cdr output? We want to log > some things about the call, such as what option was chosen, which carrier route, etc. I want to be > able to set a value in the dialplan and then figure out what it was in the CDR so we can do reports > on it later. The variables I set aren't in the section. Is there any way to enable > this, or am I missing something? > > Thanks! > > -Tim > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Aug 25 07:52:04 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Aug 2010 09:52:04 -0500 Subject: [Freeswitch-users] php esl compilation PBs In-Reply-To: References: <20100825041152.0c5a17c4@anubis.defcon1> <9AD8C995-BDDA-4D65-ACCB-AF6559CB425D@ipeva.fr> <20100825143205.325d1555@anubis.defcon1> Message-ID: <8B91BC17-41BF-48DB-9E6A-53C892B3D56A@freeswitch.org> FreeSWITCH has zero knowledge of PHP. /b On Aug 25, 2010, at 9:35 AM, David Ponzone wrote: > Isn't --with-php required too ? > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > From brian at freeswitch.org Wed Aug 25 07:52:27 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Aug 2010 09:52:27 -0500 Subject: [Freeswitch-users] MWI doubt In-Reply-To: <4C752CCF.7020408@communicatefreely.net> References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> <4C752CCF.7020408@communicatefreely.net> Message-ID: <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> Make sure the profile responsible for that domain has an alias for that domain on it. /b On Aug 25, 2010, at 9:46 AM, Tim St. Pierre wrote: > I had this problem too, although I'm using different domains on different profiles. > > I had to alias each domain I was using to the profile the phones would use. When the phones > subscribe to their MWI, freeswitch looks for the domain that is being subscribed to, and it has to > exist. I don't know if you can have the same domain aliased to more than one profile. I haven't > tried that. > > Anyone else? From juanito1982 at gmail.com Wed Aug 25 08:02:48 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Wed, 25 Aug 2010 17:02:48 +0200 Subject: [Freeswitch-users] Outbound codecs preference In-Reply-To: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> References: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> Message-ID: It doesn't worked as I thought. I'd like caller codecs priority has preferente over provider codecs preference: Now, doing inbound late negotiation and adding code you suggest: IP phone offers: G729,PCMA (${ep_codec_string}= G729 at 8000h,PCMA at 8000h) FS codecs prefs: global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM" outbound_codec_prefs=G729,PCMU,PCMA,GSM Provider offers: PCMU,PCMA,G729 Doing above config FS selects PCMA instead G729. Disabling inboud-late-negotiation FS select G729 for A-leg and PCMU for B-leg hanging up the call because there is no g729 licenses installed. It could use G729 not dropping the call... Regards 2010/8/25 David Ponzone > The best way is probably to use inbound-late-negotiation, and in the > dialplan before bridging to add: > data="absolute_codec_string=${ep_codec_string}" /> > > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 24/08/2010 ? 20:00, Juan Antonio Iba?ez Santorum a ?crit : > > I can see when I try to stablish one B-leg call FS gives more precedence to > remote codecs order. This way it is difficult to avoid transcoding in some > situations where it could be possible. Is there any way to get a behaviour > similar inbound-codec-negotiation=greedy for outbounds calls? What would be > the best way to avoid transcoding? May be to use inbound-late-negotiation > and rewrite codecs strings? > > Regards > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/4173d39b/attachment-0001.html From david.ponzone at ipeva.fr Wed Aug 25 08:06:45 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 17:06:45 +0200 Subject: [Freeswitch-users] php esl compilation PBs In-Reply-To: <8B91BC17-41BF-48DB-9E6A-53C892B3D56A@freeswitch.org> References: <20100825041152.0c5a17c4@anubis.defcon1> <9AD8C995-BDDA-4D65-ACCB-AF6559CB425D@ipeva.fr> <20100825143205.325d1555@anubis.defcon1> <8B91BC17-41BF-48DB-9E6A-53C892B3D56A@freeswitch.org> Message-ID: Brian, you're right of course, I think I am confusing with the --with-lua I add to use some months ago. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 16:52, Brian West a ?crit : > FreeSWITCH has zero knowledge of PHP. > > /b > > On Aug 25, 2010, at 9:35 AM, David Ponzone wrote: > >> Isn't --with-php required too ? >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/04f137ca/attachment.html From david.ponzone at ipeva.fr Wed Aug 25 08:21:57 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 17:21:57 +0200 Subject: [Freeswitch-users] Outbound codecs preference In-Reply-To: References: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> Message-ID: <8B3F425E-05E2-40AD-A0C9-E1A3BD323DE0@ipeva.fr> Try: David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 17:02, Juan Antonio Iba?ez Santorum a ?crit : > It doesn't worked as I thought. I'd like caller codecs priority has > preferente over provider codecs preference: > > Now, doing inbound late negotiation and adding code you suggest: > > IP phone offers: > G729,PCMA (${ep_codec_string}= G729 at 8000h,PCMA at 8000h) > > FS codecs prefs: > > global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM" > outbound_codec_prefs=G729,PCMU,PCMA,GSM > > Provider offers: > PCMU,PCMA,G729 > > Doing above config FS selects PCMA instead G729. > Disabling inboud-late-negotiation FS select G729 for A-leg and PCMU > for B-leg hanging up the call because there is no g729 licenses > installed. It could use G729 not dropping the call... > > Regards > > > 2010/8/25 David Ponzone > The best way is probably to use inbound-late-negotiation, and in the > dialplan before bridging to add: > > > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 24/08/2010 ? 20:00, Juan Antonio Iba?ez Santorum a ?crit : > >> I can see when I try to stablish one B-leg call FS gives more >> precedence to remote codecs order. This way it is difficult to >> avoid transcoding in some situations where it could be possible. Is >> there any way to get a behaviour similar inbound-codec- >> negotiation=greedy for outbounds calls? What would be the best way >> to avoid transcoding? May be to use inbound-late-negotiation and >> rewrite codecs strings? >> >> Regards >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/124d8ce7/attachment-0001.html From fdelawarde at wirelessmundi.com Wed Aug 25 08:32:49 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 25 Aug 2010 17:32:49 +0200 Subject: [Freeswitch-users] MWI doubt In-Reply-To: <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> <4C752CCF.7020408@communicatefreely.net> <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> Message-ID: <1282750369.2970.157.camel@luna.tc.commsmundi.com> On Wed, 2010-08-25 at 09:52 -0500, Brian West wrote: > Make sure the profile responsible for that domain has an alias for that domain on it. Ok, but is it possible to have more than one profile responsible for a domain for some type of multi-homed setup? I tried using a single presence db for all profiles with: It works well with the 'sofia_contact' command, using: sofia_contact /@ It will return the dialstring with the profile where the phone was registered (no need to know it beforehand). Thanks, Fran?ois. From william.suffill at gmail.com Wed Aug 25 08:33:47 2010 From: william.suffill at gmail.com (William Suffill) Date: Wed, 25 Aug 2010 11:33:47 -0400 Subject: [Freeswitch-users] Can't find mod_callcenter In-Reply-To: <1282719422.11822.6.camel@localhost.localdomain> References: <1282719422.11822.6.camel@localhost.localdomain> Message-ID: Should be in the git tree tho. On Aug 25, 2010 9:52 AM, "EliFS" wrote: Hi I can see in the wiki that there is a module called mod_callcenter but I cannot find it. I downloaded the latest snapshot and nada. Its not there. Any help? Eli _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/4484ba02/attachment.html From brian at freeswitch.org Wed Aug 25 08:37:31 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Aug 2010 10:37:31 -0500 Subject: [Freeswitch-users] MWI doubt In-Reply-To: <1282750369.2970.157.camel@luna.tc.commsmundi.com> References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> <4C752CCF.7020408@communicatefreely.net> <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> <1282750369.2970.157.camel@luna.tc.commsmundi.com> Message-ID: On Aug 25, 2010, at 10:32 AM, Fran?ois Delawarde wrote: > Ok, but is it possible to have more than one profile responsible for a > domain for some type of multi-homed setup? NO. > I tried using a single presence db for all profiles with: > Sure. But thats more for calling users registered. > It works well with the 'sofia_contact' command, using: > sofia_contact /@ > > It will return the dialstring with the profile where the phone was > registered (no need to know it beforehand). > > Thanks, > Fran?ois. From mnhassan at usa.net Wed Aug 25 08:39:44 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Wed, 25 Aug 2010 21:39:44 +0600 Subject: [Freeswitch-users] Can't find mod_callcenter In-Reply-To: References: <1282719422.11822.6.camel@localhost.localdomain> Message-ID: The fs-devs have moved to git. SVN might not be as updated. Regards HASSAN On Wed, Aug 25, 2010 at 21:33, William Suffill wrote: > Should be in the git tree tho. > > On Aug 25, 2010 9:52 AM, "EliFS" wrote: > > Hi > I can see in the wiki that there is a module called mod_callcenter but I > cannot find it. > I downloaded the latest snapshot and nada. Its not there. > > Any help? > > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/e3f8cc34/attachment.html From juanito1982 at gmail.com Wed Aug 25 08:45:09 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Wed, 25 Aug 2010 17:45:09 +0200 Subject: [Freeswitch-users] Outbound codecs preference In-Reply-To: <8B3F425E-05E2-40AD-A0C9-E1A3BD323DE0@ipeva.fr> References: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> <8B3F425E-05E2-40AD-A0C9-E1A3BD323DE0@ipeva.fr> Message-ID: It applies to inboud leg forcing not outbound. Tested and call fails :( 2010/8/25 David Ponzone > Try: > > > > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 25/08/2010 ? 17:02, Juan Antonio Iba?ez Santorum a ?crit : > > It doesn't worked as I thought. I'd like caller codecs priority has > preferente over provider codecs preference: > > Now, doing inbound late negotiation and adding code you suggest: > > IP phone offers: > G729,PCMA (${ep_codec_string}= G729 at 8000h,PCMA at 8000h) > > FS codecs prefs: > global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM" > outbound_codec_prefs=G729,PCMU,PCMA,GSM > > Provider offers: > PCMU,PCMA,G729 > > Doing above config FS selects PCMA instead G729. > Disabling inboud-late-negotiation FS select G729 for A-leg and PCMU for > B-leg hanging up the call because there is no g729 licenses installed. It > could use G729 not dropping the call... > > Regards > > > 2010/8/25 David Ponzone > >> The best way is probably to use inbound-late-negotiation, and in the >> dialplan before bridging to add: >> > data="absolute_codec_string=${ep_codec_string}" /> >> >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 24/08/2010 ? 20:00, Juan Antonio Iba?ez Santorum a ?crit : >> >> I can see when I try to stablish one B-leg call FS gives more precedence >> to remote codecs order. This way it is difficult to avoid transcoding in >> some situations where it could be possible. Is there any way to get a >> behaviour similar inbound-codec-negotiation=greedy for outbounds calls? What >> would be the best way to avoid transcoding? May be to use >> inbound-late-negotiation and rewrite codecs strings? >> >> Regards >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/1534fa91/attachment-0001.html From 12ukwn at gmail.com Wed Aug 25 08:46:32 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Wed, 25 Aug 2010 17:46:32 +0200 Subject: [Freeswitch-users] Can't find mod_callcenter In-Reply-To: References: <1282719422.11822.6.camel@localhost.localdomain> Message-ID: <20100825174632.03ba71d7@anubis.defcon1> On Wed, 25 Aug 2010 11:33:47 -0400, William Suffill wrote: in latest git: ./src/mod/applications/mod_callcenter/mod_callcenter.c > I can see in the wiki that there is a module called mod_callcenter but I > cannot find it. > I downloaded the latest snapshot and nada. Its not there. -- Fuck, I can't compile the damn thing and I wrote it ! From fdelawarde at wirelessmundi.com Wed Aug 25 08:57:22 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 25 Aug 2010 17:57:22 +0200 Subject: [Freeswitch-users] MWI doubt In-Reply-To: References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> <4C752CCF.7020408@communicatefreely.net> <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> <1282750369.2970.157.camel@luna.tc.commsmundi.com> Message-ID: <1282751842.2970.178.camel@luna.tc.commsmundi.com> On Wed, 2010-08-25 at 10:37 -0500, Brian West wrote: > On Aug 25, 2010, at 10:32 AM, Fran?ois Delawarde wrote: > > > Ok, but is it possible to have more than one profile responsible for > a > > domain for some type of multi-homed setup? > > NO. > > > I tried using a single presence db for all profiles with: > > > > Sure. But thats more for calling users registered. Well I just tested and... IT SEEMS TO WORK (at least with MWI, will test other presence stuff later)!!! I did the following: sofia.conf.xml: ... ... ... And whatever domain and profile the phone connects to, it just works! The only trick is that they all need to have the same dbname, and one of them needs to have an alias to all domains. Also, in all the domain configuration something like: will work even if phones are connected to profile2 or profile1!! That's an awesome feature, and FreeSwitch rocks again! Now I just hope it's an intended feature and not just a bug... By the way sorry, I should have tested this before sending crap on the ML. Brian, as you previously said it was not possible: is this something normal and "supported"? Thanks, Fran?ois From brian at freeswitch.org Wed Aug 25 09:01:03 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Aug 2010 11:01:03 -0500 Subject: [Freeswitch-users] MWI doubt In-Reply-To: <1282751842.2970.178.camel@luna.tc.commsmundi.com> References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> <4C752CCF.7020408@communicatefreely.net> <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> <1282750369.2970.157.camel@luna.tc.commsmundi.com> <1282751842.2970.178.camel@luna.tc.commsmundi.com> Message-ID: <2909F380-BC65-4021-AAF8-7D22E02B2253@freeswitch.org> The issue is you want the generic MWI to figure out which profile to send the notify out... it can't what you want is a feature request. You can NOT have the same domain on multiple profiles not doable. /b On Aug 25, 2010, at 10:57 AM, Fran?ois Delawarde wrote: > hat's an awesome feature, and FreeSwitch rocks again! Now I just hope > it's an intended feature and not just a bug... By the way sorry, I > should have tested this before sending crap on the ML. > > Brian, as you previously said it was not possible: is this something > normal and "supported"? From macedoslm at gmail.com Wed Aug 25 09:30:36 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Wed, 25 Aug 2010 13:30:36 -0300 Subject: [Freeswitch-users] Suppress DTMF In-Reply-To: References: Message-ID: Is there any config that I can do in the dtmf mode? Thanks, -- Samuel Macedo On 23 August 2010 23:08, Samuel Macedo wrote: > Hi, > > I want to suppress the dtmf tone in the conference room. When a user sends > a DTMF in the conference all others users can hear that DTMF tone. > How can I suppress it? > > My = "waste". > > Thanks, > -- > Samuel Macedo > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/9bd64f9c/attachment.html From brian at freeswitch.org Wed Aug 25 09:39:55 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Aug 2010 11:39:55 -0500 Subject: [Freeswitch-users] Suppress DTMF In-Reply-To: References: Message-ID: If they are sending a DTMF inband aka audio you can't. /b On Aug 25, 2010, at 11:30 AM, Samuel Macedo wrote: > Is there any config that I can do in the dtmf mode? > > Thanks, > -- > Samuel Macedo From tculjaga at gmail.com Wed Aug 25 09:46:08 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 25 Aug 2010 18:46:08 +0200 Subject: [Freeswitch-users] long DTMF In-Reply-To: References: <1282650110.2970.52.camel@luna.tc.commsmundi.com> <1282675278265-5458183.post@n2.nabble.com> Message-ID: On Tue, Aug 24, 2010 at 9:05 PM, Tihomir Culjaga wrote: > > > On Tue, Aug 24, 2010 at 8:41 PM, Jeff Lenk wrote: > >> >> Move the playback inside the script >> >> not really feasible, all the logic is already done in DP... Im missing > just this feature. > > what is the bgapi equivalent of: > > > > > anyone who can know this ? how to run a lua script in background ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/527c3c3c/attachment.html From macedoslm at gmail.com Wed Aug 25 09:58:32 2010 From: macedoslm at gmail.com (Samuel Macedo) Date: Wed, 25 Aug 2010 13:58:32 -0300 Subject: [Freeswitch-users] Suppress DTMF In-Reply-To: References: Message-ID: Is there any way to change this?! Thanks -- Samuel Macedo On 25 August 2010 13:39, Brian West wrote: > If they are sending a DTMF inband aka audio you can't. > > /b > > On Aug 25, 2010, at 11:30 AM, Samuel Macedo wrote: > > > Is there any config that I can do in the dtmf mode? > > > > Thanks, > > -- > > Samuel Macedo > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/4d7760d6/attachment.html From fdelawarde at wirelessmundi.com Wed Aug 25 10:03:30 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 25 Aug 2010 19:03:30 +0200 Subject: [Freeswitch-users] MWI doubt In-Reply-To: <2909F380-BC65-4021-AAF8-7D22E02B2253@freeswitch.org> References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> <4C752CCF.7020408@communicatefreely.net> <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> <1282750369.2970.157.camel@luna.tc.commsmundi.com> <1282751842.2970.178.camel@luna.tc.commsmundi.com> <2909F380-BC65-4021-AAF8-7D22E02B2253@freeswitch.org> Message-ID: <1282755810.2970.194.camel@luna.tc.commsmundi.com> On Wed, 2010-08-25 at 11:01 -0500, Brian West wrote: > The issue is you want the generic MWI to figure out which profile to > send the notify out... it can't what you want is a feature request. > You can NOT have the same domain on multiple profiles not doable. Then it's a bug, because MWI works beautifully across profiles with the previous trick (did you try?). I believe it's because FreeSwitch uses the "profile_name" stored in database to do whatever its work is. If database is shared between all profiles, then having a bogus profile with all domains aliased to it is enough for Freeswitch to find the database and once the "select" is done, it just uses the db field. At least that's how sofia_contact does the trick, the profile passed in parameter is not used to build the reply, only the "profile_name" field in "sip_registrations". Best bug ever, please don't repair it! Fran?ois. From brian at freeswitch.org Wed Aug 25 10:08:05 2010 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Aug 2010 12:08:05 -0500 Subject: [Freeswitch-users] MWI doubt In-Reply-To: <1282755810.2970.194.camel@luna.tc.commsmundi.com> References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> <4C752CCF.7020408@communicatefreely.net> <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> <1282750369.2970.157.camel@luna.tc.commsmundi.com> <1282751842.2970.178.camel@luna.tc.commsmundi.com> <2909F380-BC65-4021-AAF8-7D22E02B2253@freeswitch.org> <1282755810.2970.194.camel@luna.tc.commsmundi.com> Message-ID: <40456555-8929-4C6C-AD74-EEB81AA2D08B@freeswitch.org> No the bonding of profiles is fine... and makes sofia_contact behave differently... thats a totally different use case... The bottom line is you can't have the same domain aliased on multiple profiles... thats all I said in my first response. If this works for you then thats fine I'm pretty sure that was our intended behavior for this use. /b On Aug 25, 2010, at 12:03 PM, Fran?ois Delawarde wrote: > Then it's a bug, because MWI works beautifully across profiles with the > previous trick (did you try?). > > I believe it's because FreeSwitch uses the "profile_name" stored in > database to do whatever its work is. If database is shared between all > profiles, then having a bogus profile with all domains aliased to it is > enough for Freeswitch to find the database and once the "select" is > done, it just uses the db field. > > At least that's how sofia_contact does the trick, the profile passed in > parameter is not used to build the reply, only the "profile_name" field > in "sip_registrations". > > Best bug ever, please don't repair it! > > Fran?ois. From david.ponzone at ipeva.fr Wed Aug 25 10:29:57 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Wed, 25 Aug 2010 19:29:57 +0200 Subject: [Freeswitch-users] Outbound codecs preference In-Reply-To: References: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> <8B3F425E-05E2-40AD-A0C9-E1A3BD323DE0@ipeva.fr> Message-ID: <63D9FCFC-282E-4A38-8F1C-0D9BE354EFDA@ipeva.fr> Juan, I see your issue now. Well, first solution is to only send to the provider the codec you want. If you prefer G729 for a phone, it's likely the provider will support that, so offer only G729. If you prefer PCM for another phone, it's likely the provider will support that, so offer only PCM. The strange thing, but I guess it depends, is that your provider sends you back several codecs (is that even acceptable ?). When I sent them a list of codecs, my providers always replies with only one, the one they want, generally it's the first one in the list I offered. It seems your provider is trying to refuse your preference, which is not very nice. I don't think there is any other way than offering them only the codec you want. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 17:45, Juan Antonio Iba?ez Santorum a ?crit : > It applies to inboud leg forcing not outbound. Tested and call > fails :( > > 2010/8/25 David Ponzone > Try: > > > > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou > falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le > d?truire imm?diatement et d'avertir l'exp?diteur. > > > > > Le 25/08/2010 ? 17:02, Juan Antonio Iba?ez Santorum a ?crit : > >> It doesn't worked as I thought. I'd like caller codecs priority has >> preferente over provider codecs preference: >> >> Now, doing inbound late negotiation and adding code you suggest: >> >> IP phone offers: >> G729,PCMA (${ep_codec_string}= G729 at 8000h,PCMA at 8000h) >> >> FS codecs prefs: >> >> global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM" >> outbound_codec_prefs=G729,PCMU,PCMA,GSM >> >> Provider offers: >> PCMU,PCMA,G729 >> >> Doing above config FS selects PCMA instead G729. >> Disabling inboud-late-negotiation FS select G729 for A-leg and PCMU >> for B-leg hanging up the call because there is no g729 licenses >> installed. It could use G729 not dropping the call... >> >> Regards >> >> >> 2010/8/25 David Ponzone >> The best way is probably to use inbound-late-negotiation, and in >> the dialplan before bridging to add: >> >> >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? l'intention exclusive de ses destinataires. Toute >> utilisation ou diffusion non autoris?e est interdite. Tout message >> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute >> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou >> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de >> le d?truire imm?diatement et d'avertir l'exp?diteur. >> >> >> >> >> Le 24/08/2010 ? 20:00, Juan Antonio Iba?ez Santorum a ?crit : >> >>> I can see when I try to stablish one B-leg call FS gives more >>> precedence to remote codecs order. This way it is difficult to >>> avoid transcoding in some situations where it could be possible. >>> Is there any way to get a behaviour similar inbound-codec- >>> negotiation=greedy for outbounds calls? What would be the best way >>> to avoid transcoding? May be to use inbound-late-negotiation and >>> rewrite codecs strings? >>> >>> Regards >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/bb297d38/attachment-0001.html From ktngl at yahoo.co.uk Wed Aug 25 10:47:06 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Wed, 25 Aug 2010 17:47:06 +0000 (GMT) Subject: [Freeswitch-users] long DTMF In-Reply-To: Message-ID: <984085.22821.qm@web29214.mail.ird.yahoo.com> Have you tried using the event socket library http://wiki.freeswitch.org/wiki/Esl --- On Wed, 25/8/10, Tihomir Culjaga wrote: From: Tihomir Culjaga Subject: Re: [Freeswitch-users] long DTMF To: "FreeSWITCH Users Help" Date: Wednesday, 25 August, 2010, 16:46 On Tue, Aug 24, 2010 at 9:05 PM, Tihomir Culjaga wrote: On Tue, Aug 24, 2010 at 8:41 PM, Jeff Lenk wrote: Move the playback inside the script not really feasible, all the logic is already done in DP... Im missing just this feature. what is the bgapi equivalent of: anyone who can know this ? how to run a lua script in background ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/ca454edc/attachment.html From juanito1982 at gmail.com Wed Aug 25 10:58:02 2010 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Wed, 25 Aug 2010 19:58:02 +0200 Subject: [Freeswitch-users] Outbound codecs preference In-Reply-To: <63D9FCFC-282E-4A38-8F1C-0D9BE354EFDA@ipeva.fr> References: <1947DDA5-0A0C-464A-B8BC-EA7A6D40AFDF@ipeva.fr> <8B3F425E-05E2-40AD-A0C9-E1A3BD323DE0@ipeva.fr> <63D9FCFC-282E-4A38-8F1C-0D9BE354EFDA@ipeva.fr> Message-ID: 2010/8/25 David Ponzone > Juan, > > I see your issue now. > > Well, first solution is to only send to the provider the codec you want. > If you prefer G729 for a phone, it's likely the provider will support that, > so offer only G729. > If you prefer PCM for another phone, it's likely the provider will support > that, so offer only PCM. > I think a parameter may be called 'outbound-codec-negotiation' would be a better solution. This way, when you will offer several codecs to the provider and the provider offer several codecs to you, you could set that your codec preferences have higher priority than the providers' as done with inbound calls. > > The strange thing, but I guess it depends, is that your provider sends you > back several codecs (is that even acceptable ?). > When I sent them a list of codecs, my providers always replies with only > one, the one they want, generally it's the first one in the list I offered. > I suppose similar we offer several codecs to our registered ip phones. > It seems your provider is trying to refuse your preference, which is not > very nice. > I don't think there is any other way than offering them only the codec you > want. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 25/08/2010 ? 17:45, Juan Antonio Iba?ez Santorum a ?crit : > > It applies to inboud leg forcing not outbound. Tested and call fails :( > > 2010/8/25 David Ponzone > >> Try: >> >> >> >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 25/08/2010 ? 17:02, Juan Antonio Iba?ez Santorum a ?crit : >> >> It doesn't worked as I thought. I'd like caller codecs priority has >> preferente over provider codecs preference: >> >> Now, doing inbound late negotiation and adding code you suggest: >> >> IP phone offers: >> G729,PCMA (${ep_codec_string}= G729 at 8000h,PCMA at 8000h) >> >> FS codecs prefs: >> global_codec_prefs=G729,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM" >> outbound_codec_prefs=G729,PCMU,PCMA,GSM >> >> Provider offers: >> PCMU,PCMA,G729 >> >> Doing above config FS selects PCMA instead G729. >> Disabling inboud-late-negotiation FS select G729 for A-leg and PCMU for >> B-leg hanging up the call because there is no g729 licenses installed. It >> could use G729 not dropping the call... >> >> Regards >> >> >> 2010/8/25 David Ponzone >> >>> The best way is probably to use inbound-late-negotiation, and in the >>> dialplan before bridging to add: >>> >> data="absolute_codec_string=${ep_codec_string}" /> >>> >>> >>> David Ponzone Direction Technique >>> email: david.ponzone at ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr - www.ipeva-studio.com >>> >>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >>> l'exp?diteur.* >>> * >>> * >>> >>> >>> >>> Le 24/08/2010 ? 20:00, Juan Antonio Iba?ez Santorum a ?crit : >>> >>> I can see when I try to stablish one B-leg call FS gives more precedence >>> to remote codecs order. This way it is difficult to avoid transcoding in >>> some situations where it could be possible. Is there any way to get a >>> behaviour similar inbound-codec-negotiation=greedy for outbounds calls? What >>> would be the best way to avoid transcoding? May be to use >>> inbound-late-negotiation and rewrite codecs strings? >>> >>> Regards >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/3c920b82/attachment-0001.html From chat2jesse at gmail.com Wed Aug 25 11:25:34 2010 From: chat2jesse at gmail.com (jesse) Date: Wed, 25 Aug 2010 11:25:34 -0700 Subject: [Freeswitch-users] Lua script help In-Reply-To: <8C671BAE-612E-4A85-B404-833BDA61A2A3@ipeva.fr> References: <8C671BAE-612E-4A85-B404-833BDA61A2A3@ipeva.fr> Message-ID: It happened to me before that my post didn't appear in the mail list. Not sure whether the same issue was repeating, so I posted it 7 hours later without seeing the 1st post show up. just curious, is the mail list moderated manually? How long should I wait for a new topic post to show up in the mail-list? thanks! -jesse On Wed, Aug 25, 2010 at 1:37 AM, David Ponzone wrote: > Jesse, > it's unlikely you will get a quicker answer by sending the same a second > time after 7 hours. > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 25/08/2010 ? 09:10, jesse a ?crit : > > I have a bridge.lua script: > > phone1 = argv[1]; > phone2 = argv[2]; > > dialstring1 = "sofia/gateway/xyz.com/" .. phone1; > dialstring2 = "sofia/gateway/xyz.com7" .. phone2; > > session1 = freeswitch.Session(dialstring1); > session2 = freeswitch.Session(dialstring2, session1); > freeswitch.bridge(session1, session2); > > > > freeswitch at xyz.com> luarun bridge.lua 16502222222 16503333333 > 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel > sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] > > +OK > > freeswitch at xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 > sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> > 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer > sofia/external/16502222222! > 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel > sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] > 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 > sofia/external/16503333333 Update Callee ID to "16503333333" > <16503333333> > 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer > sofia/external/16503333333! > 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not ready > 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup > sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] > 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup > sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session > 25 (sofia/external/16503333333) Ended > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/16503333333 [CS_DESTROY] > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session > 24 (sofia/external/16502222222) Ended > 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close > Channel sofia/external/16502222222 [CS_DESTROY] > > As you can see the call gets dropped immediately after couple rings. > > however, it works well if i do like this: > > originate sofia/gateway/xyz.com/16502222222 > &bridge(sofia/gateway/xyz.com/16503333333) > > What is the reason Lua script will fail? ?any difference between the > two approaches? > > thanks! > > jesse > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mthakershi at gmail.com Wed Aug 25 11:28:55 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Wed, 25 Aug 2010 13:28:55 -0500 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: References: <1282691200.4420.23.camel@anthony-desktop> Message-ID: Can someone suggest me wholesale and cheaper provider than Vitelity but offering similar service? In/Out bound both USA. Thank you. On Wed, Aug 25, 2010 at 3:33 AM, David Ponzone wrote: > AFAIK, most (all ?) providers wont accept the transfer. > They are not allowed to change the rate of the call from the caller > perspective after the call was initiated. > Imagine you transfer the call to an Iridium number... > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce > message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > destinataire de ce message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur.* > * > * > > > > Le 25/08/2010 ? 07:44, Ghulam Mustafa a ?crit : > > Hi, > > i am wondering what will happen if you send a SIP REFER (transfer) after > validating leg-a without actually answering the call; will it still cost you > 2 * 1.44 > > :/ > > -mustafa > > On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi wrote: > >> I am currently looking at Vitelity. They have 1.44 cents per minute. They >> do charge for incoming/outgoing both. >> >> I am not literally dialing out to a phone. I want to dial a mobile number >> or any other US number for that matter. >> >> Here is what I intent to do: >> 1. A US phone dials to Vitelity number -- comes to my FS box >> 2. I validate few things >> 3. Dial another US number and connect received call to that one >> >> So as you said I will be charged 2 * 1.44 (since I can't terminate the >> arrived call after validation). >> >> Is there any other way to the sequence I have specified above? >> >> Are there providers similar to Vitelity but cheaper (with relatively same >> features)? >> >> Thank you. >> >> On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove wrote: >> >>> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: >>> > Hello, >>> > >>> > >>> > It would be a great help if someone can guide me. >>> > >>> > >>> > 1. I would like to first receive a call, perform certain validations. >>> > (Able to do this via mod_managed application that handles call from >>> > dialplan). >>> > >>> >>> That should not be a problem, I don't know your requirements so can't >>> provide a full answer. >>> >>> > >>> > 2. Now, I would like to dial out to a PSTN number so that received >>> > call is connected to this new outbound number. >>> > >>> >>> This can be done and is called hairpinning. >>> >>> > >>> > How can this be done? Do I use Originate from within my .NET >>> > (mod_managed) code? >>> > >>> >>> Yes, you would be bridging the two legs like any normal call. Instead of >>> going to an endpoint you're going back out over the PSTN. >>> >>> > >>> > Do I get charged for both incoming and outbound call until the entire >>> > session ends? Is there a way to receive call, validate and then sort >>> > of transfer and then terminate the received call so I do not get >>> > charged for both? >>> >>> That would depend on your provider but most likely yes. As for >>> terminating one end after validation that is not going to happen. A leg >>> terminates to you on an agreed fee schedule. No carrier that I know of >>> supports that. Now if you kept everything SIP... you could do a transfer >>> after the validation. >>> >>> >>> >>> Anthony C. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/3a293b0b/attachment.html From chat2jesse at gmail.com Wed Aug 25 11:29:24 2010 From: chat2jesse at gmail.com (jesse) Date: Wed, 25 Aug 2010 11:29:24 -0700 Subject: [Freeswitch-users] Lua script issue. In-Reply-To: References: Message-ID: hmm, I do not think that is the reason. Just tried your suggestion, the same error. -jesse On Wed, Aug 25, 2010 at 3:15 AM, Seven Du wrote: > On Wed, Aug 25, 2010 at 7:56 AM, jesse wrote: >> I have a bridge.lua script: >> >> phone1 = argv[1]; >> phone2 = argv[2]; >> >> dialstring1 = "sofia/gateway/xyz.com/" .. phone1; >> dialstring2 = "sofia/gateway/xyz.com7" .. phone2; >> >> session1 = freeswitch.Session(dialstring1); >> session2 = freeswitch.Session(dialstring2, session1); > > According to http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.bridge > , you may try > session2 = freeswitch.Session(dialstring2); > >> freeswitch.bridge(session1, session2); >> >> >> >> freeswitch at xyz.com> luarun bridge.lua 16502222222 16503333333 >> 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel >> sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] >> >> +OK >> >> freeswitch at xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 >> sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> >> 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer >> sofia/external/16502222222! >> 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel >> sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] >> 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 >> sofia/external/16503333333 Update Callee ID to "16503333333" >> <16503333333> >> 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer >> sofia/external/16503333333! >> 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not ready >> 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup >> sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] >> 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup >> sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] >> 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session >> 25 (sofia/external/16503333333) Ended >> 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close >> Channel sofia/external/16503333333 [CS_DESTROY] >> 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session >> 24 (sofia/external/16502222222) Ended >> 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close >> Channel sofia/external/16502222222 [CS_DESTROY] >> >> As you can see the call gets dropped immediately after couple rings. >> however, it works well if i do like this: >> >> originate sofia/gateway/xyz.com/16502222222 >> &bridge(sofia/gateway/xyz.com/16503333333) >> >> What is the reason Lua script will fail? ?any difference between the >> two approaches? >> >> thanks! >> >> jesse >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Blog: http://www.dujinfang.com > Proj:? http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From neil.burgess at redmatter.com Wed Aug 25 11:34:47 2010 From: neil.burgess at redmatter.com (Neil Burgess) Date: Wed, 25 Aug 2010 19:34:47 +0100 Subject: [Freeswitch-users] Remote-Party-ID manipulation in SIP 200 OK Message-ID: <787302A89ACCE24DA8F56DA101E77C842B3AE95E00@THHS2E12BE1X.hostedservice2.net> Hi, I am bridging a call through FS and have a problem with a Remote-Party-ID header being added to the returned 200 OK SIP message that is sent back on the A-Leg! I need to either re-write or strip it The scenario is that:- 1/ a user dials a number such as 07740777777 2/ we have to rewrite the number in the dial plan to 447740980777777 for routing to the upstream provider on the B leg 3/ the call is then bridged through to our upstream provider who attempts to connect the call successfully, returning a 200 OK. 4/ the a leg then has the Remote-Party-ID field inserted (looks to be done by FS) into the responding 200 OK, looking something like Remote-Party-ID: "447740777777" sip:07740777777 at 172.30.98.4 Unfortunately the Yealink phones we use then re-write the number from the original dialled to the number provided in the Remote-Party-ID (e.g. 447740980777777), which ends up in the phone's history. This now causes redials to fail! Any ideas on whether we can remove this header/ re-write or indeed control the number inserted into the display name component of the Remote Party ID? Many thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/9a14a492/attachment-0001.html From sos at sokhapkin.dyndns.org Wed Aug 25 11:48:59 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 25 Aug 2010 14:48:59 -0400 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: References: Message-ID: <201008251448.59835.sos@sokhapkin.dyndns.org> callwithus? On Wednesday 25 August 2010, Malay Thakershi wrote: > Can someone suggest me wholesale and cheaper provider than Vitelity but > offering similar service? > > In/Out bound both USA. > > Thank you. > > On Wed, Aug 25, 2010 at 3:33 AM, David Ponzone wrote: > > AFAIK, most (all ?) providers wont accept the transfer. > > They are not allowed to change the rate of the call from the caller > > perspective after the call was initiated. > > Imagine you transfer the call to an Iridium number... > > > > David Ponzone Direction Technique > > email: david.ponzone at ipeva.fr > > tel: 01 74 03 18 97 > > gsm: 06 66 98 76 34 > > > > Service Client IPeva > > tel: 0811 46 26 26 > > www.ipeva.fr - www.ipeva-studio.com > > > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > > l'intention exclusive de ses destinataires. Toute utilisation ou > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > susceptible d'alt?ration. **IPeva** d?cline toute responsabilit? au titre > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas > > destinataire de ce message, merci de le d?truire imm?diatement et > > d'avertir l'exp?diteur.* > > * > > * > > > > > > > > Le 25/08/2010 ? 07:44, Ghulam Mustafa a ?crit : > > > > Hi, > > > > i am wondering what will happen if you send a SIP REFER (transfer) after > > validating leg-a without actually answering the call; will it still cost > > you 2 * 1.44 > > > > :/ > > > > -mustafa > > > > On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi wrote: > >> I am currently looking at Vitelity. They have 1.44 cents per minute. > >> They do charge for incoming/outgoing both. > >> > >> I am not literally dialing out to a phone. I want to dial a mobile > >> number or any other US number for that matter. > >> > >> Here is what I intent to do: > >> 1. A US phone dials to Vitelity number -- comes to my FS box > >> 2. I validate few things > >> 3. Dial another US number and connect received call to that one > >> > >> So as you said I will be charged 2 * 1.44 (since I can't terminate the > >> arrived call after validation). > >> > >> Is there any other way to the sequence I have specified above? > >> > >> Are there providers similar to Vitelity but cheaper (with relatively > >> same features)? > >> > >> Thank you. > >> > >> On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove wrote: > >>> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: > >>> > Hello, > >>> > > >>> > > >>> > It would be a great help if someone can guide me. > >>> > > >>> > > >>> > 1. I would like to first receive a call, perform certain validations. > >>> > (Able to do this via mod_managed application that handles call from > >>> > dialplan). > >>> > >>> That should not be a problem, I don't know your requirements so can't > >>> provide a full answer. > >>> > >>> > 2. Now, I would like to dial out to a PSTN number so that received > >>> > call is connected to this new outbound number. > >>> > >>> This can be done and is called hairpinning. > >>> > >>> > How can this be done? Do I use Originate from within my .NET > >>> > (mod_managed) code? > >>> > >>> Yes, you would be bridging the two legs like any normal call. Instead > >>> of going to an endpoint you're going back out over the PSTN. > >>> > >>> > Do I get charged for both incoming and outbound call until the entire > >>> > session ends? Is there a way to receive call, validate and then sort > >>> > of transfer and then terminate the received call so I do not get > >>> > charged for both? > >>> > >>> That would depend on your provider but most likely yes. As for > >>> terminating one end after validation that is not going to happen. A leg > >>> terminates to you on an agreed fee schedule. No carrier that I know of > >>> supports that. Now if you kept everything SIP... you could do a > >>> transfer after the validation. > >>> > >>> > >>> > >>> Anthony C. > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > >>>s http://www.freeswitch.org > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > Ghulam Mustafa > > cell: +92 333.611.7681 > > sip: cyrenity at ekiga.net > > mail: mustafa.pk at gmail.com > > web: cyrenity.wordpress.com > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From michael.scheidell at secnap.com Wed Aug 25 12:04:02 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Wed, 25 Aug 2010 15:04:02 -0400 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: References: <1282691200.4420.23.camel@anthony-desktop> Message-ID: <4C756922.4060206@secnap.com> don't know vitelity, but voip.ms. supports user/password or ip based auth. has built in Vmail, redirect on network outage. us did's are .99/c per month, and calls 1.1c/ month. (20? trunks per did?) On 8/25/10 2:28 PM, Malay Thakershi wrote: > Can someone suggest me wholesale and cheaper provider than Vitelity > but offering similar service? > > In/Out bound both USA. > > Thank you. > > On Wed, Aug 25, 2010 at 3:33 AM, David Ponzone > wrote: > > AFAIK, most (all ?) providers wont accept the transfer. > They are not allowed to change the rate of the call from the > caller perspective after the call was initiated. > Imagine you transfer the call to an Iridium number... > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > /Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline > toute responsabilit? au titre de ce message s'il a ?t? alt?r?, > d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur./ > / > / > > > > Le 25/08/2010 ? 07:44, Ghulam Mustafa a ?crit : > >> Hi, >> >> i am wondering what will happen if you send a SIP REFER >> (transfer) after validating leg-a without actually answering the >> call; will it still cost you 2 * 1.44 >> >> :/ >> >> -mustafa >> >> On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi >> > wrote: >> >> I am currently looking at Vitelity. They have 1.44 cents per >> minute. They do charge for incoming/outgoing both. >> >> I am not literally dialing out to a phone. I want to dial a >> mobile number or any other US number for that matter. >> >> Here is what I intent to do: >> 1. A US phone dials to Vitelity number -- comes to my FS box >> 2. I validate few things >> 3. Dial another US number and connect received call to that one >> >> So as you said I will be charged 2 * 1.44 (since I can't >> terminate the arrived call after validation). >> >> Is there any other way to the sequence I have specified above? >> >> Are there providers similar to Vitelity but cheaper (with >> relatively same features)? >> >> Thank you. >> >> On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove >> > wrote: >> >> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: >> > Hello, >> > >> > >> > It would be a great help if someone can guide me. >> > >> > >> > 1. I would like to first receive a call, perform >> certain validations. >> > (Able to do this via mod_managed application that >> handles call from >> > dialplan). >> > >> >> That should not be a problem, I don't know your >> requirements so can't >> provide a full answer. >> >> > >> > 2. Now, I would like to dial out to a PSTN number so >> that received >> > call is connected to this new outbound number. >> > >> >> This can be done and is called hairpinning. >> >> > >> > How can this be done? Do I use Originate from within my >> .NET >> > (mod_managed) code? >> > >> >> Yes, you would be bridging the two legs like any normal >> call. Instead of >> going to an endpoint you're going back out over the PSTN. >> >> > >> > Do I get charged for both incoming and outbound call >> until the entire >> > session ends? Is there a way to receive call, validate >> and then sort >> > of transfer and then terminate the received call so I >> do not get >> > charged for both? >> >> That would depend on your provider but most likely yes. >> As for >> terminating one end after validation that is not going to >> happen. A leg >> terminates to you on an agreed fee schedule. No carrier >> that I know of >> supports that. Now if you kept everything SIP... you >> could do a transfer >> after the validation. >> >> >> >> Anthony C. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Ghulam Mustafa >> cell: +92 333.611.7681 >> sip: cyrenity at ekiga.net >> mail: mustafa.pk @gmail.com >> web: cyrenity.wordpress.com >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/214bb0c4/attachment-0001.html From mthakershi at gmail.com Wed Aug 25 12:07:18 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Wed, 25 Aug 2010 14:07:18 -0500 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: <201008251448.59835.sos@sokhapkin.dyndns.org> References: <201008251448.59835.sos@sokhapkin.dyndns.org> Message-ID: Provide link to your business. On Wed, Aug 25, 2010 at 1:48 PM, Sergey Okhapkin wrote: > callwithus? > > On Wednesday 25 August 2010, Malay Thakershi wrote: > > Can someone suggest me wholesale and cheaper provider than Vitelity but > > offering similar service? > > > > In/Out bound both USA. > > > > Thank you. > > > > On Wed, Aug 25, 2010 at 3:33 AM, David Ponzone > wrote: > > > AFAIK, most (all ?) providers wont accept the transfer. > > > They are not allowed to change the rate of the call from the caller > > > perspective after the call was initiated. > > > Imagine you transfer the call to an Iridium number... > > > > > > David Ponzone Direction Technique > > > email: david.ponzone at ipeva.fr > > > tel: 01 74 03 18 97 > > > gsm: 06 66 98 76 34 > > > > > > Service Client IPeva > > > tel: 0811 46 26 26 > > > www.ipeva.fr - www.ipeva-studio.com > > > > > > *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > ? > > > l'intention exclusive de ses destinataires. Toute utilisation ou > > > diffusion non autoris?e est interdite. Tout message ?lectronique est > > > susceptible d'alt?ration. **IPeva** d?cline toute responsabilit? au > titre > > > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes > pas > > > destinataire de ce message, merci de le d?truire imm?diatement et > > > d'avertir l'exp?diteur.* > > > * > > > * > > > > > > > > > > > > Le 25/08/2010 ? 07:44, Ghulam Mustafa a ?crit : > > > > > > Hi, > > > > > > i am wondering what will happen if you send a SIP REFER (transfer) > after > > > validating leg-a without actually answering the call; will it still > cost > > > you 2 * 1.44 > > > > > > :/ > > > > > > -mustafa > > > > > > On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi > wrote: > > >> I am currently looking at Vitelity. They have 1.44 cents per minute. > > >> They do charge for incoming/outgoing both. > > >> > > >> I am not literally dialing out to a phone. I want to dial a mobile > > >> number or any other US number for that matter. > > >> > > >> Here is what I intent to do: > > >> 1. A US phone dials to Vitelity number -- comes to my FS box > > >> 2. I validate few things > > >> 3. Dial another US number and connect received call to that one > > >> > > >> So as you said I will be charged 2 * 1.44 (since I can't terminate the > > >> arrived call after validation). > > >> > > >> Is there any other way to the sequence I have specified above? > > >> > > >> Are there providers similar to Vitelity but cheaper (with relatively > > >> same features)? > > >> > > >> Thank you. > > >> > > >> On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove > wrote: > > >>> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: > > >>> > Hello, > > >>> > > > >>> > > > >>> > It would be a great help if someone can guide me. > > >>> > > > >>> > > > >>> > 1. I would like to first receive a call, perform certain > validations. > > >>> > (Able to do this via mod_managed application that handles call from > > >>> > dialplan). > > >>> > > >>> That should not be a problem, I don't know your requirements so can't > > >>> provide a full answer. > > >>> > > >>> > 2. Now, I would like to dial out to a PSTN number so that received > > >>> > call is connected to this new outbound number. > > >>> > > >>> This can be done and is called hairpinning. > > >>> > > >>> > How can this be done? Do I use Originate from within my .NET > > >>> > (mod_managed) code? > > >>> > > >>> Yes, you would be bridging the two legs like any normal call. Instead > > >>> of going to an endpoint you're going back out over the PSTN. > > >>> > > >>> > Do I get charged for both incoming and outbound call until the > entire > > >>> > session ends? Is there a way to receive call, validate and then > sort > > >>> > of transfer and then terminate the received call so I do not get > > >>> > charged for both? > > >>> > > >>> That would depend on your provider but most likely yes. As for > > >>> terminating one end after validation that is not going to happen. A > leg > > >>> terminates to you on an agreed fee schedule. No carrier that I know > of > > >>> supports that. Now if you kept everything SIP... you could do a > > >>> transfer after the validation. > > >>> > > >>> > > >>> > > >>> Anthony C. > > >>> > > >>> > > >>> _______________________________________________ > > >>> FreeSWITCH-users mailing list > > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-user > > >>>s http://www.freeswitch.org > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > -- > > > Ghulam Mustafa > > > cell: +92 333.611.7681 > > > sip: cyrenity at ekiga.net > > > mail: mustafa.pk at gmail.com > > > web: cyrenity.wordpress.com > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/c62df9cf/attachment.html From michael.scheidell at secnap.com Wed Aug 25 12:17:43 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Wed, 25 Aug 2010 15:17:43 -0400 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> Message-ID: <4C756C57.8060700@secnap.com> On 8/24/10 9:43 PM, David Ponzone wrote: > Michael, > > what you want to achieve is pretty simple to do with FreeSWITCH, and > is quite close from what I do. > Thanks for the reply (simple once I know where everything goes.. :-) I just loaded this last night. nothing like the Mitel SX50's (bit mapped options, digital pbx), 3com nbx (not really VOIP. was vo layer2 multicast), and more powerful then sipxecs.. > Or you do like me: you split calls coming into public to a specific > contex, based on source IP with such extensions in public.xml: > > > > > > > > is above right? from_provider, then $1 XML provider? (assuming _{provider} is the friendly name) > > > > > > > > And you just have to add the right extensions in those dialplans to > bridge calls to the other side. > That's the easy part. > famous last words. where caller is 561-555-1212 and destination is 301-555-1212. I see it coming from provider in public context:(i hope) close: Dialplan: sofia/external/5615551212 at 68.100.226.97 parsing [public->from_provider] continue=false Dialplan: sofia/external/5615551212 at 68.100.226.97 Regex (PASS) [from_provider] network_addr(68.233.226.97) =~ /68.233.226.97/ break=on-false Dialplan: sofia/external/5615551212 at 68.100.226.97 Regex (PASS) [from_provider] destination_number(3015551212) =~ /^(.*)$/ break=on-false Dialplan: sofia/external/5615551212 at 68.100.226.97 Action transfer(3015551212 XML provider) [snip] 2010-08-25 10:52:16.276232 [DEBUG] mod_dptools.c:748 sofia/external/5615551212 at 68.100.226.97 SET [outside_call]=[true] [snip] gets this far: 2010-08-25 10:52:16.278237 [INFO] mod_dialplan_xml.c:315 Processing 5615551212->3015551212 in context provider 2010-08-25 10:52:16.278237 [WARNING] mod_dialplan_xml.c:345 Context provider not found I guess I don't understand. as a test, PSTN number is 3015551212, coming from provider to sipxecs, do I put something like this in dialplan/provider.xml? dialplan/public/provider.xml? (I just want the call send to sipxecs. I don't care about ringback, timeouts.. I guess.. maybe I do for safety) so if a call come in to 3015551212, and its handled by sipxecs, and, MAYBE, I think I want it to fall back to a cell phone if sipxecs is down (say a 120 second timeout?) what do I put here? (what do I per where?) or this? (with 'bridge' I get 'so far', but still get 2010-08-25 15:14:15.590404 [INFO] mod_dialplan_xml.c:315 Processing 5615551212->3015551212 in context provider 2010-08-25 15:14:15.590404 [WARNING] mod_dialplan_xml.c:345 Context provider not found 2010-08-25 15:14:15.590404 [INFO] switch_core_state_machine.c:136 No Route, Aborting -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 sip:michael.scheidell at secnap.com > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/c6d505b4/attachment-0001.html From tculjaga at gmail.com Wed Aug 25 13:14:47 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 25 Aug 2010 22:14:47 +0200 Subject: [Freeswitch-users] long DTMF In-Reply-To: <984085.22821.qm@web29214.mail.ird.yahoo.com> References: <984085.22821.qm@web29214.mail.ird.yahoo.com> Message-ID: On Wed, Aug 25, 2010 at 7:47 PM, Nigel Kent wrote: > Have you tried using the event socket library > http://wiki.freeswitch.org/wiki/Esl > > well, i could have used esl from the start but thats and external application controling FS... and its totally a different approach from where im now:(. I have everything up & running except this feature .. so is there any chance to run a lua script in background (by allowing the normal callflow to go on) to collect a special DTMF event ? something like this: ? T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/dafcf915/attachment.html From steveayre at gmail.com Wed Aug 25 13:35:22 2010 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 25 Aug 2010 21:35:22 +0100 Subject: [Freeswitch-users] dialplan variables in XML cdr In-Reply-To: <4C752D64.3080506@communicatefreely.net> References: <4C752D64.3080506@communicatefreely.net> Message-ID: Are you setting them on the A-leg and looking for them on the b-leg cdr? If so, look at the export app: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export Or are you setting it on the B-leg and looking for it on the A-leg? If so, look at setting the import variable: http://wiki.freeswitch.org/wiki/Variable_import Apart from that, as brian said the xml cdr will contain every variable. If it's not there, it's not set. -Steve On 25 August 2010 15:49, Tim St. Pierre wrote: > Hello, > > Is there a way to have some custom dialplan variables appear in the XML cdr > output? We want to log > some things about the call, such as what option was chosen, which carrier > route, etc. I want to be > able to set a value in the dialplan and then figure out what it was in the > CDR so we can do reports > on it later. The variables I set aren't in the section. Is > there any way to enable > this, or am I missing something? > > Thanks! > > -Tim > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/15016ec5/attachment.html From michael.scheidell at secnap.com Wed Aug 25 13:46:01 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Wed, 25 Aug 2010 16:46:01 -0400 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <4C756C57.8060700@secnap.com> References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> Message-ID: <4C758109.3070002@secnap.com> this doesn't 'feel' right: On 8/25/10 3:17 PM, Michael Scheidell wrote: > > > > > > > if its FROM _provider, shouldn't you transfer to $1 XML sipxecs? and you are using 'transfer' and not 'bridge', right? -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/d320ad9f/attachment.html From david.ponzone at ipeva.fr Wed Aug 25 16:27:49 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 26 Aug 2010 01:27:49 +0200 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <4C756C57.8060700@secnap.com> References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> Message-ID: <74FBF66A-2A13-47F9-95BB-92A6D6BAF99E@ipeva.fr> What you need to add is the provider context. For instance, add in dialplan/ a file called provider.xml with the following content: David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 21:17, Michael Scheidell a ?crit : > > > On 8/24/10 9:43 PM, David Ponzone wrote: >> >> Michael, >> >> what you want to achieve is pretty simple to do with FreeSWITCH, >> and is quite close from what I do. >> > Thanks for the reply > (simple once I know where everything goes.. :-) > I just loaded this last night. nothing like the Mitel SX50's (bit > mapped options, digital pbx), 3com nbx (not really VOIP. was vo > layer2 multicast), and more powerful then sipxecs.. > >> Or you do like me: you split calls coming into public to a specific >> contex, based on source IP with such extensions in public.xml: >> >> >> >> >> >> >> >> > is above right? from_provider, then $1 XML provider? (assuming > _{provider} is the friendly name) >> >> >> >> >> >> >> >> And you just have to add the right extensions in those dialplans to >> bridge calls to the other side. >> That's the easy part. >> > famous last words. where caller is 561-555-1212 and destination is > 301-555-1212. I see it coming from provider in public context:(i hope) > close: > Dialplan: sofia/external/5615551212 at 68.100.226.97 parsing [public- > >from_provider] continue=false > Dialplan: sofia/external/5615551212 at 68.100.226.97 Regex (PASS) > [from_provider] network_addr(68.233.226.97) =~ /68.233.226.97/ > break=on-false > Dialplan: sofia/external/5615551212 at 68.100.226.97 Regex (PASS) > [from_provider] destination_number(3015551212) =~ /^(.*)$/ break=on- > false > Dialplan: sofia/external/5615551212 at 68.100.226.97 Action > transfer(3015551212 XML provider) > [snip] > 2010-08-25 10:52:16.276232 [DEBUG] mod_dptools.c:748 sofia/external/5615551212 at 68.100.226.97 > SET [outside_call]=[true] > [snip] gets this far: > 2010-08-25 10:52:16.278237 [INFO] mod_dialplan_xml.c:315 Processing > 5615551212->3015551212 in context provider > 2010-08-25 10:52:16.278237 [WARNING] mod_dialplan_xml.c:345 Context > provider not found > > I guess I don't understand. > as a test, PSTN number is 3015551212, coming from provider to > sipxecs, do I put something like this in dialplan/provider.xml? > dialplan/public/provider.xml? > > (I just want the call send to sipxecs. I don't care about ringback, > timeouts.. I guess.. maybe I do for safety) > so if a call come in to 3015551212, and its handled by sipxecs, and, > MAYBE, I think I want it to fall back to a cell phone if sipxecs is > down > (say a 120 second timeout?) > what do I put here? (what do I per where?) > > or this? > > > > > > > (with 'bridge' I get 'so far', but still get > 2010-08-25 15:14:15.590404 [INFO] mod_dialplan_xml.c:315 Processing > 5615551212->3015551212 in context provider > 2010-08-25 15:14:15.590404 [WARNING] mod_dialplan_xml.c:345 Context > provider not found > 2010-08-25 15:14:15.590404 [INFO] switch_core_state_machine.c:136 No > Route, Aborting > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 sip:michael.scheidell at secnap.com > > > | SECNAP Network Security Corporation > Certified SNORT Integrator > 2008-9 Hot Company Award Winner, World Executive Alliance > Five-Star Partner Program 2009, VARBusiness > Best in Email Security,2010: Network Products Guide > King of Spam Filters, SC Magazine 2008 > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see http://www.secnap.com/products/spammertrap/ > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/caee7fe4/attachment-0001.html From david.ponzone at ipeva.fr Wed Aug 25 16:33:05 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 26 Aug 2010 01:33:05 +0200 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <4C758109.3070002@secnap.com> References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> <4C758109.3070002@secnap.com> Message-ID: <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> It is right. You just transfer to another dialplan (context) in order to make it more readable. You'll be able to add all your rules in the provider context, because only calls coming from provider X.X.X.X will go into this dialplan. Remember: extension name has absolutely NO importance in FS. It's just a label. So to sum it all: if call from X.X.X.X, transfer to context provider, with the same destination_number ($1). In context provider, if destination_number is XXX, then bridge the call to sipxecs. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 25/08/2010 ? 22:46, Michael Scheidell a ?crit : > this doesn't 'feel' right: > > On 8/25/10 3:17 PM, Michael Scheidell wrote: >> >> >> >> >> >> >> >> > if its FROM _provider, shouldn't you transfer to $1 XML sipxecs? > > and you are using 'transfer' and not 'bridge', right? > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > > | SECNAP Network Security Corporation > Certified SNORT Integrator > 2008-9 Hot Company Award Winner, World Executive Alliance > Five-Star Partner Program 2009, VARBusiness > Best in Email Security,2010: Network Products Guide > King of Spam Filters, SC Magazine 2008 > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see http://www.secnap.com/products/spammertrap/ > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/75c01ae3/attachment.html From dujinfang at gmail.com Wed Aug 25 17:37:50 2010 From: dujinfang at gmail.com (Seven Du) Date: Thu, 26 Aug 2010 08:37:50 +0800 Subject: [Freeswitch-users] eavesdrop question In-Reply-To: References: Message-ID: On Wed, Aug 25, 2010 at 9:38 PM, Seven Du wrote: > Hi, > > Playing with eavesdrop, it's very nice. And have two questions: > > 1) A talk to B. C eavesdrop the chanel. C can control whom he want to > whisper to by DTMF. Is it possible to use an API/ESL to control ? > Would like to make a patch if makes sense. ?uuid_send_dtmf can send to > the remote party, would it make sense to implement a uuid_fake_dtmf to > ?generate dtmf instead of from the remote party? > ignore this one, I found uuid_recv_dtmf in src. awesome! will doc on wiki :p. still looking an answer for the following question. > > 2) I tested another scenario with: A bridged to B. ?C eavesdrop the > channel. ?And then D eavesdrop too. > > C can input DTMF 1/2/3, C worked as a normal eavesdrop. D has exactly > the same behavior with C. (e.g. when C talk to A, D also can talk to > A, when C talk to B, D can also talk to B). > > D cannot control anything with DTMF. > > > Is it possible for the eavesdrop also respond to D's DTMF ? ?Ideally C > and D has not effect to each other. > > Thanks. > -- > Blog: http://www.dujinfang.com > Proj:? http://www.freeswitch.org.cn > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From chat2jesse at gmail.com Wed Aug 25 18:11:32 2010 From: chat2jesse at gmail.com (jesse) Date: Wed, 25 Aug 2010 18:11:32 -0700 Subject: [Freeswitch-users] php esl compilation PBs In-Reply-To: <20100825041152.0c5a17c4@anubis.defcon1> References: <20100825041152.0c5a17c4@anubis.defcon1> Message-ID: I don't have issue to build and install phpmod in a second server. but building failed in the 1st server because of two versions of PHP 5.2.4 & 5.3.3. try a different server. the build environment set up is painful. -jesse On Tue, Aug 24, 2010 at 7:11 PM, Jean-Yves F. Barbier <12ukwn at gmail.com> wrote: > Debian sid > ========== > > Hi list, > > I'm trying to figure out a way to compile ESL for PHP on my system. > > I was obliged to copy ../src/include/* into .../esl/php and modify > them a bit, but I still have an error: > > g++ ? -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM > -I/usr/include/php5/Zend -I/usr/include/php5/ext > -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 > -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > > esl_wrap.cpp:2583: attention : deprecated conversion from string constant > to ?char*? > > g++ ?esl_wrap.o ? -lcrypt -lcrypt -lonig -ldb -lqdbm -lrt -lm > -ldl -lnsl -lcrypt -lcrypt -lpthread -o ESL.so -L. > > /usr/bin/ld: cannot find -lonig collect2: ld a retourn? 1 code d'?tat > d'ex?cution make: *** [ESL.so] > > Error 1 > > Apparently it is a missing library (onig), but which one? > > JY > -- > On a clear disk you can seek forever. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mnhassan at usa.net Wed Aug 25 20:28:41 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Thu, 26 Aug 2010 09:28:41 +0600 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: I'm not sure if "make phpmod-install" will work. I think you need to do the "make phpmod", and then copy the file to the "ESL.so" file to the php modules folder for you OS. On my RHEL 5.5 i386, it is /usr/lib/php/modules/ESL.so Can you confirm if you see the error with just "make phpmod" as on the wiki? Regards HASSAN On Wed, Aug 25, 2010 at 12:39, jesse wrote: > HASSAN: > > my pasted error message in the previous e-mail was generated from > : make phpmod-install. > > -jesse > > On Tue, Aug 24, 2010 at 6:12 PM, Nyamul Hassan wrote: > > * enter the /libs/esl directory > > * run "make" > > * run "make phpmod" > > If you face error in any step above, then please output the "full output" > > into > > http://pastebin.freeswitch.org > > and let us know the pastebin link. > > Regards > > HASSAN > > > > On Wed, Aug 25, 2010 at 06:07, jesse wrote: > >> > >> during installation: > >> > >> make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' > >> make[1]: Nothing to be done for `all'. > >> make[1]: Leaving directory `/tmp/freeswitch/libs/esl/php' > >> make -C php install > >> PHP Warning: PHP Startup: ESL: Unable to initialize module > >> Module compiled with module API=20090626, debug=0, thread-safety=0 > >> PHP compiled with module API=20060613, debug=0, thread-safety=0 > >> These options need to match > >> in Unknown on line 0 > >> make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' > >> cp ESL.so /usr/lib/php5/20060613 > >> cp ESL.php /usr/share/php > >> > >> Then when I open the PHP script from browser, I always get > >> > >> Unable to initialize module Module compiled with module API=20090626, > >> debug=0, thread-safety=0 PHP compiled with module API=20060613, > >> debug=0, thread-safety=0 These options need to match in > >> /usr/share/php/ESL.php on line 23 > >> > >> any one is good at PHP can fix this issue? I guess the root cause is > >> that I have two PHP installed and my system is totally messed. I > >> tried to remove the new php installation, but still the same result. > >> > >> > >> > >> On Tue, Aug 24, 2010 at 4:56 PM, Nyamul Hassan > wrote: > >> > I got my phpmod installed and working great! Did you follow: > >> > http://wiki.freeswitch.org/wiki/Esl#Installation? > >> > Please mention at which steps you're seeing an error. > >> > Regards > >> > HASSAN > >> > > >> > > >> > On Wed, Aug 25, 2010 at 05:42, jesse wrote: > >> >> > >> >> the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : > >> >> > >> >> At some point before July 8, 2009, mod_php was removed from the > trunk. > >> >> There don't appear to be any concrete plans to reintroduce mod_php, > so > >> >> if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket or > >> >> the PHP ESL library. > >> >> > >> >> > >> >> On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale > >> >> wrote: > >> >> > We never have had PHP in our code distribution. > >> >> > > >> >> > > >> >> > On Mon, Aug 23, 2010 at 4:39 PM, jesse > wrote: > >> >> >> The current FS version doesn't include source code for PHP. > >> >> >> What are the steps to install it on top of my current version? > >> >> >> where > >> >> >> to get the source? how to build and install? > >> >> >> > >> >> >> Iif you guys want to keep the new system slim by get rid of PHP, > >> >> >> please at least keep a doc about how to add it in case of need. > >> >> >> > >> >> >> thanks! > >> >> >> > >> >> >> -jesse > >> >> >> > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/9272d382/attachment-0001.html From farhan.husain at csebuet.org Wed Aug 25 22:27:33 2010 From: farhan.husain at csebuet.org (Farhan Husain) Date: Thu, 26 Aug 2010 00:27:33 -0500 Subject: [Freeswitch-users] Module written in C++ Message-ID: I am trying to write a module having multiple C++ source files. When I run make, I find that only the C++ file having the name same as the module name is being compiled and all others do not produce any object file. As a result when I run FS I get linker error. Can anyone tell me how to get them compiled too? Thanks, Farhan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/2abaaca2/attachment.html From chat2jesse at gmail.com Wed Aug 25 22:49:56 2010 From: chat2jesse at gmail.com (jesse) Date: Wed, 25 Aug 2010 22:49:56 -0700 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> <4C758109.3070002@secnap.com> <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> Message-ID: I feel Michael's idea should also work: If call is from X.X.X.X, then bridge the call with the same destination_number ($1) to sipxecs gateway. On Wed, Aug 25, 2010 at 4:33 PM, David Ponzone wrote: > It is right. > You just transfer to another dialplan (context) in order to make it more > readable. > You'll be able to add all your rules in the provider context, because only > calls coming from provider X.X.X.X will go into this dialplan. > Remember: extension name has absolutely NO importance in FS. It's just a > label. > So to sum it all: > if call from X.X.X.X, transfer to context provider, with the same > destination_number ($1). > In context provider, if destination_number is XXX, then bridge the call to > sipxecs. > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 25/08/2010 ? 22:46, Michael Scheidell a ?crit : > > this doesn't 'feel' right: > > On 8/25/10 3:17 PM, Michael Scheidell wrote: > > ?? > ?? ? > ?? ? > ?? ? ? > ?? ? > ?? > > if its FROM _provider, shouldn't you transfer to $1 XML sipxecs? > > and you are using 'transfer' and not 'bridge', right? > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 >>?|?SECNAP Network Security Corporation > > Certified SNORT Integrator > 2008-9 Hot Company Award Winner, World Executive Alliance > Five-Star Partner Program 2009, VARBusiness > Best in Email Security,2010: Network Products Guide > King of Spam Filters, SC Magazine 2008 > > ________________________________ > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see?http://www.secnap.com/products/spammertrap/ > > ________________________________ > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From chat2jesse at gmail.com Wed Aug 25 22:58:14 2010 From: chat2jesse at gmail.com (jesse) Date: Wed, 25 Aug 2010 22:58:14 -0700 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: I had no issue with make phpmod . the problem is related with sudo make phpmod-install. In another FS server box, my php is working perfect. Foe the the machine whose phpmod didn't work, I am 100% sure the root cause is related with two PHP versions 5.2.4 and 5.3.3. In fact, ESL.so is in /usr/lib/php/modules/ESL.so, however it was compiled with 5.3.3 version PHP and ran in LAMP apache with PHP 5.2.4. I don't know how to enforce the build of ESL.so with PHP 5.2.4....... -jesse On Wed, Aug 25, 2010 at 8:28 PM, Nyamul Hassan wrote: > I'm not sure if "make phpmod-install" will work. ?I think you need to do the > "make phpmod", and then copy the file to the "ESL.so" file to the php > modules folder for you OS. > On my RHEL 5.5 i386, it is /usr/lib/php/modules/ESL.so > Can you confirm if you see the error with just "make phpmod" as on the wiki? > Regards > HASSAN > > On Wed, Aug 25, 2010 at 12:39, jesse wrote: >> >> HASSAN: >> >> ? ?my pasted error message in the previous e-mail was generated from >> : make phpmod-install. >> >> -jesse >> >> On Tue, Aug 24, 2010 at 6:12 PM, Nyamul Hassan wrote: >> > * ?enter the /libs/esl directory >> > * ?run "make" >> > * ?run "make phpmod" >> > If you face error in any step above, then please output the "full >> > output" >> > into >> > http://pastebin.freeswitch.org >> > and let us know the pastebin link. >> > Regards >> > HASSAN >> > >> > On Wed, Aug 25, 2010 at 06:07, jesse wrote: >> >> >> >> during installation: >> >> >> >> make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' >> >> make[1]: Nothing to be done for `all'. >> >> make[1]: Leaving directory `/tmp/freeswitch/libs/esl/php' >> >> make -C php install >> >> PHP Warning: ?PHP Startup: ESL: Unable to initialize module >> >> Module compiled with module API=20090626, debug=0, thread-safety=0 >> >> PHP ? ?compiled with module API=20060613, debug=0, thread-safety=0 >> >> These options need to match >> >> ?in Unknown on line 0 >> >> make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' >> >> cp ESL.so /usr/lib/php5/20060613 >> >> cp ESL.php /usr/share/php >> >> >> >> Then when I open the PHP script from browser, I always get >> >> >> >> Unable to initialize module Module compiled with module API=20090626, >> >> debug=0, thread-safety=0 PHP compiled with module API=20060613, >> >> debug=0, thread-safety=0 These options need to match in >> >> /usr/share/php/ESL.php on line 23 >> >> >> >> any one is good at PHP can fix this issue? I guess the root cause is >> >> that I have two PHP installed and my system is totally messed. ?I >> >> tried to remove the new php installation, but still the same result. >> >> >> >> >> >> >> >> On Tue, Aug 24, 2010 at 4:56 PM, Nyamul Hassan >> >> wrote: >> >> > I got my phpmod installed and working great! ?Did you follow: >> >> > http://wiki.freeswitch.org/wiki/Esl#Installation? >> >> > Please mention at which steps you're seeing an error. >> >> > Regards >> >> > HASSAN >> >> > >> >> > >> >> > On Wed, Aug 25, 2010 at 05:42, jesse wrote: >> >> >> >> >> >> the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : >> >> >> >> >> >> At some point before July 8, 2009, mod_php was removed from the >> >> >> trunk. >> >> >> There don't appear to be any concrete plans to reintroduce mod_php, >> >> >> so >> >> >> if you'd like to use PHP with FreeSWITCH, use the PHP Event Socket >> >> >> or >> >> >> the PHP ESL library. >> >> >> >> >> >> >> >> >> On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale >> >> >> wrote: >> >> >> > We never have had PHP in our code distribution. >> >> >> > >> >> >> > >> >> >> > On Mon, Aug 23, 2010 at 4:39 PM, jesse >> >> >> > wrote: >> >> >> >> The current FS version doesn't include source code for PHP. >> >> >> >> What are the steps to install it on top of my current version? >> >> >> >> ?where >> >> >> >> to get the source? how to build and install? >> >> >> >> >> >> >> >> Iif you guys want to keep the new system slim by get rid of PHP, >> >> >> >> please at least keep a doc about how to add it in case of need. >> >> >> >> >> >> >> >> thanks! >> >> >> >> >> >> >> >> -jesse >> >> >> >> >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From farhan.husain at csebuet.org Wed Aug 25 22:58:47 2010 From: farhan.husain at csebuet.org (Farhan Husain) Date: Thu, 26 Aug 2010 00:58:47 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Module written in C++ In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper> Message-ID: I just got the solution from Math at the IRC channel. Here is the link to the solution he gave me: http://pastebin.freeswitch.org/13727 On Thu, Aug 26, 2010 at 12:36 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Have you updated the Makefile to build the other files? > > /Peter > ________________________________________ > Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [ > freeswitch-dev-bounces at lists.freeswitch.org] för Farhan Husain [ > farhan.husain at csebuet.org] > Skickat: den 26 augusti 2010 07:27 > Till: FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org > ?mne: [Freeswitch-dev] Module written in C++ > > I am trying to write a module having multiple C++ source files. When I run > make, I find that only the C++ file having the name same as the module name > is being compiled and all others do not produce any object file. As a result > when I run FS I get linker error. Can anyone tell me how to get them > compiled too? > > Thanks, > Farhan > !DSPAM:4c75fc8232934761421429! > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/c5ec345f/attachment.html From chat2jesse at gmail.com Wed Aug 25 23:00:40 2010 From: chat2jesse at gmail.com (jesse) Date: Wed, 25 Aug 2010 23:00:40 -0700 Subject: [Freeswitch-users] Module written in C++ In-Reply-To: References: Message-ID: I would suggest try a small module project without any material business logic. If you still can not get it working, please upload your project files to some website so that people can help you.. -jesse On Wed, Aug 25, 2010 at 10:27 PM, Farhan Husain wrote: > I am trying to write a module having multiple C++ source files. When I run > make, I find that only the C++ file having the name same as the module name > is being compiled and all others do not produce any object file. As a result > when I run FS I get linker error. Can anyone tell me how to get them > compiled too? > Thanks, > Farhan > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From vince.freeswitch at hightek.org Wed Aug 25 23:45:08 2010 From: vince.freeswitch at hightek.org (Vincent Stemen) Date: Thu, 26 Aug 2010 01:45:08 -0500 Subject: [Freeswitch-users] Platform requrest (DragonflyBSD) for the Jira issue management system In-Reply-To: References: <20100728053058.GA26436@quark.hightek.org> <20100821220315.GA52528@quark.hightek.org> <3BFC3227-59D0-4D45-974E-1E3C04EF6547@freeswitch.org> <20100822154128.GA62723@quark.hightek.org> Message-ID: <20100826064508.GA14271@quark.hightek.org> On Mon, Aug 23, 2010 at 02:04:25PM -0400, Michael Jerris wrote: > > On Aug 22, 2010, at 11:41 AM, Vincent Stemen wrote: > > > On Sat, Aug 21, 2010 at 05:45:47PM -0500, Brian West wrote: > >> Vince, > >> I see no value in adding it... but I added it anyway... > > > > I appreciate it. May I ask why you do not see any value in it though? > > If you prefer to encompass all the BSD's into one platform option, I > > would suggest changing the "FreeBSD/gcc" option to "BSD/gcc". > > It can't be the same classification. The reason it didn't build in > the past was due to dragonfly specific issues. There is no issue > having it in the list if there are active community members providing > patches to fix these issues. Are you able to volunteer to provide > patches for all dragonfly specific issues? > > Mike I can volunteer to provide patches for issues that I am capable of fixing. I don't know if I can guarantee that I can provide patches for *all* dragonfly issues :-). BTW, I do have it running on dragonfly and it has been working great. With just a couple of phone adapters. No heavy load. We have completely switched over from asterisk. I do have one small patch to bootstrap.sh, so far, that I have been meaning to provide. I will try to get a change to create a jira issue sometime in the next few of days for it. Vince From mthakershi at gmail.com Wed Aug 25 23:48:17 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Thu, 26 Aug 2010 01:48:17 -0500 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: <4C756922.4060206@secnap.com> References: <1282691200.4420.23.camel@anthony-desktop> <4C756922.4060206@secnap.com> Message-ID: That looks good for incoming DIDs. Any idea about Vonage Small Business Premium Unlimited plan? It does provide SIP option / integration. - Will it let me make simultaneous outbound calls from my FS box? On Wed, Aug 25, 2010 at 2:04 PM, Michael Scheidell < michael.scheidell at secnap.com> wrote: > don't know vitelity, > but voip.ms. supports user/password or ip based auth. has built in > Vmail, redirect on network outage. > us did's are .99/c per month, and calls 1.1c/ month. (20? trunks per did?) > > > > > > On 8/25/10 2:28 PM, Malay Thakershi wrote: > > Can someone suggest me wholesale and cheaper provider than Vitelity but > offering similar service? > > In/Out bound both USA. > > Thank you. > > On Wed, Aug 25, 2010 at 3:33 AM, David Ponzone wrote: > >> AFAIK, most (all ?) providers wont accept the transfer. >> They are not allowed to change the rate of the call from the caller >> perspective after the call was initiated. >> Imagine you transfer the call to an Iridium number... >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >> non autoris?e est interdite. Tout message ?lectronique est susceptible >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas >> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur.* >> * >> * >> >> >> >> Le 25/08/2010 ? 07:44, Ghulam Mustafa a ?crit : >> >> Hi, >> >> i am wondering what will happen if you send a SIP REFER (transfer) after >> validating leg-a without actually answering the call; will it still cost you >> 2 * 1.44 >> >> :/ >> >> -mustafa >> >> On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi wrote: >> >>> I am currently looking at Vitelity. They have 1.44 cents per minute. They >>> do charge for incoming/outgoing both. >>> >>> I am not literally dialing out to a phone. I want to dial a mobile >>> number or any other US number for that matter. >>> >>> Here is what I intent to do: >>> 1. A US phone dials to Vitelity number -- comes to my FS box >>> 2. I validate few things >>> 3. Dial another US number and connect received call to that one >>> >>> So as you said I will be charged 2 * 1.44 (since I can't terminate the >>> arrived call after validation). >>> >>> Is there any other way to the sequence I have specified above? >>> >>> Are there providers similar to Vitelity but cheaper (with relatively >>> same features)? >>> >>> Thank you. >>> >>> On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove wrote: >>> >>>> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: >>>> > Hello, >>>> > >>>> > >>>> > It would be a great help if someone can guide me. >>>> > >>>> > >>>> > 1. I would like to first receive a call, perform certain validations. >>>> > (Able to do this via mod_managed application that handles call from >>>> > dialplan). >>>> > >>>> >>>> That should not be a problem, I don't know your requirements so can't >>>> provide a full answer. >>>> >>>> > >>>> > 2. Now, I would like to dial out to a PSTN number so that received >>>> > call is connected to this new outbound number. >>>> > >>>> >>>> This can be done and is called hairpinning. >>>> >>>> > >>>> > How can this be done? Do I use Originate from within my .NET >>>> > (mod_managed) code? >>>> > >>>> >>>> Yes, you would be bridging the two legs like any normal call. Instead >>>> of >>>> going to an endpoint you're going back out over the PSTN. >>>> >>>> > >>>> > Do I get charged for both incoming and outbound call until the entire >>>> > session ends? Is there a way to receive call, validate and then sort >>>> > of transfer and then terminate the received call so I do not get >>>> > charged for both? >>>> >>>> That would depend on your provider but most likely yes. As for >>>> terminating one end after validation that is not going to happen. A leg >>>> terminates to you on an agreed fee schedule. No carrier that I know of >>>> supports that. Now if you kept everything SIP... you could do a transfer >>>> after the validation. >>>> >>>> >>>> >>>> Anthony C. >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Ghulam Mustafa >> cell: +92 333.611.7681 >> sip: cyrenity at ekiga.net >> mail: mustafa.pk at gmail.com >> web: cyrenity.wordpress.com >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > > *| *SECNAP Network Security Corporation > > - Certified SNORT Integrator > - 2008-9 Hot Company Award Winner, World Executive Alliance > - Five-Star Partner Program 2009, VARBusiness > - Best in Email Security,2010: Network Products Guide > - King of Spam Filters, SC Magazine 2008 > > > ------------------------------ > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see http://www.secnap.com/products/spammertrap/ > ------------------------------ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/4fcbbb02/attachment-0001.html From a.afzali2003 at gmail.com Thu Aug 26 00:21:48 2010 From: a.afzali2003 at gmail.com (afshin afzali) Date: Thu, 26 Aug 2010 11:51:48 +0430 Subject: [Freeswitch-users] 3PCC USING FIFO / CALLCENTER Message-ID: Hi FreeSWITCH, I am thinking to use mod_fifo / mod_callcenter in a 3pcc scenario which the first party of call will be selected dynamically. So I should be able to originate call into these modules. I don't know if it is possible right now or if could be requested as a new feature. BEST, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/e098f5fc/attachment.html From david.ponzone at ipeva.fr Thu Aug 26 00:59:44 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 26 Aug 2010 09:59:44 +0200 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> <4C758109.3070002@secnap.com> <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> Message-ID: <6B192D76-9AAC-4C57-A005-2AF12FDB2F71@ipeva.fr> Sure it will work, it's basically the same. My idea was just to have a specific file with only the specific dialplan for each direction. But if he never needs to add anything in the dialplan, it's perhaps overkill. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 26/08/2010 ? 07:49, jesse a ?crit : > I feel Michael's idea should also work: If call is from X.X.X.X, then > bridge the call with the same > destination_number ($1) to sipxecs gateway. > > > > > > > > > > On Wed, Aug 25, 2010 at 4:33 PM, David Ponzone > wrote: >> It is right. >> You just transfer to another dialplan (context) in order to make it >> more >> readable. >> You'll be able to add all your rules in the provider context, >> because only >> calls coming from provider X.X.X.X will go into this dialplan. >> Remember: extension name has absolutely NO importance in FS. It's >> just a >> label. >> So to sum it all: >> if call from X.X.X.X, transfer to context provider, with the same >> destination_number ($1). >> In context provider, if destination_number is XXX, then bridge the >> call to >> sipxecs. >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou >> diffusion >> non autoris?e est interdite. Tout message ?lectronique est >> susceptible >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce >> message s'il >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire >> de ce >> message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur. >> >> >> >> Le 25/08/2010 ? 22:46, Michael Scheidell a ?crit : >> >> this doesn't 'feel' right: >> >> On 8/25/10 3:17 PM, Michael Scheidell wrote: >> >> >> >> >> >> >> >> >> if its FROM _provider, shouldn't you transfer to $1 XML sipxecs? >> >> and you are using 'transfer' and not 'bridge', right? >> >> -- >> Michael Scheidell, CTO >> o: 561-999-5000 >> d: 561-948-2259 >> ISN: 1259*1300 >>> | SECNAP Network Security Corporation >> >> Certified SNORT Integrator >> 2008-9 Hot Company Award Winner, World Executive Alliance >> Five-Star Partner Program 2009, VARBusiness >> Best in Email Security,2010: Network Products Guide >> King of Spam Filters, SC Magazine 2008 >> >> ________________________________ >> >> This email has been scanned and certified safe by SpammerTrap?. >> For Information please see http://www.secnap.com/products/ >> spammertrap/ >> >> ________________________________ >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/8cd7b2fb/attachment.html From fdelawarde at wirelessmundi.com Thu Aug 26 01:04:06 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Thu, 26 Aug 2010 10:04:06 +0200 Subject: [Freeswitch-users] MWI doubt In-Reply-To: <40456555-8929-4C6C-AD74-EEB81AA2D08B@freeswitch.org> References: <1282746307.2970.135.camel@luna.tc.commsmundi.com> <4C752CCF.7020408@communicatefreely.net> <92217374-5098-460F-A949-74E8A749D176@freeswitch.org> <1282750369.2970.157.camel@luna.tc.commsmundi.com> <1282751842.2970.178.camel@luna.tc.commsmundi.com> <2909F380-BC65-4021-AAF8-7D22E02B2253@freeswitch.org> <1282755810.2970.194.camel@luna.tc.commsmundi.com> <40456555-8929-4C6C-AD74-EEB81AA2D08B@freeswitch.org> Message-ID: <1282809846.2970.210.camel@luna.tc.commsmundi.com> Well if it was intended that's a very nice and important feature for multi-homed setups, thanks! I'll test more presence stuff and try to write something about this in the wiki because there seem to be lots of confusion about the sharing of presence across profiles. As many people, I tended to think it would need a feature request as well, as domains would need to be aliased on multiple profiles which is impossible. Now I know there is no need in doing that to make it work. Thanks, Fran?ois. On Wed, 2010-08-25 at 12:08 -0500, Brian West wrote: > No the bonding of profiles is fine... and makes sofia_contact behave > differently... thats a totally different use case... The bottom line > is you can't have the same domain aliased on multiple profiles... > thats all I said in my first response. If this works for you then > thats fine I'm pretty sure that was our intended behavior for this > use. > > /b From david.ponzone at ipeva.fr Thu Aug 26 01:03:52 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 26 Aug 2010 10:03:52 +0200 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: <441A94AC-61FC-412A-9650-03EC16535619@ipeva.fr> I think having 2 different versions of PHP5 on the same machine is perhaps the issue you shoud solve. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 26/08/2010 ? 07:58, jesse a ?crit : > I had no issue with make phpmod . the problem is related with sudo > make phpmod-install. > > In another FS server box, my php is working perfect. Foe the the > machine whose phpmod didn't work, I am 100% sure the root cause is > related with two PHP versions 5.2.4 and 5.3.3. In fact, ESL.so is in > /usr/lib/php/modules/ESL.so, however it was compiled with 5.3.3 > version PHP and ran in LAMP apache with PHP 5.2.4. > > I don't know how to enforce the build of ESL.so with PHP 5.2.4....... > > -jesse > > > > On Wed, Aug 25, 2010 at 8:28 PM, Nyamul Hassan > wrote: >> I'm not sure if "make phpmod-install" will work. I think you need >> to do the >> "make phpmod", and then copy the file to the "ESL.so" file to the php >> modules folder for you OS. >> On my RHEL 5.5 i386, it is /usr/lib/php/modules/ESL.so >> Can you confirm if you see the error with just "make phpmod" as on >> the wiki? >> Regards >> HASSAN >> >> On Wed, Aug 25, 2010 at 12:39, jesse wrote: >>> >>> HASSAN: >>> >>> my pasted error message in the previous e-mail was generated from >>> : make phpmod-install. >>> >>> -jesse >>> >>> On Tue, Aug 24, 2010 at 6:12 PM, Nyamul Hassan >>> wrote: >>>> * enter the /libs/esl directory >>>> * run "make" >>>> * run "make phpmod" >>>> If you face error in any step above, then please output the "full >>>> output" >>>> into >>>> http://pastebin.freeswitch.org >>>> and let us know the pastebin link. >>>> Regards >>>> HASSAN >>>> >>>> On Wed, Aug 25, 2010 at 06:07, jesse wrote: >>>>> >>>>> during installation: >>>>> >>>>> make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' >>>>> make[1]: Nothing to be done for `all'. >>>>> make[1]: Leaving directory `/tmp/freeswitch/libs/esl/php' >>>>> make -C php install >>>>> PHP Warning: PHP Startup: ESL: Unable to initialize module >>>>> Module compiled with module API=20090626, debug=0, thread-safety=0 >>>>> PHP compiled with module API=20060613, debug=0, thread-safety=0 >>>>> These options need to match >>>>> in Unknown on line 0 >>>>> make[1]: Entering directory `/tmp/freeswitch/libs/esl/php' >>>>> cp ESL.so /usr/lib/php5/20060613 >>>>> cp ESL.php /usr/share/php >>>>> >>>>> Then when I open the PHP script from browser, I always get >>>>> >>>>> Unable to initialize module Module compiled with module >>>>> API=20090626, >>>>> debug=0, thread-safety=0 PHP compiled with module API=20060613, >>>>> debug=0, thread-safety=0 These options need to match in >>>>> /usr/share/php/ESL.php on line 23 >>>>> >>>>> any one is good at PHP can fix this issue? I guess the root >>>>> cause is >>>>> that I have two PHP installed and my system is totally messed. I >>>>> tried to remove the new php installation, but still the same >>>>> result. >>>>> >>>>> >>>>> >>>>> On Tue, Aug 24, 2010 at 4:56 PM, Nyamul Hassan >>>>> wrote: >>>>>> I got my phpmod installed and working great! Did you follow: >>>>>> http://wiki.freeswitch.org/wiki/Esl#Installation? >>>>>> Please mention at which steps you're seeing an error. >>>>>> Regards >>>>>> HASSAN >>>>>> >>>>>> >>>>>> On Wed, Aug 25, 2010 at 05:42, jesse >>>>>> wrote: >>>>>>> >>>>>>> the wiki page http://wiki.freeswitch.org/wiki/Mod_php says : >>>>>>> >>>>>>> At some point before July 8, 2009, mod_php was removed from the >>>>>>> trunk. >>>>>>> There don't appear to be any concrete plans to reintroduce >>>>>>> mod_php, >>>>>>> so >>>>>>> if you'd like to use PHP with FreeSWITCH, use the PHP Event >>>>>>> Socket >>>>>>> or >>>>>>> the PHP ESL library. >>>>>>> >>>>>>> >>>>>>> On Mon, Aug 23, 2010 at 4:41 PM, Anthony Minessale >>>>>>> wrote: >>>>>>>> We never have had PHP in our code distribution. >>>>>>>> >>>>>>>> >>>>>>>> On Mon, Aug 23, 2010 at 4:39 PM, jesse >>>>>>>> wrote: >>>>>>>>> The current FS version doesn't include source code for PHP. >>>>>>>>> What are the steps to install it on top of my current version? >>>>>>>>> where >>>>>>>>> to get the source? how to build and install? >>>>>>>>> >>>>>>>>> Iif you guys want to keep the new system slim by get rid of >>>>>>>>> PHP, >>>>>>>>> please at least keep a doc about how to add it in case of >>>>>>>>> need. >>>>>>>>> >>>>>>>>> thanks! >>>>>>>>> >>>>>>>>> -jesse >>>>>>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/d052b2cf/attachment-0001.html From mike at jerris.com Thu Aug 26 01:12:03 2010 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Aug 2010 04:12:03 -0400 Subject: [Freeswitch-users] Platform requrest (DragonflyBSD) for the Jira issue management system In-Reply-To: <20100826064508.GA14271@quark.hightek.org> References: <20100728053058.GA26436@quark.hightek.org> <20100821220315.GA52528@quark.hightek.org> <3BFC3227-59D0-4D45-974E-1E3C04EF6547@freeswitch.org> <20100822154128.GA62723@quark.hightek.org> <20100826064508.GA14271@quark.hightek.org> Message-ID: On Aug 26, 2010, at 2:45 AM, Vincent Stemen wrote: > On Mon, Aug 23, 2010 at 02:04:25PM -0400, Michael Jerris wrote: >> >> On Aug 22, 2010, at 11:41 AM, Vincent Stemen wrote: >> >>> On Sat, Aug 21, 2010 at 05:45:47PM -0500, Brian West wrote: >>>> Vince, >>>> I see no value in adding it... but I added it anyway... >>> >>> I appreciate it. May I ask why you do not see any value in it though? >>> If you prefer to encompass all the BSD's into one platform option, I >>> would suggest changing the "FreeBSD/gcc" option to "BSD/gcc". >> >> It can't be the same classification. The reason it didn't build in >> the past was due to dragonfly specific issues. There is no issue >> having it in the list if there are active community members providing >> patches to fix these issues. Are you able to volunteer to provide >> patches for all dragonfly specific issues? >> >> Mike > > I can volunteer to provide patches for issues that I am capable of > fixing. I don't know if I can guarantee that I can provide patches for > *all* dragonfly issues :-). > > BTW, I do have it running on dragonfly and it has been working great. > With just a couple of phone adapters. No heavy load. We have > completely switched over from asterisk. > > I do have one small patch to bootstrap.sh, so far, that I have been > meaning to provide. I will try to get a change to create a jira issue > sometime in the next few of days for it. Great to hear, good luck, and I look forward to your patch. Thanks Mike From david.ponzone at ipeva.fr Thu Aug 26 01:18:41 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 26 Aug 2010 10:18:41 +0200 Subject: [Freeswitch-users] Inbound and then outbound call? In-Reply-To: References: <1282691200.4420.23.camel@anthony-desktop> <4C756922.4060206@secnap.com> Message-ID: <10BE4EE1-2D46-4844-95E7-C17C220B6867@ipeva.fr> Hmmm, I suspect you can make 2 calls, but I I would guess no more. The reason is it's an unlimited plan, and if they allow you to send 10 calls at the same time, their business model will be in jeopardy. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 26/08/2010 ? 08:48, Malay Thakershi a ?crit : > That looks good for incoming DIDs. > > Any idea about Vonage Small Business Premium Unlimited plan? It does > provide SIP option / integration. > - Will it let me make simultaneous outbound calls from my FS box? > > On Wed, Aug 25, 2010 at 2:04 PM, Michael Scheidell > wrote: > don't know vitelity, > but voip.ms. supports user/password or ip based auth. has built in > Vmail, redirect on network outage. > us did's are .99/c per month, and calls 1.1c/ month. (20? trunks per > did?) > > > > > > On 8/25/10 2:28 PM, Malay Thakershi wrote: >> >> Can someone suggest me wholesale and cheaper provider than Vitelity >> but offering similar service? >> >> In/Out bound both USA. >> >> Thank you. >> >> On Wed, Aug 25, 2010 at 3:33 AM, David Ponzone > > wrote: >> AFAIK, most (all ?) providers wont accept the transfer. >> They are not allowed to change the rate of the call from the caller >> perspective after the call was initiated. >> Imagine you transfer the call to an Iridium number... >> >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? l'intention exclusive de ses destinataires. Toute >> utilisation ou diffusion non autoris?e est interdite. Tout message >> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute >> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou >> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de >> le d?truire imm?diatement et d'avertir l'exp?diteur. >> >> >> >> >> Le 25/08/2010 ? 07:44, Ghulam Mustafa a ?crit : >> >>> Hi, >>> >>> i am wondering what will happen if you send a SIP REFER (transfer) >>> after validating leg-a without actually answering the call; will >>> it still cost you 2 * 1.44 >>> >>> :/ >>> >>> -mustafa >>> >>> On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi >> > wrote: >>> I am currently looking at Vitelity. They have 1.44 cents per >>> minute. They do charge for incoming/outgoing both. >>> >>> I am not literally dialing out to a phone. I want to dial a mobile >>> number or any other US number for that matter. >>> >>> Here is what I intent to do: >>> 1. A US phone dials to Vitelity number -- comes to my FS box >>> 2. I validate few things >>> 3. Dial another US number and connect received call to that one >>> >>> So as you said I will be charged 2 * 1.44 (since I can't terminate >>> the arrived call after validation). >>> >>> Is there any other way to the sequence I have specified above? >>> >>> Are there providers similar to Vitelity but cheaper (with >>> relatively same features)? >>> >>> Thank you. >>> >>> On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove >> > wrote: >>> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote: >>> > Hello, >>> > >>> > >>> > It would be a great help if someone can guide me. >>> > >>> > >>> > 1. I would like to first receive a call, perform certain >>> validations. >>> > (Able to do this via mod_managed application that handles call >>> from >>> > dialplan). >>> > >>> >>> That should not be a problem, I don't know your requirements so >>> can't >>> provide a full answer. >>> >>> > >>> > 2. Now, I would like to dial out to a PSTN number so that received >>> > call is connected to this new outbound number. >>> > >>> >>> This can be done and is called hairpinning. >>> >>> > >>> > How can this be done? Do I use Originate from within my .NET >>> > (mod_managed) code? >>> > >>> >>> Yes, you would be bridging the two legs like any normal call. >>> Instead of >>> going to an endpoint you're going back out over the PSTN. >>> >>> > >>> > Do I get charged for both incoming and outbound call until the >>> entire >>> > session ends? Is there a way to receive call, validate and then >>> sort >>> > of transfer and then terminate the received call so I do not get >>> > charged for both? >>> >>> That would depend on your provider but most likely yes. As for >>> terminating one end after validation that is not going to happen. >>> A leg >>> terminates to you on an agreed fee schedule. No carrier that I >>> know of >>> supports that. Now if you kept everything SIP... you could do a >>> transfer >>> after the validation. >>> >>> >>> >>> Anthony C. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Ghulam Mustafa >>> cell: +92 333.611.7681 >>> sip: cyrenity at ekiga.net >>> mail: mustafa.pk at gmail.com >>> web: cyrenity.wordpress.com >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > > | SECNAP Network Security Corporation > Certified SNORT Integrator > 2008-9 Hot Company Award Winner, World Executive Alliance > Five-Star Partner Program 2009, VARBusiness > Best in Email Security,2010: Network Products Guide > King of Spam Filters, SC Magazine 2008 > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see http://www.secnap.com/products/spammertrap/ > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/ccd28be8/attachment-0001.html From abu.4000 at gmail.com Thu Aug 26 01:29:53 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Thu, 26 Aug 2010 13:59:53 +0530 Subject: [Freeswitch-users] what is manual_calls in fifo Message-ID: Dear community , whenever I issue the command "fifo list" it shows all the fifo names along with the manual_calls actually what it is , Is it a default fifo in the freeswitch ? . and also why the status for a member always nothing even he is busy with the customer ,Is this also a bug ? $ fifo list {member_wait=nowait}user/1000 {member_wait=nowait}user/1000 Thanks in advance !!! -- *Best Regards, **Abubacker systems engineer bk systems (p) ltd** * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/7b972343/attachment.html From mike at jerris.com Thu Aug 26 01:59:57 2010 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Aug 2010 04:59:57 -0400 Subject: [Freeswitch-users] what is manual_calls in fifo In-Reply-To: References: Message-ID: On Aug 26, 2010, at 4:29 AM, Abubacker siddiq wrote: > Dear community , > whenever I issue the command "fifo list" it shows all the fifo names along with the manual_calls > actually what it is , Is it a default fifo in the freeswitch ? . it is a fake queue for tracking calls made not from the queue > and also why the status for a member always > nothing even he is busy with the customer ,Is this also a bug ? Neither are bugs. An agents status is whatever it is set to in the database. From michael.scheidell at secnap.com Thu Aug 26 02:06:29 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Thu, 26 Aug 2010 05:06:29 -0400 Subject: [Freeswitch-users] Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <6B192D76-9AAC-4C57-A005-2AF12FDB2F71@ipeva.fr> References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> <4C758109.3070002@secnap.com> <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> <6B192D76-9AAC-4C57-A005-2AF12FDB2F71@ipeva.fr> Message-ID: <4C762E95.5070206@secnap.com> Thanks. I'll give them both a try. than when I have time, I'l probably be moving everything from the old sip switch to freeswitch. On 8/26/10 3:59 AM, David Ponzone wrote: > Sure it will work, it's basically the same. > My idea was just to have a specific file with only the specific > dialplan for each direction. > But if he never needs to add anything in the dialplan, it's perhaps > overkill. > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > > /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis > ? l'intention exclusive de ses destinataires. Toute utilisation ou > diffusion non autoris?e est interdite. Tout message ?lectronique est > susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au > titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous > n'?tes pas destinataire de ce message, merci de le d?truire > imm?diatement et d'avertir l'exp?diteur./ > / > / > > > > Le 26/08/2010 ? 07:49, jesse a ?crit : > >> I feel Michael's idea should also work: If call is from X.X.X.X, then >> bridge the call with the same >> destination_number ($1) to sipxecs gateway. >> >> >> >> >> >> >> >> >> >> On Wed, Aug 25, 2010 at 4:33 PM, David Ponzone >> > wrote: >>> It is right. >>> You just transfer to another dialplan (context) in order to make it more >>> readable. >>> You'll be able to add all your rules in the provider context, >>> because only >>> calls coming from provider X.X.X.X will go into this dialplan. >>> Remember: extension name has absolutely NO importance in FS. It's just a >>> label. >>> So to sum it all: >>> if call from X.X.X.X, transfer to context provider, with the same >>> destination_number ($1). >>> In context provider, if destination_number is XXX, then bridge the >>> call to >>> sipxecs. >>> David Ponzone Direction Technique >>> email: david.ponzone at ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr - www.ipeva-studio.com >>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? >>> l'intention exclusive de ses destinataires. Toute utilisation ou >>> diffusion >>> non autoris?e est interdite. Tout message ?lectronique est susceptible >>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce >>> message s'il >>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce >>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. >>> >>> >>> >>> Le 25/08/2010 ? 22:46, Michael Scheidell a ?crit : >>> >>> this doesn't 'feel' right: >>> >>> On 8/25/10 3:17 PM, Michael Scheidell wrote: >>> >>> >>> >>> >>> >>> >>> >>> >>> if its FROM _provider, shouldn't you transfer to $1 XML sipxecs? >>> >>> and you are using 'transfer' and not 'bridge', right? >>> >>> -- >>> Michael Scheidell, CTO >>> o: 561-999-5000 >>> d: 561-948-2259 >>> ISN: 1259*1300 >>>> | SECNAP Network Security Corporation >>> >>> Certified SNORT Integrator >>> 2008-9 Hot Company Award Winner, World Executive Alliance >>> Five-Star Partner Program 2009, VARBusiness >>> Best in Email Security,2010: Network Products Guide >>> King of Spam Filters, SC Magazine 2008 >>> >>> ________________________________ >>> >>> This email has been scanned and certified safe by SpammerTrap?. >>> For Information please see http://www.secnap.com/products/spammertrap/ >>> >>> ________________________________ >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/388adeff/attachment-0001.html From jim at k4gvo.com Thu Aug 26 02:21:51 2010 From: jim at k4gvo.com (Jim) Date: Thu, 26 Aug 2010 05:21:51 -0400 Subject: [Freeswitch-users] Still can't dial gateway from ZAP phone. In-Reply-To: <5B2DD198-788E-4F82-AFBC-CF67109E5330@freeswitch.org> References: <4C730A7F.6030408@k4gvo.com> <4C752312.3040800@k4gvo.com> <5B2DD198-788E-4F82-AFBC-CF67109E5330@freeswitch.org> Message-ID: <4C76322F.60800@k4gvo.com> On 08/25/2010 10:08 AM, Brian West wrote: > This isn't valid there. > > /b > > On Aug 25, 2010, at 9:05 AM, Jim wrote: > > >> > /> >> > I agree, it's an artifact. Jim. From michael.scheidell at secnap.com Thu Aug 26 03:09:42 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Thu, 26 Aug 2010 06:09:42 -0400 Subject: [Freeswitch-users] almost there: Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> <4C758109.3070002@secnap.com> <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> Message-ID: <4C763D66.70403@secnap.com> almost there, but the ../sip_profiles ../public ../default ../dialplan directories have me confused, especially since all these files 'look the same'. (almost) I must REALLY be dense: if I just want to bridge (since the provider won't ever send me anything but the did's they are getting paid for): so: just TWO files? cd /usr/local/freeswitch/conf in ../dialplan/public/from_provider.xml I have: (do I need ..... if this in in a file? tried both ways.) and in ../sip_profiles/external/sipxecs.xml I have: (I don't need authentication/registration).. im local :-) I must really be an idiot, because now, I don't even get anything in the logs. tcpdump shows INVITES from the provider. (I added to the from_provider.xml just to see) ../log/freeswitch.log doesn't show anything after the restart stuff. and when I had it wrong, at least I logged the trys and 480 'try later' error. taking a break for breakfast.. its 6am. back in 2 hours. I want a sample setup. I dial the 301 number, come in freeswitch, is gatewayed to my sipxecs. I get audio! I get DTMF, I can internally transfer, (oh, and when I finally get this to work. how do I send this to port 5080 on sipx? so I can transfer from AA and/ or CFNA?) I don't need the outbound gateway (yet) (to complicate matters, this is the freeswitch package on pfsense. no, not the 'new one', but the one I can install via the pfsense gui. and restarts are slow) On 8/26/10 1:49 AM, jesse wrote: > I feel Michael's idea should also work: If call is from X.X.X.X, then > bridge the call with the same > destination_number ($1) to sipxecs gateway. > > > > > > > > > > On Wed, Aug 25, 2010 at 4:33 PM, David Ponzone wrote: > >> It is right. >> You just transfer to another dialplan (context) in order to make it more >> readable. >> You'll be able to add all your rules in the provider context, because only >> calls coming from provider X.X.X.X will go into this dialplan. >> Remember: extension name has absolutely NO importance in FS. It's just a >> label. >> So to sum it all: >> if call from X.X.X.X, transfer to context provider, with the same >> destination_number ($1). >> In context provider, if destination_number is XXX, then bridge the call to >> sipxecs. >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/b18ab99a/attachment.html From michael.scheidell at secnap.com Thu Aug 26 03:16:09 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Thu, 26 Aug 2010 06:16:09 -0400 Subject: [Freeswitch-users] almost there: Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <4C763D66.70403@secnap.com> References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> <4C758109.3070002@secnap.com> <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> <4C763D66.70403@secnap.com> Message-ID: <4C763EE9.5010701@secnap.com> I am an idiot. I had closed the firewall port. I just opened it. I get our AA now!!! thanks. (now, how to I make it hit port 5080 on the sipxecs?) On 8/26/10 6:09 AM, Michael Scheidell wrote: > I must really be an idiot, because now, I don't even get anything in > the logs. tcpdump shows INVITES from the provider. > (I added to the from_provider.xml just to see) > > ../log/freeswitch.log doesn't show anything after the restart stuff. > and when I had it wrong, at least I logged the -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/5a1a1960/attachment.html From david.ponzone at ipeva.fr Thu Aug 26 03:17:52 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Thu, 26 Aug 2010 12:17:52 +0200 Subject: [Freeswitch-users] almost there: Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: <4C763EE9.5010701@secnap.com> References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> <4C758109.3070002@secnap.com> <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> <4C763D66.70403@secnap.com> <4C763EE9.5010701@secnap.com> Message-ID: <48911719-F935-40A8-AA91-7E924673C664@ipeva.fr> David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 26/08/2010 ? 12:16, Michael Scheidell a ?crit : > I am an idiot. I had closed the firewall port. I just opened it. I > get our AA now!!! thanks. > (now, how to I make it hit port 5080 on the sipxecs?) > > > On 8/26/10 6:09 AM, Michael Scheidell wrote: >> >> I must really be an idiot, because now, I don't even get anything >> in the logs. tcpdump shows INVITES from the provider. >> (I added to the from_provider.xml just to see) >> >> ../log/freeswitch.log doesn't show anything after the restart >> stuff. and when I had it wrong, at least I logged the > > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > > | SECNAP Network Security Corporation > Certified SNORT Integrator > 2008-9 Hot Company Award Winner, World Executive Alliance > Five-Star Partner Program 2009, VARBusiness > Best in Email Security,2010: Network Products Guide > King of Spam Filters, SC Magazine 2008 > > This email has been scanned and certified safe by SpammerTrap?. > For Information please see http://www.secnap.com/products/spammertrap/ > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/e9186112/attachment-0001.html From abu.4000 at gmail.com Thu Aug 26 03:29:44 2010 From: abu.4000 at gmail.com (Abubacker siddiq) Date: Thu, 26 Aug 2010 15:59:44 +0530 Subject: [Freeswitch-users] what is manual_calls in fifo In-Reply-To: References: Message-ID: can u explain me better how do I set the member status in the database? On Thu, Aug 26, 2010 at 2:29 PM, Michael Jerris wrote: > > On Aug 26, 2010, at 4:29 AM, Abubacker siddiq wrote: > > > Dear community , > > whenever I issue the command "fifo list" it shows all the fifo names > along with the manual_calls > > actually what it is , Is it a default fifo in the freeswitch ? . > > it is a fake queue for tracking calls made not from the queue > > > and also why the status for a member always > > nothing even he is busy with the customer ,Is this also a bug ? > > Neither are bugs. An agents status is whatever it is set to in the > database. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Best Regards, **Abubacker systems engineer bk systems (p) ltd** * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/26d5474a/attachment.html From jim at k4gvo.com Thu Aug 26 03:47:41 2010 From: jim at k4gvo.com (Jim) Date: Thu, 26 Aug 2010 06:47:41 -0400 Subject: [Freeswitch-users] Voice mail isn't working on one extension. Message-ID: <4C76464D.4070308@k4gvo.com> I have the following connected to my FS box. A GrandStream phone, two POTS phones via the Linksys SPA2002, a soft phone installed on another system, not the FS machine and another POTS connected via an A200 port. If I call the GrandStream from one of the POTS on the SPA2002, it rings for ~30 seconds and I get a fast busy. If I call on the A200 connected phone, it rings and then nothing. It just stops ringing. If I call on the soft phone after it stops ringing I see a 404 error displayed on the phone "console". The log shows: 2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel sofia/internal/1000 at 192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 404 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000 at 192.168.2.51 Standard HANGUP, cause: NO_ROUTE_DESTINATION If I do a diff of directory/default/1001.xml and 1002.xml the only differences are the numbers and passwords. Any idea what I might want to try next? Thanks, Jim. From 12ukwn at gmail.com Thu Aug 26 04:54:52 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Thu, 26 Aug 2010 13:54:52 +0200 Subject: [Freeswitch-users] How could install PHP ESL? In-Reply-To: References: Message-ID: <20100826135452.41039a31@anubis.defcon1> On Wed, 25 Aug 2010 07:12:18 +0600, Nyamul Hassan wrote: Thank Hassan, I also think that the best solt; but shall I post the genuine output (without relocating ../src/include/*h into esl/php and tweaking them) that caused a lot of error because apparently they weren't found by the compiler, or my tweaked version output? > * enter the /libs/esl directory > * run "make" > * run "make phpmod" > > If you face error in any step above, then please output the "full output" > into > http://pastebin.freeswitch.org > > and let us know the pastebin link. > > Regards > HASSAN -- 30 day money back guarantee minus shipping, 10% restocking charge, and 7% cancellation charge. From jim at k4gvo.com Thu Aug 26 06:48:26 2010 From: jim at k4gvo.com (Jim) Date: Thu, 26 Aug 2010 09:48:26 -0400 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: <4C76464D.4070308@k4gvo.com> References: <4C76464D.4070308@k4gvo.com> Message-ID: <4C7670AA.3080508@k4gvo.com> On 08/26/2010 06:47 AM, Jim wrote: > I have the following connected to my FS box. A GrandStream phone, two > POTS phones via the Linksys SPA2002, a soft phone installed on another > system, not the FS machine and another POTS connected via an A200 port. > > If I call the GrandStream from one of the POTS on the SPA2002, it rings > for ~30 seconds and I get a fast busy. If I call on the A200 connected > phone, it rings and then nothing. It just stops ringing. If I call on > the soft phone after it stops ringing I see a 404 error displayed on the > phone "console". The log shows: > > 2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/1000 at 192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION > 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE > with: 404 > 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/1000 at 192.168.2.51 Standard HANGUP, cause: > NO_ROUTE_DESTINATION > > If I do a diff of directory/default/1001.xml and 1002.xml the only > differences are the numbers and passwords. > > Any idea what I might want to try next? > > Thanks, > Jim. > > I did change the extension from 1001 to 1010 just to see if it affected the results. It did not. Jim. From jim at k4gvo.com Thu Aug 26 07:15:25 2010 From: jim at k4gvo.com (Jim) Date: Thu, 26 Aug 2010 10:15:25 -0400 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: <4C7670AA.3080508@k4gvo.com> References: <4C76464D.4070308@k4gvo.com> <4C7670AA.3080508@k4gvo.com> Message-ID: <4C7676FD.10902@k4gvo.com> On 08/26/2010 09:48 AM, Jim wrote: > On 08/26/2010 06:47 AM, Jim wrote: > >> I have the following connected to my FS box. A GrandStream phone, two >> POTS phones via the Linksys SPA2002, a soft phone installed on another >> system, not the FS machine and another POTS connected via an A200 port. >> >> If I call the GrandStream from one of the POTS on the SPA2002, it rings >> for ~30 seconds and I get a fast busy. If I call on the A200 connected >> phone, it rings and then nothing. It just stops ringing. If I call on >> the soft phone after it stops ringing I see a 404 error displayed on the >> phone "console". The log shows: >> >> 2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel >> sofia/internal/1000 at 192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION >> 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE >> with: 404 >> 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 >> sofia/internal/1000 at 192.168.2.51 Standard HANGUP, cause: >> NO_ROUTE_DESTINATION >> >> If I do a diff of directory/default/1001.xml and 1002.xml the only >> differences are the numbers and passwords. >> >> Any idea what I might want to try next? >> >> Thanks, >> Jim. >> >> >> > I did change the extension from 1001 to 1010 just to see if it affected > the results. It did not. > > Jim. > I also did a power off M/C of the phone and looked at the settings. They are mostly factory default and I know for a fact that voice mail worked with that phone when I was running Asterisk. I did a git of the latest yesterday to pickup freetdm so I'm running the latest and greatest code, or a very close likeness. I'm thinking voicemail on FS has some bugs. I also found that when I bridged an outside line (sip) to my A200 connected phone, the phone rings, lets me answer but if I let it ring, I never get voicemail. I do get VM if I dial it from a local extension. If I bridge to a POTS phone connected to the SPA2002 it works as expected, i. e. I get VM. Jim. From peder at networkoblivion.com Thu Aug 26 07:23:03 2010 From: peder at networkoblivion.com (Peder) Date: Thu, 26 Aug 2010 09:23:03 -0500 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: <4C7670AA.3080508@k4gvo.com> References: <4C76464D.4070308@k4gvo.com> <4C7670AA.3080508@k4gvo.com> Message-ID: <04e501cb452a$31e4eae0$95aec0a0$@com> Can you send more of the debug from a call that works and one that fails? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jim Sent: Thursday, August 26, 2010 8:48 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voice mail isn't working on one extension. On 08/26/2010 06:47 AM, Jim wrote: > I have the following connected to my FS box. A GrandStream phone, two > POTS phones via the Linksys SPA2002, a soft phone installed on another > system, not the FS machine and another POTS connected via an A200 port. > > If I call the GrandStream from one of the POTS on the SPA2002, it rings > for ~30 seconds and I get a fast busy. If I call on the A200 connected > phone, it rings and then nothing. It just stops ringing. If I call on > the soft phone after it stops ringing I see a 404 error displayed on the > phone "console". The log shows: > > 2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/1000 at 192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION > 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE > with: 404 > 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/1000 at 192.168.2.51 Standard HANGUP, cause: > NO_ROUTE_DESTINATION > > If I do a diff of directory/default/1001.xml and 1002.xml the only > differences are the numbers and passwords. > > Any idea what I might want to try next? > > Thanks, > Jim. > > I did change the extension from 1001 to 1010 just to see if it affected the results. It did not. Jim. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Thu Aug 26 08:56:00 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Aug 2010 08:56:00 -0700 Subject: [Freeswitch-users] Everyone come join IRC! Message-ID: Hey all, We are having a big IRC day today so we'd love to have everyone join and help answer questions. Please join #freeswitch on irc.freenode.net. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/a9ab0de1/attachment.html From msc at freeswitch.org Thu Aug 26 09:00:47 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Aug 2010 09:00:47 -0700 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: <4C76464D.4070308@k4gvo.com> References: <4C76464D.4070308@k4gvo.com> Message-ID: Peder mentioned it already, but a complete log from start to finish would be helpful. Use pastebin.freeswitch.org and then reply to this thread with the URL to your pastebin post. -MC On Thu, Aug 26, 2010 at 3:47 AM, Jim wrote: > I have the following connected to my FS box. A GrandStream phone, two > POTS phones via the Linksys SPA2002, a soft phone installed on another > system, not the FS machine and another POTS connected via an A200 port. > > If I call the GrandStream from one of the POTS on the SPA2002, it rings > for ~30 seconds and I get a fast busy. If I call on the A200 connected > phone, it rings and then nothing. It just stops ringing. If I call on > the soft phone after it stops ringing I see a 404 error displayed on the > phone "console". The log shows: > > 2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel > sofia/internal/1000 at 192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION > 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE > with: 404 > 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/1000 at 192.168.2.51 Standard HANGUP, cause: > NO_ROUTE_DESTINATION > > If I do a diff of directory/default/1001.xml and 1002.xml the only > differences are the numbers and passwords. > > Any idea what I might want to try next? > > Thanks, > Jim. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/4728f914/attachment-0001.html From chat2jesse at gmail.com Thu Aug 26 09:15:04 2010 From: chat2jesse at gmail.com (jesse) Date: Thu, 26 Aug 2010 09:15:04 -0700 Subject: [Freeswitch-users] Lua script issue. In-Reply-To: References: Message-ID: Anyone has other advice or opinion for this issue? -jesse On Wed, Aug 25, 2010 at 11:29 AM, jesse wrote: > hmm, I do not think that is the reason. Just tried your suggestion, > the same error. > > -jesse > > On Wed, Aug 25, 2010 at 3:15 AM, Seven Du wrote: >> On Wed, Aug 25, 2010 at 7:56 AM, jesse wrote: >>> I have a bridge.lua script: >>> >>> phone1 = argv[1]; >>> phone2 = argv[2]; >>> >>> dialstring1 = "sofia/gateway/xyz.com/" .. phone1; >>> dialstring2 = "sofia/gateway/xyz.com7" .. phone2; >>> >>> session1 = freeswitch.Session(dialstring1); >>> session2 = freeswitch.Session(dialstring2, session1); >> >> According to http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.bridge >> , you may try >> session2 = freeswitch.Session(dialstring2); >> >>> freeswitch.bridge(session1, session2); >>> >>> >>> >>> freeswitch at xyz.com> luarun bridge.lua 16502222222 16503333333 >>> 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel >>> sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] >>> >>> +OK >>> >>> freeswitch at xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 >>> sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> >>> 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer >>> sofia/external/16502222222! >>> 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel >>> sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] >>> 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 >>> sofia/external/16503333333 Update Callee ID to "16503333333" >>> <16503333333> >>> 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer >>> sofia/external/16503333333! >>> 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not ready >>> 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup >>> sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] >>> 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup >>> sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] >>> 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session >>> 25 (sofia/external/16503333333) Ended >>> 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close >>> Channel sofia/external/16503333333 [CS_DESTROY] >>> 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session >>> 24 (sofia/external/16502222222) Ended >>> 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close >>> Channel sofia/external/16502222222 [CS_DESTROY] >>> >>> As you can see the call gets dropped immediately after couple rings. >>> however, it works well if i do like this: >>> >>> originate sofia/gateway/xyz.com/16502222222 >>> &bridge(sofia/gateway/xyz.com/16503333333) >>> >>> What is the reason Lua script will fail? ?any difference between the >>> two approaches? >>> >>> thanks! >>> >>> jesse >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Blog: http://www.dujinfang.com >> Proj:? http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From jim at k4gvo.com Thu Aug 26 09:31:13 2010 From: jim at k4gvo.com (Jim) Date: Thu, 26 Aug 2010 12:31:13 -0400 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: References: <4C76464D.4070308@k4gvo.com> Message-ID: <4C7696D1.6000103@k4gvo.com> On 08/26/2010 12:00 PM, Michael Collins wrote: > Peder mentioned it already, but a complete log from start to finish > would be helpful. Use pastebin.freeswitch.org > and then reply to this thread with > the URL to your pastebin post. > -MC > I don't like to waste computer space if it's not necessary. Sometimes the answer is simple and the answer doesn't require additional info. http://pastebin.freeswitch.org/13731 Note there are actually two problems demonstrated here. The first attempt is calling the GrandStream from a SPA2102 connected phone. The second is calling a POTS phone connected to the A200 from the SPA2102 and the third successful attempt is the reverse. There are notes at the beginning of the file giving approximate starting line numbers. Note, the second problem is intermittent. I though I had fixed it but I see I didn't. Thanks, Jim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/6d913a70/attachment.html From michael.scheidell at secnap.com Thu Aug 26 09:46:43 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Thu, 26 Aug 2010 12:46:43 -0400 Subject: [Freeswitch-users] one last piece? maybe? Easy question, I hope just need to proxy port 5060 to 5080 In-Reply-To: References: <4C74612C.8010009@secnap.com> <95225D68-6D7F-4D7C-ABE0-2E8C1CCC5894@ipeva.fr> <4C756C57.8060700@secnap.com> <4C758109.3070002@secnap.com> <7640436D-28E3-4C64-AEBB-8D967A5F103C@ipeva.fr> Message-ID: <4C769A73.4020003@secnap.com> That all works below. got one way audio, and no dtmf (and I know why) media handoff is wrong. system is the freewitch 1.2.3 package on pfsense (freebsd, 7.2, i386) during the invite, this gets sent: Contact: Contact Binding: URI: SIP contact address: sip:gw+secnap.com at 204.89.241.135:5060 NOW, that doesn't look all that bad, except that 1) the public ip address is 204.89.241.151 2) the private ip is 192.168.0.3 [pfsense: wan port is 204.89.241.135] [pfsense lan port is 192.168.0.1] sipx is at 192.168.0.2 see sip_profiles: ./conf/sip_profiles/external/sipx.xml: I edited vars and have I still see this in ../freeswitch.log (and I can't find that ip ANYWHERE. yes, its o=FreeSWITCH 1282812037 1282812038 IN IP4 204.89.241.135 c=IN IP4 204.89.241.135 o=FreeSWITCH 1282814358 1282814359 IN IP4 204.89.241.135 c=IN IP4 204.89.241.135 2010-08-26 12:03:22.237567 [NOTICE] switch_core.c:915 Adding 204.89.241.135/255.255.255.0 (deny) to list nat.auto 2010-08-26 12:03:22.237607 [NOTICE] switch_core.c:934 Adding 204.89.241.135/255.255.255.0 (allow) to list localnet.auto I have verified the OUTBOUND natting works: (im assuming when I set local_ip_ to 192.168.0.3 that it actually did it. telnet -s 192.168.0.3 www.whatismyip.org 80 Trying 98.207.226.113... Connected to www.whatismyip.org. Escape character is '^]'. GET / HTTP/1.0 HTTP/1.0 200 OK Content-Type: text/plain 204.89.241.151Connection closed by foreign host. wireshark dumps confirm it: Message Header Via: SIP/2.0/UDP 192.168..0.3;rport=5060;branch=z9hG4bKXQQUvXFZDUymB;received=192.168..0.3 Transport: UDP -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/cbef474c/attachment.html From jim at k4gvo.com Thu Aug 26 09:50:42 2010 From: jim at k4gvo.com (Jim) Date: Thu, 26 Aug 2010 12:50:42 -0400 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: <4C7696D1.6000103@k4gvo.com> References: <4C76464D.4070308@k4gvo.com> <4C7696D1.6000103@k4gvo.com> Message-ID: <4C769B62.2050406@k4gvo.com> On 08/26/2010 12:31 PM, Jim wrote: > On 08/26/2010 12:00 PM, Michael Collins wrote: >> Peder mentioned it already, but a complete log from start to finish >> would be helpful. Use pastebin.freeswitch.org >> and then reply to this thread with >> the URL to your pastebin post. >> -MC >> > I don't like to waste computer space if it's not necessary. Sometimes > the answer is simple and the answer doesn't require additional info. > > http://pastebin.freeswitch.org/13731 > > Note there are actually two problems demonstrated here. The first > attempt is calling the GrandStream from a SPA2102 connected phone. > The second is calling a POTS phone connected to the A200 from the > SPA2102 and the third successful attempt is the reverse. > There are notes at the beginning of the file giving approximate > starting line numbers. > > Note, the second problem is intermittent. I though I had fixed it but > I see I didn't. > > Thanks, > Jim. Not that it make a difference but I think that's a SPA2002. Jim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/3d9ee37e/attachment.html From peter.waldheim at framesoft.com Thu Aug 26 03:32:13 2010 From: peter.waldheim at framesoft.com (Peter Waldheim) Date: Thu, 26 Aug 2010 10:32:13 +0000 Subject: [Freeswitch-users] External caller-number not shown on forward to external number Message-ID: Hi all, In a diaplan I'm forwarding a call to an external (PRI) number when it fails to reach my SIP-Client: Then in that dialplan: So far this works just fine but one thing: the external party only sees the original caller-id if it is an internal one. If the caller was external, I only see the default caller-number of my installation. Ideally I'd like to always see the original caller id - especially if it is an external number. Can anybody please give me a hint whether I'm missing something or this might be some call-spoof protection in freeswitch or openzap or libpri or my PRI provider? Thanks a lot Peter ________________________________ Framesoft AG Software Applications Sumpfstrasse 15 6301 Zug Switzerland Handelsregister des Kantons Zug: CH-170.3.022.876-2 Pr?sident & Vorsitzender der Gesch?ftsleitung: Toralf Dittmann Framesoft AG Software Applications Reuterweg 49 D-60323 Frankfurt am Main Germany HRB 34142 Vorsitzender des Aufsichtsrates: Toralf Dittmann Vorstand: Jens Saarholz Framesoft Ltd. Business Address: 60 Lombard Street London EC3V9EA, UK Registered Office: Hackwood Secretaries Limited 1 silk Street London EC2Y 8HQ, UK Company Number: 4055017 Directors: Toralf Dittmann, Volker Weber Secretary: Volker Weber Confidentiality Notice: The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail and delete this message. Thank you for your cooperation. From locutis at sect001.net Thu Aug 26 06:04:16 2010 From: locutis at sect001.net (Locutis of Borg) Date: Thu, 26 Aug 2010 09:04:16 -0400 Subject: [Freeswitch-users] FreeSWITCH Sofia DNS Cache SIP/SDP INVITE Auth Fail Message-ID: Hi, I have been struggling with failed outbound calls to my ITSP. Seems that my FreeSWITCH does not know who to send SIP/SDP INVITE packets to. FS has the DNS cached and uses that to send the first INVITE, then it gets expired and refreshes it to another IP, just in time for the second INVITE, causing an AUTHENTICATION failure. The packet capture would only show the 1st INVITE SIP to one resolved IP. Then, the 2nd INVITE SIP/SDP to a different resolved IP. FS is on a DMZ. I have extensively tested IPTABLES, NAT, and even rebuilt FS from the latest GIT 2 days ago. In the ext. SIP profile, if I use hostname for proxy, initial calls fail, but on redial, they complete. Now, in the same SIP profile, I use the IP address, call are OK. (for both, REG is false) A reasonable solution seems to be to use the IP in the ext. SIP profile. But, is that the best? Any experience with this kind of issue? Any suggestion would be greatly appreciated. Thank you, Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/453517a6/attachment.html From pipiwei03 at gmail.com Thu Aug 26 08:06:05 2010 From: pipiwei03 at gmail.com (pipiwei03) Date: Thu, 26 Aug 2010 23:06:05 +0800 Subject: [Freeswitch-users] Sending fax problem Message-ID: hello, I can receive and send a fax in php already. Now, I want to set a specific fxo port, not the whole gateway, to send a fax. What configuration should I set? The environment is described as follow: a freeswitch server based on winxp. server ip: 192.168.1.1 a WellGate 2644 gateway which has 4 fxo and 4 fxs. each fxs/fxo port can register an sip account to freeswitch gateway ip: 192.168.1.2 i config my gateway as below in freeswitch: the gateway register 2 user as below in my php file, i can send the fax via fxo1 pstn line (use event_socket_request ) and the command is "api originate sofia/gateway/faxgw/". $dest_phone_number. " &txfax(tmp.tif)" my problem is 1. How to send fax via fxo2 pstn line Should i bridge to user "pstn_fxo2" first and then dial number? I try to use "api originate sofia/internal/pstn_fxo2 ".$dest_phone_number. " &txfax(tmp.tif)" but it didnt work 2. In config The "username" and "password" seem no effect. I just set the "realm" to my gateway ip and it works. And the "register" i dont know it too. The document didnt describe it clearly, can u explain it for me Thank you sincerely. Joy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/2625204d/attachment.html From mike at jerris.com Thu Aug 26 09:54:02 2010 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Aug 2010 12:54:02 -0400 Subject: [Freeswitch-users] what is manual_calls in fifo In-Reply-To: References: Message-ID: <598A9A30-1397-4569-9411-6259DCA0FDA2@jerris.com> http://www.w3schools.com/sql/sql_update.asp On Aug 26, 2010, at 6:29 AM, Abubacker siddiq wrote: > can u explain me better how do I set the member status in the database? > > On Thu, Aug 26, 2010 at 2:29 PM, Michael Jerris wrote: > > On Aug 26, 2010, at 4:29 AM, Abubacker siddiq wrote: > > > Dear community , > > whenever I issue the command "fifo list" it shows all the fifo names along with the manual_calls > > actually what it is , Is it a default fifo in the freeswitch ? . > > it is a fake queue for tracking calls made not from the queue > > > and also why the status for a member always > > nothing even he is busy with the customer ,Is this also a bug ? > > Neither are bugs. An agents status is whatever it is set to in the database. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/0dcdcbeb/attachment.html From q4innovation at gmail.com Thu Aug 26 10:02:20 2010 From: q4innovation at gmail.com (sdemers@q4innovation.com) Date: Thu, 26 Aug 2010 13:02:20 -0400 Subject: [Freeswitch-users] Cause IE in Progress Message (Sangoma, LibPRI) Message-ID: Hi everyone, I'm trying to catch the CAUSE IE into the ISDN NI2 Progress Message with Sangoma A108 card using LibPRI. Based on the Log bellow, the OZMOD_libpri catch the CAUSE IE, but at the event_socket, there is no element with that specification. This is very important because the most carrier doesn't send (90% of the time) the SIT tone UNALLOCATED, and also return NORMAL_CLEARING cause in the disconnect. They send a progress message with an Unallocated (unassigned) Cause IE and play a message on the voice path for a end user. When you have a machine on the other side it's not easy to catch. I've take a look into the ozmod_libpri.c and I didn't found any code related to LPWRAP_PRI_EVENT_PROGRESS. The print-out come probably from LPWRAP_PRI_EVENT_ANY (on_anything) function. Is there a way to push it to the event_socket remote application? Thanks for your answer. ---------------------------------------------------------------------- 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 4 to (but not including) 5 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Protocol Discriminator: Q.931 (8) len=13 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Call Ref: len= 2 (reference 3/0x3) (Terminator) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Message type: PROGRESS (3) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < [08 02 80 81] *2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0)* *2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]* 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < [1e 02 80 88] 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] *2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Processing IE 8 (cs0, Cause)* 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Processing IE 30 (cs0, Progress Indicator) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 Sending Receiver Ready (6) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 > [ 02 01 01 0c ] 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 > Supervisory frame: 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 > SAPI: 00 C/R: 1 EA: 0 > TEI: 000 EA: 1 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 > Zero: 0 S: 0 01: 1 [ RR (receive ready) ] > N(R): 006 P/F: 0 > 0 bytes of data 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Restarting T203 timer 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:979 Caught Event span 1 17 (PROGRESS) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/6cdb7217/attachment.html From jim at k4gvo.com Thu Aug 26 10:07:59 2010 From: jim at k4gvo.com (Jim) Date: Thu, 26 Aug 2010 13:07:59 -0400 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: <4C769B62.2050406@k4gvo.com> References: <4C76464D.4070308@k4gvo.com> <4C7696D1.6000103@k4gvo.com> <4C769B62.2050406@k4gvo.com> Message-ID: <4C769F6F.5060706@k4gvo.com> On 08/26/2010 12:50 PM, Jim wrote: > On 08/26/2010 12:31 PM, Jim wrote: >> On 08/26/2010 12:00 PM, Michael Collins wrote: >>> Peder mentioned it already, but a complete log from start to finish >>> would be helpful. Use pastebin.freeswitch.org >>> and then reply to this thread with >>> the URL to your pastebin post. >>> -MC >>> >> I don't like to waste computer space if it's not necessary. >> Sometimes the answer is simple and the answer doesn't require >> additional info. >> >> http://pastebin.freeswitch.org/13731 >> >> Note there are actually two problems demonstrated here. The first >> attempt is calling the GrandStream from a SPA2102 connected phone. >> The second is calling a POTS phone connected to the A200 from the >> SPA2102 and the third successful attempt is the reverse. >> There are notes at the beginning of the file giving approximate >> starting line numbers. >> >> Note, the second problem is intermittent. I though I had fixed it >> but I see I didn't. >> >> Thanks, >> Jim. > Not that it make a difference but I think that's a SPA2002. > > Jim. > Looks like the line numbers changed from my vi session to pastebin. Ignore the notes at the top. Jim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/4801356a/attachment-0001.html From jim at k4gvo.com Thu Aug 26 10:37:00 2010 From: jim at k4gvo.com (Jim) Date: Thu, 26 Aug 2010 13:37:00 -0400 Subject: [Freeswitch-users] New wiki page Message-ID: <4C76A63C.7010506@k4gvo.com> For what it's worth, I documented my experience with getting a Sangoma A200 card working on FreeSWITCH. I have no idea if it's the right way but it works sort of. http://wiki.freeswitch.org/wiki/Sangoma_A200 Jim. From mike at jerris.com Thu Aug 26 10:41:43 2010 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Aug 2010 13:41:43 -0400 Subject: [Freeswitch-users] FreeSWITCH Sofia DNS Cache SIP/SDP INVITE Auth Fail In-Reply-To: References: Message-ID: The Borg have many technologies, apparently sip nat traversal is not one of them. Defense against assimilation: 1. Provide open sip server for communications with collective 2. Place behind 2-wire router 3. Laugh at their inability to traverse NAT 4. Repeat On Aug 26, 2010, at 9:04 AM, Locutis of Borg wrote: > Hi, > > I have been struggling with failed outbound calls to my ITSP. > > Seems that my FreeSWITCH does not know who to send SIP/SDP INVITE packets to. FS has the DNS cached and uses that to send the first INVITE, then it gets expired and refreshes it to another IP, just in time for the second INVITE, causing an AUTHENTICATION failure. > > The packet capture would only show the 1st INVITE SIP to one resolved IP. Then, the 2nd INVITE SIP/SDP to a different resolved IP. > > FS is on a DMZ. I have extensively tested IPTABLES, NAT, and even rebuilt FS from the latest GIT 2 days ago. In the ext. SIP profile, if I use hostname for proxy, initial calls fail, but on redial, they complete. Now, in the same SIP profile, I use the IP address, call are OK. (for both, REG is false) > > A reasonable solution seems to be to use the IP in the ext. SIP profile. But, is that the best? > > Any experience with this kind of issue? Any suggestion would be greatly appreciated. > > Thank you, > Cheers > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tgraziano at myitdepartment.net Thu Aug 26 10:50:07 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Thu, 26 Aug 2010 13:50:07 -0400 Subject: [Freeswitch-users] FreeSWITCH Sofia DNS Cache SIP/SDP INVITE Auth Fail In-Reply-To: References: Message-ID: put two wire in bridged mode and use superior assimilated firewall technology in lieu of striking rocks together to scare the packets into alignment... On Thu, Aug 26, 2010 at 1:41 PM, Michael Jerris wrote: > The Borg have many technologies, apparently sip nat traversal is not one of > them. > > Defense against assimilation: > > 1. Provide open sip server for communications with collective > 2. Place behind 2-wire router > 3. Laugh at their inability to traverse NAT > 4. Repeat > > > On Aug 26, 2010, at 9:04 AM, Locutis of Borg wrote: > > > Hi, > > > > I have been struggling with failed outbound calls to my ITSP. > > > > Seems that my FreeSWITCH does not know who to send SIP/SDP INVITE packets > to. FS has the DNS cached and uses that to send the first INVITE, then it > gets expired and refreshes it to another IP, just in time for the second > INVITE, causing an AUTHENTICATION failure. > > > > The packet capture would only show the 1st INVITE SIP to one resolved IP. > Then, the 2nd INVITE SIP/SDP to a different resolved IP. > > > > FS is on a DMZ. I have extensively tested IPTABLES, NAT, and even > rebuilt FS from the latest GIT 2 days ago. In the ext. SIP profile, if I > use hostname for proxy, initial calls fail, but on redial, they complete. > Now, in the same SIP profile, I use the IP address, call are OK. (for > both, REG is false) > > > > A reasonable solution seems to be to use the IP in the ext. SIP profile. > But, is that the best? > > > > Any experience with this kind of issue? Any suggestion would be greatly > appreciated. > > > > Thank you, > > Cheers > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/63aa79ee/attachment.html From brian at freeswitch.org Thu Aug 26 10:56:22 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Aug 2010 12:56:22 -0500 Subject: [Freeswitch-users] FreeSWITCH Sofia DNS Cache SIP/SDP INVITE Auth Fail In-Reply-To: References: Message-ID: <600AC4F0-1EAD-4303-89CD-FA8A68B29194@freeswitch.org> We are Dyslexic of Borg. You will have your ass laminated. /b On Aug 26, 2010, at 12:50 PM, Tony Graziano wrote: > put two wire in bridged mode and use superior assimilated firewall technology in lieu of striking rocks together to scare the packets into alignment... > > On Thu, Aug 26, 2010 at 1:41 PM, Michael Jerris wrote: > The Borg have many technologies, apparently sip nat traversal is not one of them. > > Defense against assimilation: > > 1. Provide open sip server for communications with collective > 2. Place behind 2-wire router > 3. Laugh at their inability to traverse NAT > 4. Repeat > From anthony.minessale at gmail.com Thu Aug 26 12:02:50 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Aug 2010 14:02:50 -0500 Subject: [Freeswitch-users] what is manual_calls in fifo In-Reply-To: References: Message-ID: manual calls is a special queue that shows any tracked outbound calls made by an agent who is part of the outbound dialing queue. It should be ignored unless you use the fifo_track_call application On Thu, Aug 26, 2010 at 3:29 AM, Abubacker siddiq wrote: > Dear community , > whenever I issue the command "fifo list" it shows all the fifo names along > with the manual_calls > actually what it is , Is it a default fifo in the freeswitch ? . and also > why the status for a member always > nothing even he is busy with the customer ,Is this also a bug ? > > > $ fifo list > > ? waiting_count="0" importance="0" outbound_per_cycle="1" > outbound_priority="5" outbound_strategy="ringall"> > ??? > ????? outbound-call-count="0" outbound-fail-count="4" taking-calls="1" status="" > outbound-call-total-count="0" outbound-fail-total-count="4" > logged-on-since="2010-08-26 11:57:46" manual-calls-out-count="0" > manual-calls-in-count="0" manual-calls-out-total-count="0" > manual-calls-in-total-count="0" ring-count="0" start-time="never" > stop-time="never" next-available="2010-08-26 > 12:26:23">{member_wait=nowait}user/1000 > ??? > ??? > ??? > ??? > ? > ? waiting_count="0" importance="0" outbound_per_cycle="0" > outbound_priority="5" outbound_strategy="ringall"> > ??? > ????? outbound-call-count="0" outbound-fail-count="4" taking-calls="1" status="" > outbound-call-total-count="0" outbound-fail-total-count="4" > logged-on-since="2010-08-26 11:57:46" manual-calls-out-count="0" > manual-calls-in-count="0" manual-calls-out-total-count="0" > manual-calls-in-total-count="0" next-available="2010-08-26 > 12:26:23">{member_wait=nowait}user/1000 > ??? > ??? > ??? > ??? > ? > > > > Thanks in advance !!! > -- > Best Regards, > Abubacker > systems engineer > bk systems (p) ltd > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From dswardstrom at remotelink.com Thu Aug 26 13:04:21 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Thu, 26 Aug 2010 13:04:21 -0700 (PDT) Subject: [Freeswitch-users] Misc. Dialplan Tools say types Message-ID: <1282853061697-5467035.post@n2.nabble.com> I tried using the type "TELEPHONE_NUMBER" and got a failure message: session.execute("say", "en TELEPHONE_NUMBER pronounced " + myani]); mod_say_en.c:481 Unknown Say type=[9] There appears to be no code to support TELEPHONE_NUMBER in mod_say_en.c. Note: I looked. I have not done a code refresh since before ClueCon. Has it been added or is this part of code from another language. Actually, it seems that it may be in French (mod_say_fr.c). -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Misc-Dialplan-Tools-say-types-tp5467035p5467035.html Sent from the freeswitch-users mailing list archive at Nabble.com. From dswardstrom at remotelink.com Thu Aug 26 13:07:22 2010 From: dswardstrom at remotelink.com (David Swardstrom) Date: Thu, 26 Aug 2010 13:07:22 -0700 (PDT) Subject: [Freeswitch-users] Using streamFile to steam from memory Message-ID: <1282853242142-5467050.post@n2.nabble.com> I would like to be able to extract a short voice segment from a Database and stream it directly from memory using JavaScript. Is there a version of streamFile which would do this? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Using-streamFile-to-steam-from-memory-tp5467050p5467050.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Aug 26 13:15:54 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Aug 2010 15:15:54 -0500 Subject: [Freeswitch-users] Using streamFile to steam from memory In-Reply-To: <1282853242142-5467050.post@n2.nabble.com> References: <1282853242142-5467050.post@n2.nabble.com> Message-ID: <704524BB-0F81-43A7-BF66-0876A8E2AD71@freeswitch.org> None exists. You could write one in C possibly. /b On Aug 26, 2010, at 3:07 PM, David Swardstrom wrote: > > I would like to be able to extract a short voice segment from a Database and > stream it directly from memory > using JavaScript. Is there a version of streamFile which would do this? From brian at freeswitch.org Thu Aug 26 13:16:31 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Aug 2010 15:16:31 -0500 Subject: [Freeswitch-users] Misc. Dialplan Tools say types In-Reply-To: <1282853061697-5467035.post@n2.nabble.com> References: <1282853061697-5467035.post@n2.nabble.com> Message-ID: Could possibly be something that needs to be added to mod_say_en.c /b On Aug 26, 2010, at 3:04 PM, David Swardstrom wrote: > > There appears to be no code to support TELEPHONE_NUMBER in mod_say_en.c. > Note: I looked. I have not done a code refresh since before ClueCon. > Has it been added or is this part of code from another language. > Actually, it seems that it may be in French (mod_say_fr.c). From msc at freeswitch.org Thu Aug 26 13:19:27 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Aug 2010 13:19:27 -0700 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: <4C7696D1.6000103@k4gvo.com> References: <4C76464D.4070308@k4gvo.com> <4C7696D1.6000103@k4gvo.com> Message-ID: On Thu, Aug 26, 2010 at 9:31 AM, Jim wrote: > On 08/26/2010 12:00 PM, Michael Collins wrote: > > Peder mentioned it already, but a complete log from start to finish would > be helpful. Use pastebin.freeswitch.org and then reply to this thread with > the URL to your pastebin post. > -MC > > I don't like to waste computer space if it's not necessary. Sometimes the > answer is simple and the answer doesn't require additional info. > > http://pastebin.freeswitch.org/13731 > > Note there are actually two problems demonstrated here. The first attempt > is calling the GrandStream from a SPA2102 connected phone. The second is > calling a POTS phone connected to the A200 from the SPA2102 and the third > successful attempt is the reverse. > There are notes at the beginning of the file giving approximate starting > line numbers. > #1 From what I can see, the GS is doing a redirect. Look at line 226 of the pastebin down to 240. Do you have call forwarding enabled on your GS? You may want to enable a SIP trace to see the exact SIP packet from the GS to see what it's doing. It is forwarding to destination number of "service" and there isn't an extension for that in your dialplan. (See lines leading up to 409 - "service" doesn't match anything until it gets to enum, where it eventually fails altogether at line 458.) #2 Does the phone connected to port 1 on the A200 even ring? It's interesting that there is a 9 second gap (between pb lines 732 and 733) which is where I would assume the ringing was taking place. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/f5092609/attachment.html From b_ball_henry at hotmail.com Thu Aug 26 13:21:17 2010 From: b_ball_henry at hotmail.com (Henry Huang) Date: Fri, 27 Aug 2010 04:21:17 +0800 Subject: [Freeswitch-users] Enterprise setup registration question Message-ID: Hi: I have followed wiki: http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS To deploy OpenSIPS in front of FreeSWITCH. I have played with the load balancer part before, so I know how that works. I have now been able to register through the OpenSIPS to FreeSWITCH box via the dispatcher module of OpenSIPS. But what happened is that the registration IP is the IP of the OpenSIPS server, not the end user's IP address. I am wondering how would we process incoming call for the registered user? Obviously the call will be routed back to the OpenSIPS server with an invite, which gets processed by the load balance module according to the script. So what I am asking is that is this a logic flaw in the original script? or did I misunderstood it? If it is a flaw, does any one know how to fix the senario? Because I am not that good with OpenSIPS to fix it. Thanks, Henry Huang Unified Communication System R&D US: +1 (626) 606-3306 ??(Taiwan): +886 933847619 Chat Google Talk: red_rain_seven at gmail.com Skype: unicsolution MSN: b_ball_henry at hotmail.com Contact Me [image: Linkedin][image: Facebook] [image: Twitter] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/13aa646e/attachment.html From michael.scheidell at secnap.com Thu Aug 26 14:47:07 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Thu, 26 Aug 2010 17:47:07 -0400 Subject: [Freeswitch-users] issue with media on two internal gateways Message-ID: <4C76E0DB.3030408@secnap.com> I don't want FS1 who talks to FS2 to try to use the PUBLIC rtp port. but I can't seem to see how. FS1: 192.168.0.3 (public ip 204.89.241.135) listeing for trunk calls on 5060 FS2: 192.168.0.2 (public ip 204.89.241.150) listening for trunk calls on 5080 FS1 -> FS2 in Inivite: (then it tries to bind media to public ip, i get one way audio, and no DTMF) Contact: Contact Binding: URI: SIP contact address: sip:gw+secnap.com at 204.89.241.135:5060 (204.89.241.135 is the public ip, but these two boxes are internal) FS2 listens on port 5080, FS1 listens on port 5060. FS2-> FS1 (what is fair is FAIR..) Contact: Contact Binding: URI: SIP contact address: sip:gw+secnap.com at 204.89.241.150:5080 if tried so far: put outbound-proxy in secnap.com.xml in sofia.conf.xml (noop) in acl.conf.xml: no.. internal xml: --> --> in external.xml also making lots of calls from my cell phone.. watching traffic. -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/e6b6509c/attachment.html From michael.scheidell at secnap.com Thu Aug 26 14:58:15 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Thu, 26 Aug 2010 17:58:15 -0400 Subject: [Freeswitch-users] more info: CONTACT header via fs_cli: was issue with media on two internal gateways In-Reply-To: <4C76E0DB.3030408@secnap.com> References: <4C76E0DB.3030408@secnap.com> Message-ID: <4C76E377.8070103@secnap.com> More info: Two things: found the great diagnostics information at wiki don't mention you need to use fs_cli for this.. anyway, this one says: sofia status gateway secnap.com ================================================================================================= Name secnap.com Scheme Digest Realm 10.72.0.2:5080 Username username Password yes From Contact (note: this is just wrong. the public ip is 204.89.241.151.. but still.. I want secnap.com, internal, to Contact me back at 192.168.0.2 Exten username To sip:username at 192.168.0.2:5080 Proxy sip:192.168.0.2:5080 Context public Expires 600 Freq 600 Ping 0 PingFreq 0 State NOREG Status UP CallsIN 0 CallsOUT 0 ================================================================================================= On 8/26/10 5:47 PM, Michael Scheidell wrote: > I don't want FS1 who talks to FS2 to try to use the PUBLIC rtp port. > but I can't seem to see how. > -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/bc7c9d09/attachment-0001.html From brian at freeswitch.org Thu Aug 26 15:04:34 2010 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Aug 2010 17:04:34 -0500 Subject: [Freeswitch-users] more info: CONTACT header via fs_cli: was issue with media on two internal gateways In-Reply-To: <4C76E377.8070103@secnap.com> References: <4C76E0DB.3030408@secnap.com> <4C76E377.8070103@secnap.com> Message-ID: <27E7F516-EA13-449E-97FB-8F59729909AD@freeswitch.org> Then you need to preface the ip's in ext-sip-ip and ext-rtp-ip with autonat: /b On Aug 26, 2010, at 4:58 PM, Michael Scheidell wrote: > More info: > > Two things: > > found the great diagnostics information at > > wiki don't mention you need to use fs_cli for this.. > > anyway, this one says: > > sofia status gateway secnap.com > ================================================================================================= > Name secnap.com > Scheme Digest > Realm 10.72.0.2:5080 > Username username > Password yes > From > Contact > > (note: this is just wrong. the public ip is 204.89.241.151.. but still.. I want secnap.com, internal, to Contact me back at 192.168.0.2 > > > Exten username > To sip:username at 192.168.0.2:5080 > Proxy sip:192.168.0.2:5080 > Context public > Expires 600 > Freq 600 > Ping 0 > PingFreq 0 > State NOREG > Status UP > CallsIN 0 > CallsOUT 0 > ================================================================================================= > From michael.scheidell at secnap.com Thu Aug 26 15:18:31 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Thu, 26 Aug 2010 18:18:31 -0400 Subject: [Freeswitch-users] more info: CONTACT header via fs_cli: was issue with media on two internal gateways In-Reply-To: <27E7F516-EA13-449E-97FB-8F59729909AD@freeswitch.org> References: <4C76E0DB.3030408@secnap.com> <4C76E377.8070103@secnap.com> <27E7F516-EA13-449E-97FB-8F59729909AD@freeswitch.org> Message-ID: <4C76E837.9000402@secnap.com> pretend I am stupid (this is my 6th phone system, 4th VOIP based) still trying to find my way around: Freeswitch is BY FAR, the most feature rich of anything I have ever seen. (and if natting worked, would it get the REAL public address of 204.89.241.151? for outbound? some background, this is the freeswitch. package (not the NEW one on your web site with fusionpbx) but the one that you can install on pfsense. yes, the WAN address is 204.89.241.135. I assume some process is doing an outbound call to something to try to find the public ip. But still for calls from an internal freeswich, to an internal sip switch. internal.xml and xml was this: was this: --> are you saying: (what, ?) --> On 8/26/10 6:04 PM, Brian West wrote: > Then you need to preface the ip's in ext-sip-ip and ext-rtp-ip with autonat: > -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/bf1111d6/attachment.html From michael.scheidell at secnap.com Thu Aug 26 15:33:16 2010 From: michael.scheidell at secnap.com (Michael Scheidell) Date: Thu, 26 Aug 2010 18:33:16 -0400 Subject: [Freeswitch-users] found it. CONTACT header via fs_cli: was issue with media on two internal gateways In-Reply-To: <27E7F516-EA13-449E-97FB-8F59729909AD@freeswitch.org> References: <4C76E0DB.3030408@secnap.com> <4C76E377.8070103@secnap.com> <27E7F516-EA13-449E-97FB-8F59729909AD@freeswitch.org> Message-ID: <4C76EBAC.9070000@secnap.com> *found it.* increadable diagnostics. sofia status gateway secnap.com ================================================================================================= Name secnap.com Scheme Digest Realm 192.168.0.2:5080 Username username Password yes From Contact On 8/26/10 6:04 PM, Brian West wrote: > Then you need to preface the ip's in ext-sip-ip and ext-rtp-ip with autonat: > > /b > -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/8da39a0a/attachment.html From locutis at sect001.net Thu Aug 26 18:21:25 2010 From: locutis at sect001.net (Locutis of Borg) Date: Thu, 26 Aug 2010 21:21:25 -0400 Subject: [Freeswitch-users] FreeSWITCH Sofia DNS Cache SIP/SDP INVITE Auth Fail In-Reply-To: <600AC4F0-1EAD-4303-89CD-FA8A68B29194@freeswitch.org> References: <600AC4F0-1EAD-4303-89CD-FA8A68B29194@freeswitch.org> Message-ID: You assimilated too much * code into FS. No, but seriously. I may have fixed it without understanding that it was a NAT issue. Ditched the gateway. Went to IP auth. Seems to be working ok. Tried hostname in the dialplan and it seems to resolve ok. I know you guys are probably sick-to-death of explaining NAT. So, don't worry about it. On Thu, Aug 26, 2010 at 1:56 PM, Brian West wrote: > We are Dyslexic of Borg. You will have your ass laminated. > > /b > > On Aug 26, 2010, at 12:50 PM, Tony Graziano wrote: > > > put two wire in bridged mode and use superior assimilated firewall > technology in lieu of striking rocks together to scare the packets into > alignment... > > > > On Thu, Aug 26, 2010 at 1:41 PM, Michael Jerris wrote: > > The Borg have many technologies, apparently sip nat traversal is not one > of them. > > > > Defense against assimilation: > > > > 1. Provide open sip server for communications with collective > > 2. Place behind 2-wire router > > 3. Laugh at their inability to traverse NAT > > 4. Repeat > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100826/717cb4da/attachment.html From Victor at isptelecom.net Thu Aug 26 20:38:10 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Thu, 26 Aug 2010 23:38:10 -0400 Subject: [Freeswitch-users] How to set .wav file as a voice-mail greeting? In-Reply-To: <4C7412F1.9000509@isptelecom.net> References: <4C7412F1.9000509@isptelecom.net> Message-ID: <4C773322.7030508@isptelecom.net> Quick question: Is there an elegant way of using pre-recorder .wav file as a greeting for voice mail box of a user? My feeling is that No unless editing DB. But I'd be happy to learn I'm mistaken :-) Thank you, Victor From fs-list at communicatefreely.net Thu Aug 26 20:50:02 2010 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 26 Aug 2010 23:50:02 -0400 Subject: [Freeswitch-users] any disaster lurking in this over all plan? In-Reply-To: References: Message-ID: <4C7735EA.5010408@communicatefreely.net> For what it's worth, I have about 250 Aastra phones in the field, at least of every model. I haven't had any issues with them (except the wireless ones). They seem to work just fine with Freeswitch. The features I have working are: Calls (in and out) Distinctive ringing and auto-answer Blind and attended transfer Message waiting indicator Busy lamps with ringing and in-use status All our endpoints our out across the open internet, not in a tidy office setup like you have, so I think yours will be easier. Some things to be wary of: How are you going to "roll-over" to the VoIP DID? Most carriers I know of will let you put multiple lines in a rollover group, but will they redirect to another carrier's DID when all the lines are full? It sounds good on paper to go low risk and stick with POTS lines, but I have found them to be more trouble than they are worth. Some sort of digital interface is a huge step up. You also get proper signaling, which is a bit limited on POTS. When you start sending voice traffic across your DSL connection, make sure that you have some sort of QoS mechanism in place. I can't stress how important this is. Your DSL connection will be a bottleneck, so your router needs to ensure that all those little voice packets get out right away, ahead of the big data packets. If you don't, you will get choppy audio whenever someone sends an e-mail with a big attachment. Easy ways to do this in a small setup: Linksys WRT54GL with Tomato flashed onto it and some rules set up to match voice traffic (IP range and port seems to be more reliable than the L7 filters) For DSL, I'm a really big fan of the Netopia 3300 series, like the 3346N-ENT for DSL. It's a combination ADSL2+ multimode modem, and a pretty decent router. If you toggle on the prioritization feature, it will make everything fit down the DSL link. Since it knows what the line speed is, it will always work without any adjustment. It will also use the DSCP/TOS tags to prioritize, so you don't have to configure anything. These are about $150 here in Canada, and have a lot of features. You will probably still stay up all weekend, especially if this is your first project, but it will be worth it when you get everything working well. Good luck! Robust Process wrote: >>The only disaster I see is the Aastra in the mix. Thats just my > personal preference. > > thanks for the response Brian, > > since I've never had my hands on any of the IP phones, what feature, > lack of feature, quality issue, or configuration problems have you had > with the Aastra's compared to other IP phones? > > Orin > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jonas.gauffin at gmail.com Thu Aug 26 23:26:37 2010 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 27 Aug 2010 08:26:37 +0200 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION Message-ID: Hello, Why do this call result in INCOMPATIBLE_DESTINATION? http://pastebin.freeswitch.org/13736 Regards, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/f00f8d19/attachment.html From steveayre at gmail.com Thu Aug 26 23:43:50 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 27 Aug 2010 07:43:50 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: Ordinarily it'd mean there was a problem with the codecs (e.g. needing to use an unsupported codec or transcode a codec that only works as a passthrough one). Looks like it should have gone through with PCMA (8) though. Can you repeat the call with sip trace on? Perhaps the incompatible destination comes from an endpoint. 'sofia profile siptrace on' from the CLI, replace on with off to turn it off again. -Steve On 27 August 2010 07:26, Jonas Gauffin wrote: > Hello, > > Why do this call result in INCOMPATIBLE_DESTINATION? > > http://pastebin.freeswitch.org/13736 > > Regards, > Jonas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/add1818c/attachment.html From jonas.gauffin at gmail.com Fri Aug 27 00:03:29 2010 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 27 Aug 2010 09:03:29 +0200 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: It doesn't happen every time and it's on a production system with a bit of volume. therefore a bit hard to get SIP traces. I'll try if Anthony really needs them. FS do say this: 2010-08-27 07:14:10.758750 [DEBUG] switch_ivr_originate.c:3111sofia/external/ 0700123456 at 212.151.Y.Y:5060 Media Establishment Failed In which RFC are codec names defined? rfc3551 defines "G729" but no "G.729A" or "G.729B". But as you say, shouldn't FS use PCMA in any case? On Fri, Aug 27, 2010 at 8:43 AM, Steven Ayre wrote: > Ordinarily it'd mean there was a problem with the codecs (e.g. needing to > use an unsupported codec or transcode a codec that only works as a > passthrough one). > > Looks like it should have gone through with PCMA (8) though. Can you repeat > the call with sip trace on? Perhaps the incompatible destination comes from > an endpoint. > > 'sofia profile siptrace on' from the CLI, replace on with off > to turn it off again. > > -Steve > > > > On 27 August 2010 07:26, Jonas Gauffin wrote: > >> Hello, >> >> Why do this call result in INCOMPATIBLE_DESTINATION? >> >> http://pastebin.freeswitch.org/13736 >> >> Regards, >> Jonas >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/92a22353/attachment.html From steveayre at gmail.com Fri Aug 27 00:57:57 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 27 Aug 2010 08:57:57 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: Yes, G.729a and G.729b are incorrect and your device is at fault... only other phones of the same type would probably recognise it without issue. PCMA should be used in this case though. RTP payload numbers are spread through multiple RFCs. Anything in the 96-127 range is dynamic and the codec is determined from the matching rtpmap line, any of the static numbers don't need a rtpmap line to work. IANA oversees assignment of the static numbers and they have the full list: http://www.iana.org/assignments/rtp-parameters As you can see 0=PCMU, 8=PCMA, and G.729 should use 18. Support of annex B is specified in the fmtp parameter, not the codec name - e.g. "a=fmtp:18 annexb=no". Annex A never needs to be specified as it can be read normally by plain G.729, so it's just up to the implementation on whether it wants to save quality or cpu when encoding. Do you have any other applications running which would also be using the RTP port range? A call will fail if it tries to use a port that's already in use, perhaps with that message. FS should avoid using ports it's already using, but can't know about any other programs on the system. -Steve On 27 August 2010 08:03, Jonas Gauffin wrote: > It doesn't happen every time and it's on a production system with a bit of > volume. therefore a bit hard to get SIP traces. I'll try if Anthony really > needs them. > > FS do say this: > 2010-08-27 07:14:10.758750 [DEBUG] switch_ivr_originate.c:3111sofia/external/ > 0700123456 at 212.151.Y.Y:5060 Media Establishment Failed > > In which RFC are codec names defined? rfc3551 defines "G729" but no > "G.729A" or "G.729B". But as you say, shouldn't FS use PCMA in any case? > > > > On Fri, Aug 27, 2010 at 8:43 AM, Steven Ayre wrote: > >> Ordinarily it'd mean there was a problem with the codecs (e.g. needing to >> use an unsupported codec or transcode a codec that only works as a >> passthrough one). >> >> Looks like it should have gone through with PCMA (8) though. Can you >> repeat the call with sip trace on? Perhaps the incompatible destination >> comes from an endpoint. >> >> 'sofia profile siptrace on' from the CLI, replace on with >> off to turn it off again. >> >> -Steve >> >> >> >> On 27 August 2010 07:26, Jonas Gauffin wrote: >> >>> Hello, >>> >>> Why do this call result in INCOMPATIBLE_DESTINATION? >>> >>> http://pastebin.freeswitch.org/13736 >>> >>> Regards, >>> Jonas >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/80e8c862/attachment.html From jim at k4gvo.com Fri Aug 27 00:59:56 2010 From: jim at k4gvo.com (Jim) Date: Fri, 27 Aug 2010 03:59:56 -0400 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: References: <4C76464D.4070308@k4gvo.com> <4C7696D1.6000103@k4gvo.com> Message-ID: <4C77707C.8060301@k4gvo.com> On 08/26/2010 04:19 PM, Michael Collins wrote: > > > On Thu, Aug 26, 2010 at 9:31 AM, Jim > wrote: > > On 08/26/2010 12:00 PM, Michael Collins wrote: >> Peder mentioned it already, but a complete log from start to >> finish would be helpful. Use pastebin.freeswitch.org >> and then reply to this thread >> with the URL to your pastebin post. >> -MC >> > I don't like to waste computer space if it's not necessary. > Sometimes the answer is simple and the answer doesn't require > additional info. > > http://pastebin.freeswitch.org/13731 > > Note there are actually two problems demonstrated here. The first > attempt is calling the GrandStream from a SPA2102 connected > phone. The second is calling a POTS phone connected to the A200 > from the SPA2102 and the third successful attempt is the reverse. > There are notes at the beginning of the file giving approximate > starting line numbers. > > #1 From what I can see, the GS is doing a redirect. Look at line 226 > of the pastebin down to 240. Do you have call forwarding enabled on > your GS? You may want to enable a SIP trace to see the exact SIP > packet from the GS to see what it's doing. It is forwarding to > destination number of "service" and there isn't an extension for that > in your dialplan. (See lines leading up to 409 - "service" doesn't > match anything until it gets to enum, where it eventually fails > altogether at line 458.) How interesting. The phone is pretty simple, only a couple of pages and nothing that remotely mentions forwarding and/or "service". I guess I can put a service in the dial plan. > > #2 Does the phone connected to port 1 on the A200 even ring? It's > interesting that there is a 9 second gap (between pb lines 732 and > 733) which is where I would assume the ringing was taking place. Yes, it's ringing. I set the delay to 10 seconds rather than 30 when I was debugging it so I wouldn't have to wait so long. The phone rings, dials out, answers when I pick it up, but doesn't transfer to VM reliably. Thanks, Jim. > > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/727c1740/attachment-0001.html From odermann at googlemail.com Fri Aug 27 01:50:23 2010 From: odermann at googlemail.com (Dennis) Date: Fri, 27 Aug 2010 10:50:23 +0200 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: <150CAE7F-0509-4073-A49D-8BA4D45FE1F7@jerris.com> References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> <150CAE7F-0509-4073-A49D-8BA4D45FE1F7@jerris.com> Message-ID: hi all, thanks a lot for your support! the codec we are using is G711a, so normally using inband should work. no transcoding is done. we get G711, fs works with G711 and G711 leaves our servers. although the only short way, we are using voip is a 10 meter long CAT7 wire connected to a cirpack, we would prefer to use rfc-2833 for signaling dtmf inputs. the dtmf detector is off. we spoke to someone from the cirpack-support and he told us, that they do not filter the inband tone, if G711 is used, because we do not need rfc-2833 when using G711. if a lower quality codec is used (like G729), cirpack will filter the inband tone and send rfc-2833 instead. we have the following scenarios: 1.) we set dtmf to "info": cirpack sends us a clean tone and fs sees one clean tone (both: ||||||||||) conclusion: we could drop rfc-2833 and only use inband. 2.) we set dtmf to "rfc-2833": cirpack send us the tone with a tiny gap (2ms) and rfc-2833 (|| |||||||||). the guy from cirpack told us, that this is not a real gap, but only wireshark sees it as a gap. the gap is where cipack adds the requested rfc-2833 information. but, fs hears the inband tone as if it were 2 tones and the rfc-2833. this worked for us before, because we only reacted on the rfc-2833 signal. now we have to send the audio to another side. on the other side the 2 inband (one splitted) tones are heard, which leads to problems. under the following url you can see a screenshot taken from wireshark. the first row is, how cirpack sends the inband and rfc-2833. to last row shows, how fs sees the tone. https://photos-1.dropbox.com/i/o/uy03PGj0ZEtxIi3r7RV9dBkXXvRoxmJ-KlcBi1i9ovg/3475265/1282986000/8a42bd3 are there some timing-settings in fs, we could play with to make the tiny gap dissaper or something like that, or should we just switch completely to inband? thanks dennis From jim at k4gvo.com Fri Aug 27 01:52:25 2010 From: jim at k4gvo.com (Jim) Date: Fri, 27 Aug 2010 04:52:25 -0400 Subject: [Freeswitch-users] Voice mail isn't working on one extension. In-Reply-To: References: <4C76464D.4070308@k4gvo.com> <4C7696D1.6000103@k4gvo.com> Message-ID: <4C777CC9.2030309@k4gvo.com> On 08/26/2010 04:19 PM, Michael Collins wrote: > > > > #2 Does the phone connected to port 1 on the A200 even ring? It's > interesting that there is a 9 second gap (between pb lines 732 and > 733) which is where I would assume the ringing was taking place. > Looking at the log, I see where there was a EXECUTE ... answer followed immediately by a what looks like the state_core_machine telling FreeTDM to hang up, or am I reading that wrong? By the way, I do get a very short message before being dumped that says "Goodbye". She's pleasant, but curt. ;) By the way, I put a service in my dial plan and all is fine wrt the GS, thanks. Jim. From tculjaga at gmail.com Fri Aug 27 02:03:19 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 27 Aug 2010 11:03:19 +0200 Subject: [Freeswitch-users] long DTMF In-Reply-To: References: <984085.22821.qm@web29214.mail.ird.yahoo.com> Message-ID: as usual we are alone here ... I just lost my patience and did it directly in code! Once rewriting according to FS requirements done , will provide a patch if anyone interested. -----------------<-snip->----------------------- /*it's hardcoded but who cares at that point :))*/ if (dtmf->digit == '#') { int ok = 0; *digit = dtmf->digit; dval = atoi(digit); dval=0; /* temporary fake the bind key */ if (direction == SWITCH_DTMF_RECV && (md->sr[direction].map[dval].bind_flags & SBF_DIAL_ALEG)) { ok = 1; } else if (direction == SWITCH_DTMF_SEND && (md->sr[direction].map[dval].bind_flags & SBF_DIAL_BLEG)) { ok = 1; } if (ok && md->sr[direction].map[dval].app) { uint32_t flags = md->sr[direction].map[dval].flags; if ((md->sr[direction].map[dval].bind_flags & SBF_EXEC_OPPOSITE)) { if (direction == SWITCH_DTMF_SEND) { flags |= SMF_ECHO_ALEG; } else { flags |= SMF_ECHO_BLEG; } } else if ((md- >sr[direction].map[dval].bind_flags & SBF_EXEC_SAME)) { if (direction == SWITCH_DTMF_SEND) { flags |= SMF_ECHO_BLEG; } else { flags |= SMF_ECHO_ALEG; } } else if ((md- >sr[direction].map[dval].bind_flags & SBF_EXEC_ALEG)) { flags |= SMF_ECHO_ALEG; } else if ((md->sr[direction].map[dval].bind_flags & SBF_EXEC_BLEG)) { flags |= SMF_ECHO_BLEG; } else { flags |= SMF_ECHO_ALEG; } if ((md->sr[direction].map[dval].bind_flags & SBF_EXEC_INLINE)) { flags |= SMF_EXEC_INLINE; } switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "%s Processing meta digit '%c' [%s]\n", switch_channel_get_name(channel), dtmf->digit, md->sr[direction].map[dval].app); if (switch_channel_test_flag(channel, CF_PROXY_MODE)) { broadcast_in_thread(session, md->sr[direction].map[dval].app, flags | SMF_REBRIDGE); } else { switch_ivr_broadcast(switch_core_session_get_uuid(session), md->sr[direction].map[dval].app, flags); } if ((md->sr[direction].map[dval].bind_flags & SBF_ONCE)) { memset(&md->sr[direction].map[dval], 0, sizeof(md->sr[direction].map[dval])); switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "%s Unbinding meta digit '%c'\n", switch_channel_get_name(channel), dtmf->digit); } } else { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "%s Ignoring meta digit '%c' not mapped\n", switch_channel_get_name(channel), dtmf->digit); } /* return SWITCH_STATUS_SUCCESS; */ return SWITCH_STATUS_FALSE; } -----------------<-snip->----------------------- usage in DP: in myExtension you do your stuff! On Wed, Aug 25, 2010 at 10:14 PM, Tihomir Culjaga wrote: > > > On Wed, Aug 25, 2010 at 7:47 PM, Nigel Kent wrote: > >> Have you tried using the event socket library >> http://wiki.freeswitch.org/wiki/Esl >> >> > well, i could have used esl from the start but thats and external > application controling FS... and its totally a different approach from where > im now:(. > I have everything up & running except this feature .. > > so is there any chance to run a lua script in background (by allowing the > normal callflow to go on) to collect a special DTMF event ? > > something like this: > > > > ? > > T. > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/42f3af1a/attachment.html From spambox at haruhiism.net Thu Aug 26 22:59:00 2010 From: spambox at haruhiism.net (Kamigishi Rei) Date: Fri, 27 Aug 2010 09:59:00 +0400 Subject: [Freeswitch-users] Deployment questions: Freeswitch or Freeswitch with a SIP proxy server In-Reply-To: <4C752BE3.1070407@communicatefreely.net> References: <4C74B7EF.7070102@haruhiism.net> <4C752BE3.1070407@communicatefreely.net> Message-ID: <4C775424.6090109@haruhiism.net> On 25.08.2010 18:42, Tim St. Pierre wrote: > "public" profile: > Listens on port 5080 > Has it's own dialplan that can deliver a call to extensions, but doesn't allow calls to much else. > The dialplan is written with security in mind. > The DNS SRV record for our domain points to the public IP address for this profile, port 5080. > Anyone dialing EXTEN at our.domain gets sent here, and the call is processed. Yeah, currently I have the same setup for DNS, just the profile is called 'external' like in the default configuration. > "internal" profile. > listens on port 5070 (and eventually 5071 when I get TLS working) > Is used for all the SIP endpoints, and authenticates them when they register. So basically it works mostly like the default 'internal' profile, right? > If a call comes in on this profile for anything other than pbx.our.domain, we can match this in the > dialplan and send a call out on the "public" profile to whomever they want to call. No ACL checking? Because the default internal profile just rejects everything that arrives to it and is not authenticated. -- Kamigishi Rei KREI-RIPE From bwibowo at gmail.com Fri Aug 27 02:25:04 2010 From: bwibowo at gmail.com (Budi wibowo) Date: Fri, 27 Aug 2010 09:25:04 +0000 Subject: [Freeswitch-users] long DTMF In-Reply-To: References: <984085.22821.qm@web29214.mail.ird.yahoo.com> Message-ID: <771742469-1282901283-cardhu_decombobulator_blackberry.rim.net-1526623500-@bda009.bisx.prodap.on.blackberry> TtT -----Original Message----- From: Tihomir Culjaga Sender: freeswitch-users-bounces at lists.freeswitch.org Date: Fri, 27 Aug 2010 11:03:19 To: FreeSWITCH Users Help Reply-To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] long DTMF _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mnhassan at usa.net Fri Aug 27 04:17:44 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Fri, 27 Aug 2010 17:17:44 +0600 Subject: [Freeswitch-users] long DTMF In-Reply-To: References: <984085.22821.qm@web29214.mail.ird.yahoo.com> Message-ID: I did read your emails, and I thought the Lua script idea was nice. The ESL was more complicated, but also good. But, you didn't accept either. So, I was wondering, is this "bind_long_DTMF" something that allows you to capture DTMF during the full length of the call, i.e. even after bridge? Is that what you wished to accomplish? Regards HASSAN On Fri, Aug 27, 2010 at 15:03, Tihomir Culjaga wrote: > as usual we are alone here ... > I just lost my patience and did it directly in code! > > > Once rewriting according to FS requirements done , will provide a patch if > anyone interested. > > > > -----------------<-snip->----------------------- > /*it's hardcoded but who cares at that point :))*/ > if (dtmf->digit == '#') { > int ok = 0; > *digit = dtmf->digit; > dval = atoi(digit); > dval=0; /* temporary fake the bind key */ > if (direction == SWITCH_DTMF_RECV && > (md->sr[direction].map[dval].bind_flags & SBF_DIAL_ALEG)) { > ok = 1; > } else if (direction == SWITCH_DTMF_SEND && > (md->sr[direction].map[dval].bind_flags & SBF_DIAL_BLEG)) { > ok = 1; > } > > if (ok && md->sr[direction].map[dval].app) { > uint32_t flags = > md->sr[direction].map[dval].flags; > > if ((md->sr[direction].map[dval].bind_flags > & > SBF_EXEC_OPPOSITE)) { if (direction == > SWITCH_DTMF_SEND) { flags |= SMF_ECHO_ALEG; > } > else { flags |= SMF_ECHO_BLEG; } } else if > ((md- > >sr[direction].map[dval].bind_flags & > SBF_EXEC_SAME)) { if (direction == > SWITCH_DTMF_SEND) { flags |= SMF_ECHO_BLEG; > } > else { flags |= SMF_ECHO_ALEG; } } else if > ((md- > >sr[direction].map[dval].bind_flags & > SBF_EXEC_ALEG)) { flags |= SMF_ECHO_ALEG; } > else > if ((md->sr[direction].map[dval].bind_flags > & > SBF_EXEC_BLEG)) { flags |= SMF_ECHO_BLEG; } > else > { flags |= SMF_ECHO_ALEG; } > > if ((md->sr[direction].map[dval].bind_flags > & SBF_EXEC_INLINE)) { > flags |= SMF_EXEC_INLINE; > } > > > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "%s > Processing meta digit '%c' [%s]\n", > > switch_channel_get_name(channel), dtmf->digit, > md->sr[direction].map[dval].app); > > if (switch_channel_test_flag(channel, > CF_PROXY_MODE)) { > broadcast_in_thread(session, > md->sr[direction].map[dval].app, flags | SMF_REBRIDGE); > } else { > > switch_ivr_broadcast(switch_core_session_get_uuid(session), > md->sr[direction].map[dval].app, flags); > } > if ((md->sr[direction].map[dval].bind_flags > & SBF_ONCE)) { > > memset(&md->sr[direction].map[dval], 0, > sizeof(md->sr[direction].map[dval])); > > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "%s > Unbinding meta digit '%c'\n", > > switch_channel_get_name(channel), dtmf->digit); > } > } else { > > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, > "%s Ignoring meta digit '%c' not mapped\n", > > switch_channel_get_name(channel), dtmf->digit); > } > > > /* return SWITCH_STATUS_SUCCESS; */ > return SWITCH_STATUS_FALSE; > > } > > > > -----------------<-snip->----------------------- > > > > usage in DP: > > > > > > in myExtension you do your stuff! > > > > > > > > > > > On Wed, Aug 25, 2010 at 10:14 PM, Tihomir Culjaga wrote: > >> >> >> On Wed, Aug 25, 2010 at 7:47 PM, Nigel Kent wrote: >> >>> Have you tried using the event socket library >>> http://wiki.freeswitch.org/wiki/Esl >>> >>> >> well, i could have used esl from the start but thats and external >> application controling FS... and its totally a different approach from where >> im now:(. >> I have everything up & running except this feature .. >> >> so is there any chance to run a lua script in background (by allowing the >> normal callflow to go on) to collect a special DTMF event ? >> >> something like this: >> >> >> >> ? >> >> T. >> >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/c11f5f46/attachment-0001.html From jonas.gauffin at gmail.com Fri Aug 27 04:36:13 2010 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 27 Aug 2010 13:36:13 +0200 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: Excellent information thanks. The server is a dedicated voip server and nothing else is running on it (and therefore the port should be free). I've talked to my voip provider and they were kind enough to give me their traces. Freeswitch get's their SDP on the bleg (their software have modified the trace a bit): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 212.247.E.E;rport=5060;branch=z9hG4bK52XcXtpQBp3BH From: "blablabla" ;tag=rccD75cXD61Kr To: >;tag=686577264 Call-ID: c3e8e0e8-2c3c-122e-479c-1fc6e9408ca4 CSeq: 1106961 INVITE Contact: Record-Route: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Content-Type: application/sdp Content-Length: 169 v=0 o=- 46541649 0 IN IP4 130.244.x.x s=Cisco SDP 0 c=IN IP4 130.244.x.x t=0 0 m=audio 18928 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 which makes FS cancel the call leg: CANCEL sip:0771221122 at voipprovider.se SIP/2.0 Via: SIP/2.0/UDP 212.247.E.E;rport;branch=z9hG4bK52XcXtpQBp3BH Max-Forwards: 68 From: "blablabla" ;tag=rccD75cXD61Kr To: > Call-ID: c3e8e0e8-2c3c-122e-479c-1fc6e9408ca4 CSeq: 1106961 CANCEL Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 On Fri, Aug 27, 2010 at 9:57 AM, Steven Ayre wrote: > Yes, G.729a and G.729b are incorrect and your device is at fault... only > other phones of the same type would probably recognise it without issue. > > PCMA should be used in this case though. > > RTP payload numbers are spread through multiple RFCs. Anything in the > 96-127 range is dynamic and the codec is determined from the matching rtpmap > line, any of the static numbers don't need a rtpmap line to work. IANA > oversees assignment of the static numbers and they have the full list: > http://www.iana.org/assignments/rtp-parameters > > As you can see 0=PCMU, 8=PCMA, and G.729 should use 18. Support of annex B > is specified in the fmtp parameter, not the codec name - e.g. "a=fmtp:18 > annexb=no". Annex A never needs to be specified as it can be read normally > by plain G.729, so it's just up to the implementation on whether it wants to > save quality or cpu when encoding. > > Do you have any other applications running which would also be using the > RTP port range? A call will fail if it tries to use a port that's already in > use, perhaps with that message. FS should avoid using ports it's already > using, but can't know about any other programs on the system. > > -Steve > > > > On 27 August 2010 08:03, Jonas Gauffin wrote: > >> It doesn't happen every time and it's on a production system with a bit of >> volume. therefore a bit hard to get SIP traces. I'll try if Anthony really >> needs them. >> >> FS do say this: >> 2010-08-27 07:14:10.758750 [DEBUG] switch_ivr_originate.c:3111sofia/external/ >> 0700123456 at 212.151.Y.Y:5060 Media Establishment Failed >> >> In which RFC are codec names defined? rfc3551 defines "G729" but no >> "G.729A" or "G.729B". But as you say, shouldn't FS use PCMA in any case? >> >> >> >> On Fri, Aug 27, 2010 at 8:43 AM, Steven Ayre wrote: >> >>> Ordinarily it'd mean there was a problem with the codecs (e.g. needing to >>> use an unsupported codec or transcode a codec that only works as a >>> passthrough one). >>> >>> Looks like it should have gone through with PCMA (8) though. Can you >>> repeat the call with sip trace on? Perhaps the incompatible destination >>> comes from an endpoint. >>> >>> 'sofia profile siptrace on' from the CLI, replace on with >>> off to turn it off again. >>> >>> -Steve >>> >>> >>> >>> On 27 August 2010 07:26, Jonas Gauffin wrote: >>> >>>> Hello, >>>> >>>> Why do this call result in INCOMPATIBLE_DESTINATION? >>>> >>>> http://pastebin.freeswitch.org/13736 >>>> >>>> Regards, >>>> Jonas >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/25608c00/attachment.html From rupa at rupa.com Fri Aug 27 05:14:39 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 27 Aug 2010 07:14:39 -0500 Subject: [Freeswitch-users] How to set .wav file as a voice-mail greeting? In-Reply-To: <4C773322.7030508@isptelecom.net> References: <4C7412F1.9000509@isptelecom.net> <4C773322.7030508@isptelecom.net> Message-ID: Have you looked at the vm_prefs api? On Thu, Aug 26, 2010 at 10:38 PM, Victor Chukalovskiy wrote: > Quick question: > Is there an elegant way of using pre-recorder .wav file as a greeting > for voice mail box of a user? > My feeling is that No unless editing DB. But I'd be happy to learn I'm > mistaken :-) > > Thank you, > Victor > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/105ac322/attachment.html From lloydie.t at gmail.com Fri Aug 27 02:46:24 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Fri, 27 Aug 2010 10:46:24 +0100 Subject: [Freeswitch-users] weird bootstrap error Message-ID: Got a wierd error (i think) running bootstrap.sh. It stop at the following line and did not return to the linus prompt. /usr/src/freeswitch# automake: compiling `check_dlopen_sofia.c' with per-target flags requires `AM_PROG_CC_C_O' in `configure.ac' tests/Makefile.am:37: while processing program `check_dlopen_sofia' Is this normal? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/f9707879/attachment.html From steveayre at gmail.com Fri Aug 27 06:39:59 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 27 Aug 2010 14:39:59 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: Message-ID: Try pressing enter a few times. Are you then at the command prompt? I had this earlier too, and got my command line then another automake line appeared - it appeared that for me at least the bootstrap had gone into the background. -Steve On 27 August 2010 10:46, lloyd thomas wrote: > Got a wierd error (i think) running bootstrap.sh. It stop at the following > line and did not return to the linus prompt. > > /usr/src/freeswitch# automake: compiling `check_dlopen_sofia.c' with > per-target flags requires `AM_PROG_CC_C_O' in `configure.ac' > tests/Makefile.am:37: while processing program `check_dlopen_sofia' > > Is this normal? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/b8b265bd/attachment.html From lloydie.t at gmail.com Fri Aug 27 07:05:40 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Fri, 27 Aug 2010 15:05:40 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: Message-ID: I did that and the command prompt appeared. unfortunately make failed with a spandsp problem. Trying again. Thanks On 27 August 2010 14:39, Steven Ayre wrote: > Try pressing enter a few times. Are you then at the command prompt? > > I had this earlier too, and got my command line then another automake line > appeared - it appeared that for me at least the bootstrap had gone into the > background. > > -Steve > > > On 27 August 2010 10:46, lloyd thomas wrote: > >> Got a wierd error (i think) running bootstrap.sh. It stop at the following >> line and did not return to the linus prompt. >> >> /usr/src/freeswitch# automake: compiling `check_dlopen_sofia.c' with >> per-target flags requires `AM_PROG_CC_C_O' in `configure.ac' >> tests/Makefile.am:37: while processing program `check_dlopen_sofia' >> >> Is this normal? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/00dc2a4b/attachment-0001.html From brian at freeswitch.org Fri Aug 27 07:12:53 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Aug 2010 09:12:53 -0500 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: Message-ID: What crazy distro are you on? /b On Aug 27, 2010, at 9:05 AM, lloyd thomas wrote: > I did that and the command prompt appeared. unfortunately make failed with a spandsp problem. Trying again. > > Thanks > > On 27 August 2010 14:39, Steven Ayre wrote: > Try pressing enter a few times. Are you then at the command prompt? > > I had this earlier too, and got my command line then another automake line appeared - it appeared that for me at least the bootstrap had gone into the background. > > -Steve From me at nevian.org Fri Aug 27 07:24:27 2010 From: me at nevian.org (Serge Yuriev) Date: Fri, 27 Aug 2010 18:24:27 +0400 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: Message-ID: Hello, I think is debian based. I got the same also on squeeze. Need just do what it said in libs/sofia-sip/configure.ac On 27 August 2010 18:12, Brian West wrote: > What crazy distro are you on? > > /b > > On Aug 27, 2010, at 9:05 AM, lloyd thomas wrote: > >> I did that and the command prompt appeared. unfortunately make failed with a spandsp problem. Trying again. >> >> Thanks >> >> On 27 August 2010 14:39, Steven Ayre wrote: >> Try pressing enter a few times. Are you then at the command prompt? >> >> I had this earlier too, and got my command line then another automake line appeared - it appeared that for me at least the bootstrap had gone into the background. -- With kind regards, Serge From brian at freeswitch.org Fri Aug 27 07:33:28 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Aug 2010 09:33:28 -0500 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: Message-ID: <7BF70117-585D-4C7B-84F5-C42E645632D3@freeswitch.org> Please put this on Jira. /b On Aug 27, 2010, at 9:24 AM, Serge Yuriev wrote: > Hello, > > I think is debian based. I got the same also on squeeze. > Need just do what it said in libs/sofia-sip/configure.ac > > On 27 August 2010 18:12, Brian West wrote: >> What crazy distro are you on? >> >> /b From 12ukwn at gmail.com Fri Aug 27 07:58:11 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Fri, 27 Aug 2010 16:58:11 +0200 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: Message-ID: <20100827165811.333ca9a2@anubis.defcon1> On Fri, 27 Aug 2010 18:24:27 +0400, Serge Yuriev wrote: Hi Serge, try reloading the last git: I recently had a PB (make clean saying args list was too long), so I deleted the tree, then reloaded it, and it worked fine :) > I think is debian based. I got the same also on squeeze. > Need just do what it said in libs/sofia-sip/configure.ac -- who gives a shit about US law anyone living in the US. From lloydie.t at gmail.com Fri Aug 27 07:57:45 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Fri, 27 Aug 2010 15:57:45 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: <7BF70117-585D-4C7B-84F5-C42E645632D3@freeswitch.org> References: <7BF70117-585D-4C7B-84F5-C42E645632D3@freeswitch.org> Message-ID: ubuntu server 8.10 On 27 August 2010 15:33, Brian West wrote: > Please put this on Jira. > > /b > > On Aug 27, 2010, at 9:24 AM, Serge Yuriev wrote: > > > Hello, > > > > I think is debian based. I got the same also on squeeze. > > Need just do what it said in libs/sofia-sip/configure.ac > > > > On 27 August 2010 18:12, Brian West wrote: > >> What crazy distro are you on? > >> > >> /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/0bf1427d/attachment.html From mike at jerris.com Fri Aug 27 08:33:36 2010 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Aug 2010 11:33:36 -0400 Subject: [Freeswitch-users] Serious and urgent problem with DTMF! Please help... In-Reply-To: References: <2BEBDA90-D960-49B5-B0BF-9FD0136C3B6D@gmail.com> <150CAE7F-0509-4073-A49D-8BA4D45FE1F7@jerris.com> Message-ID: <283900C8-142E-4484-ADB3-693ECD1BC839@jerris.com> From your explanation, the cirpack is sending the dtmf both 2833 AND inband. They ARE sending it to you twice, in 2 different methods. How do you want the other side to get the dtmf? try pass_2833 param. Mike On Aug 27, 2010, at 4:50 AM, Dennis wrote: > hi all, > > thanks a lot for your support! > > the codec we are using is G711a, so normally using inband should work. > no transcoding is done. we get G711, fs works with G711 and G711 > leaves our servers. > although the only short way, we are using voip is a 10 meter long CAT7 > wire connected to a cirpack, we would prefer to use rfc-2833 for > signaling dtmf inputs. > > the dtmf detector is off. > > we spoke to someone from the cirpack-support and he told us, that they > do not filter the inband tone, if G711 is used, because we do not need > rfc-2833 when using G711. if a lower quality codec is used (like > G729), cirpack will filter the inband tone and send rfc-2833 instead. > > > we have the following scenarios: > > 1.) we set dtmf to "info": > > cirpack sends us a clean tone and fs sees one clean tone (both: ||||||||||) > > conclusion: we could drop rfc-2833 and only use inband. > > > 2.) we set dtmf to "rfc-2833": > > cirpack send us the tone with a tiny gap (2ms) and rfc-2833 (|| > |||||||||). the guy from cirpack told us, that this is not a real gap, > but only wireshark sees it as a gap. the gap is where cipack adds the > requested rfc-2833 information. > but, fs hears the inband tone as if it were 2 tones and the rfc-2833. > this worked for us before, because we only reacted on the rfc-2833 > signal. > now we have to send the audio to another side. on the other side the 2 > inband (one splitted) tones are heard, which leads to problems. > > > under the following url you can see a screenshot taken from wireshark. > the first row is, how cirpack sends the inband and rfc-2833. to last > row shows, how fs sees the tone. > https://photos-1.dropbox.com/i/o/uy03PGj0ZEtxIi3r7RV9dBkXXvRoxmJ-KlcBi1i9ovg/3475265/1282986000/8a42bd3 > > > are there some timing-settings in fs, we could play with to make the > tiny gap dissaper or something like that, or should we just switch > completely to inband? > > > thanks > dennis > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fdelawarde at wirelessmundi.com Fri Aug 27 08:44:54 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Fri, 27 Aug 2010 17:44:54 +0200 Subject: [Freeswitch-users] voicemail profile dbname Message-ID: <1282923894.2736.3.camel@luna.tc.commsmundi.com> Hello, Is there any way to setup the name of the internal database for voicemail? The default seem to use one database per voicemail profile such as: voicemail_.db I would like to be able to use multiple voicemail profiles with the same users if possible, and share the database between all voicemail profiles. Are the vm profiles made to be able to work that way? Thanks, Fran?ois. From lloydie.t at gmail.com Fri Aug 27 08:56:31 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Fri, 27 Aug 2010 16:56:31 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: <20100827165811.333ca9a2@anubis.defcon1> References: <20100827165811.333ca9a2@anubis.defcon1> Message-ID: Not going well. From last night GIT Wl,-soname -Wl,mod_spandsp.so -o .libs/mod_spandsp.so make[4]: *** [mod_spandsp.la] Error 1 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 As I am not that clever, should I revert to 1.0.6 official release? On 27 August 2010 15:58, Jean-Yves F. Barbier <12ukwn at gmail.com> wrote: > On Fri, 27 Aug 2010 18:24:27 +0400, Serge Yuriev wrote: > > > Hi Serge, try reloading the last git: I recently had a PB (make clean > saying > args list was too long), so I deleted the tree, then reloaded it, and it > worked fine :) > > > I think is debian based. I got the same also on squeeze. > > Need just do what it said in libs/sofia-sip/configure.ac > > -- > who gives a shit about US law > anyone living in the US. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/3d230326/attachment.html From moises.silva at gmail.com Fri Aug 27 08:59:22 2010 From: moises.silva at gmail.com (Moises Silva) Date: Fri, 27 Aug 2010 11:59:22 -0400 Subject: [Freeswitch-users] New wiki page In-Reply-To: <4C76A63C.7010506@k4gvo.com> References: <4C76A63C.7010506@k4gvo.com> Message-ID: Again, I'd like to thank you here for this. We're so much in need of people like you getting out of their way to document what they learned! Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com On Thu, Aug 26, 2010 at 1:37 PM, Jim wrote: > For what it's worth, I documented my experience with getting a Sangoma > A200 card working on FreeSWITCH. I have no idea if it's the right way > but it works sort of. > > http://wiki.freeswitch.org/wiki/Sangoma_A200 > > Jim. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/aec7c4f5/attachment-0001.html From brian at freeswitch.org Fri Aug 27 09:01:06 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Aug 2010 11:01:06 -0500 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: <20100827165811.333ca9a2@anubis.defcon1> Message-ID: rebootstrap and ./configure /b On Aug 27, 2010, at 10:56 AM, lloyd thomas wrote: > Not going well. From last night GIT > > Wl,-soname -Wl,mod_spandsp.so -o .libs/mod_spandsp.so > make[4]: *** [mod_spandsp.la] Error 1 > make[3]: *** [mod_spandsp-all] Error 1 > make[2]: *** [all-recursive] Error 1 > > As I am not that clever, should I revert to 1.0.6 official release? From steveayre at gmail.com Fri Aug 27 09:12:54 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 27 Aug 2010 17:12:54 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: Message-ID: Well, for me that was on Ubuntu. I know, I know... I was just testing a patch (FSCORE-662) on my desktop PC, it wasn't the production system, or even my test system for anything requiring audio. I haven't had any problems on Debian myself, but I use Lenny and haven't compiled from git for a week or two. -Steve On 27 August 2010 15:12, Brian West wrote: > What crazy distro are you on? > > /b > > On Aug 27, 2010, at 9:05 AM, lloyd thomas wrote: > > > I did that and the command prompt appeared. unfortunately make failed > with a spandsp problem. Trying again. > > > > Thanks > > > > On 27 August 2010 14:39, Steven Ayre wrote: > > Try pressing enter a few times. Are you then at the command prompt? > > > > I had this earlier too, and got my command line then another automake > line appeared - it appeared that for me at least the bootstrap had gone into > the background. > > > > -Steve > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/ea15ed70/attachment.html From mike at jerris.com Fri Aug 27 09:26:57 2010 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Aug 2010 12:26:57 -0400 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: Message-ID: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> I just got a patch on this today, from you maybe? I'll test this out on a few platforms and try to get this in this weekend. Thanks Mike On Aug 27, 2010, at 12:12 PM, Steven Ayre wrote: > Well, for me that was on Ubuntu. I know, I know... I was just testing a patch (FSCORE-662) on my desktop PC, it wasn't the production system, or even my test system for anything requiring audio. > > I haven't had any problems on Debian myself, but I use Lenny and haven't compiled from git for a week or two. > > -Steve > > > On 27 August 2010 15:12, Brian West wrote: > What crazy distro are you on? > > /b > > On Aug 27, 2010, at 9:05 AM, lloyd thomas wrote: > > > I did that and the command prompt appeared. unfortunately make failed with a spandsp problem. Trying again. > > > > Thanks > > > > On 27 August 2010 14:39, Steven Ayre wrote: > > Try pressing enter a few times. Are you then at the command prompt? > > > > I had this earlier too, and got my command line then another automake line appeared - it appeared that for me at least the bootstrap had gone into the background. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/7eea4a9e/attachment.html From steveayre at gmail.com Fri Aug 27 09:38:03 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 27 Aug 2010 17:38:03 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> References: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> Message-ID: Yea, uploaded it earlier today. :) What do you think of it? Good or a bad idea? -Steve On 27 August 2010 17:26, Michael Jerris wrote: > I just got a patch on this today, from you maybe? I'll test this out on a > few platforms and try to get this in this weekend. > > Thanks > Mike > > On Aug 27, 2010, at 12:12 PM, Steven Ayre wrote: > > Well, for me that was on Ubuntu. I know, I know... I was just testing a > patch (FSCORE-662) on my desktop PC, it wasn't the production system, or > even my test system for anything requiring audio. > > I haven't had any problems on Debian myself, but I use Lenny and haven't > compiled from git for a week or two. > > -Steve > > > On 27 August 2010 15:12, Brian West wrote: > >> What crazy distro are you on? >> >> /b >> >> On Aug 27, 2010, at 9:05 AM, lloyd thomas wrote: >> >> > I did that and the command prompt appeared. unfortunately make failed >> with a spandsp problem. Trying again. >> > >> > Thanks >> > >> > On 27 August 2010 14:39, Steven Ayre wrote: >> > Try pressing enter a few times. Are you then at the command prompt? >> > >> > I had this earlier too, and got my command line then another automake >> line appeared - it appeared that for me at least the bootstrap had gone into >> the background. >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/d955e121/attachment.html From msc at freeswitch.org Fri Aug 27 09:54:51 2010 From: msc at freeswitch.org (Michael Collins) Date: Fri, 27 Aug 2010 09:54:51 -0700 Subject: [Freeswitch-users] External caller-number not shown on forward to external number In-Reply-To: References: Message-ID: Put in some log apps or the info app prior to your transfer and bridge apps. Make sure that these variables actually contain what you hope they do. Also, check with the telco to make sure that they allow you to pass customized caller ID like this. Some support it and some don't, but sometimes you have to ask explicitly to have the feature allowed. Finally, make sure that you are passing the phone number in a format that the telco wants, i.e. do they require the leading 1. -MC On Thu, Aug 26, 2010 at 3:32 AM, Peter Waldheim < peter.waldheim at framesoft.com> wrote: > Hi all, > > In a diaplan I'm forwarding a call to an external (PRI) number when it > fails to reach my SIP-Client: > > > data="outbound_caller_id_number=${caller_id_number}"/> > data="outbound_caller_id_name=${caller_id_name}"/> > > Then in that dialplan: > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > > So far this works just fine but one thing: the external party only sees the > original caller-id if it is an internal one. If the caller was external, I > only see the default caller-number of my installation. Ideally I'd like to > always see the original caller id - especially if it is an external number. > Can anybody please give me a hint whether I'm missing something or this > might be some call-spoof protection in freeswitch or openzap or libpri or my > PRI provider? > > Thanks a lot > Peter > ________________________________ > > > Framesoft AG Software Applications > Sumpfstrasse 15 > 6301 Zug > Switzerland > Handelsregister des Kantons Zug: CH-170.3.022.876-2 > Pr?sident & Vorsitzender der Gesch?ftsleitung: Toralf Dittmann > > Framesoft AG Software Applications > Reuterweg 49 > D-60323 Frankfurt am Main > Germany > HRB 34142 > Vorsitzender des Aufsichtsrates: Toralf Dittmann > Vorstand: Jens Saarholz > > Framesoft Ltd. > Business Address: > 60 Lombard Street > London EC3V9EA, UK > Registered Office: > Hackwood Secretaries Limited > 1 silk Street > London EC2Y 8HQ, UK > Company Number: 4055017 > Directors: Toralf Dittmann, Volker Weber > Secretary: Volker Weber > > Confidentiality Notice: The information contained in this e-mail is > intended for the named recipient(s) only. It may contain privileged and > confidential information, and if you are not the addressee or the person > responsible for delivering this to the addressee, you may not copy, > distribute or take action in reliance on it. If you have received this > e-mail in error, please notify us immediately by returning the original > message to the sender by e-mail and delete this message. Thank you for your > cooperation. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/4132c4ae/attachment-0001.html From mike at jerris.com Fri Aug 27 09:57:59 2010 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Aug 2010 12:57:59 -0400 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> Message-ID: I think it is fine, but I have a recollection of trying this before and it not solving the problem, but the problem I had was the same warning message from bootstrap, but it did not actually cause it to fail, so I want to do some testing on different platforms to confirm for sure. Anyone else willing to test and comment on http://jira.freeswitch.org/browse/FSBUILD-300 would be appreciated. Mike On Aug 27, 2010, at 12:38 PM, Steven Ayre wrote: > Yea, uploaded it earlier today. :) What do you think of it? Good or a bad idea? > > -Steve > > On 27 August 2010 17:26, Michael Jerris wrote: > I just got a patch on this today, from you maybe? I'll test this out on a few platforms and try to get this in this weekend. > > Thanks > Mike > > On Aug 27, 2010, at 12:12 PM, Steven Ayre wrote: > >> Well, for me that was on Ubuntu. I know, I know... I was just testing a patch (FSCORE-662) on my desktop PC, it wasn't the production system, or even my test system for anything requiring audio. >> >> I haven't had any problems on Debian myself, but I use Lenny and haven't compiled from git for a week or two. >> >> -Steve >> >> >> On 27 August 2010 15:12, Brian West wrote: >> What crazy distro are you on? >> >> /b >> >> On Aug 27, 2010, at 9:05 AM, lloyd thomas wrote: >> >> > I did that and the command prompt appeared. unfortunately make failed with a spandsp problem. Trying again. >> > >> > Thanks >> > >> > On 27 August 2010 14:39, Steven Ayre wrote: >> > Try pressing enter a few times. Are you then at the command prompt? >> > >> > I had this earlier too, and got my command line then another automake line appeared - it appeared that for me at least the bootstrap had gone into the background. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/f437046b/attachment.html From tculjaga at gmail.com Fri Aug 27 10:23:50 2010 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 27 Aug 2010 19:23:50 +0200 Subject: [Freeswitch-users] long DTMF In-Reply-To: References: <984085.22821.qm@web29214.mail.ird.yahoo.com> Message-ID: On Fri, Aug 27, 2010 at 1:17 PM, Nyamul Hassan wrote: > I did read your emails, and I thought the Lua script idea was nice. lua was nice but it was blocking ... no further dialplan processing before it ends (at least this is what i went into)... and if it ends there is no Callback to collect DTMF digits :( I tried to put it in background but i was not able to... anyhow it was a nice experience ... my 1st lua scripting :) > The ESL was more complicated, but also good. But, you didn't accept > either. ESL ... huh its really nice but not applicable to my application :( ... I decided to go dialplan way and did everything within. honestly a separate daemon than interacts with FS and decides what to do according to events scares me a bit ... I have never done something like this and i was not confident i would complete the task on time. This means moving the logic outside FS... anyhow this is something on my next ToDo list :) > So, I was wondering, is this "bind_long_DTMF" something that allows you to > capture DTMF during the full length of the call, i.e. even after bridge? Is > that what you wished to accomplish? > > > This is a callingCard application and one of the requirments was to be able to cancel the wrong destination number entry with a single # (* is end of dial digit) starting from the point it asks you for the destination number afterwards. This means you can cancel an while; getting destination number, calls in being bridged, already bridged calls and return the A-leg call state back to enter another destination. As i didn't find any solution around here to do such a thing in dialplan or via scripting i went down to the original code and wrote my own function. BTW: Also, i will extend the function to do what it's name says "long". If someone press # for a certain amount of time (topically 2 seconds) an action will be triggered. If the duration is less than the specified time nothing will happen. Thanks for your e-mail. T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/10abd405/attachment.html From me at nevian.org Fri Aug 27 10:42:12 2010 From: me at nevian.org (Serge Yuriev) Date: Fri, 27 Aug 2010 21:42:12 +0400 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> Message-ID: Hello, Applied my patch on CentOS - success. Commented this on Jira On 27 August 2010 20:57, Michael Jerris wrote: > I think it is fine, but I have a recollection of trying this before and it > not solving the problem, but the problem I had was the same warning message > from bootstrap, but it did not actually cause it to fail, so I want to do > some testing on different platforms to confirm for sure. ?Anyone else > willing to test and comment on?http://jira.freeswitch.org/browse/FSBUILD-300 > would be appreciated. -- With kind regards, Serge From gmaruzz at celliax.org Fri Aug 27 10:56:01 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 27 Aug 2010 19:56:01 +0200 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Hi Jason, can you retest with latest git? Your problem was a bug in the management of the early media coming from the leg B (eg: sofia, etc) to the A leg (eg skypopen). Now is fixed, I believe. Please let me know how it goes. -giovanni On Tue, Aug 24, 2010 at 1:34 AM, Jason Jeffords wrote: > Hi Giovanni, > Our case 3 test was actually being bridged to a SIP endpoint (not in a > conference, although > this probably should not matter). > We tested two type 3 cases: > 1) Skype to FreeSWITCH Skype bridged to an extension of a registered SIP > phone > 2) Skype to FreeSWITCH Skype bridged to an outbound call through a PSTN > gateway > In both cases we are transiting FreeSWITCH, not terminating on it (could > there be a > race condition when doing signaling coordination to remote SIP endpoints, > not FreeSWITCH > itself?). > When we terminate Skype calls on FreeSWITCH this works (case 2). ?It also > works > for outbound Skype calls (case1 - SIP to FreeSWITCH, FreeSWITCH Skype to > Skype). > Also, we are running the very latest (well, as of this morning) git version, > so that > may introduce one more variable (if you are running an older version). > Thanks for your help, > Jason -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From Victor at isptelecom.net Fri Aug 27 08:35:31 2010 From: Victor at isptelecom.net (Victor Chukalovskiy) Date: Fri, 27 Aug 2010 11:35:31 -0400 Subject: [Freeswitch-users] How to set .wav file as a voice-mail greeting? In-Reply-To: References: <4C7412F1.9000509@isptelecom.net> <4C773322.7030508@isptelecom.net> Message-ID: <4C77DB43.4060806@isptelecom.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/effef345/attachment.html From lloydie.t at gmail.com Fri Aug 27 11:27:56 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Fri, 27 Aug 2010 19:27:56 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> Message-ID: "rebootstrap and ./configure" Did that already. Is there a newer snapshot that yesterday 6pm BST? On 27 August 2010 18:42, Serge Yuriev wrote: > Hello, > > Applied my patch on CentOS - success. > Commented this on Jira > > On 27 August 2010 20:57, Michael Jerris wrote: > > I think it is fine, but I have a recollection of trying this before and > it > > not solving the problem, but the problem I had was the same warning > message > > from bootstrap, but it did not actually cause it to fail, so I want to do > > some testing on different platforms to confirm for sure. Anyone else > > willing to test and comment on > http://jira.freeswitch.org/browse/FSBUILD-300 > > would be appreciated. > > -- > With kind regards, > Serge > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/89699409/attachment-0001.html From steveayre at gmail.com Fri Aug 27 11:57:21 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 27 Aug 2010 19:57:21 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> Message-ID: Use git. On 27 August 2010 19:27, lloyd thomas wrote: > "rebootstrap and ./configure" > Did that already. > Is there a newer snapshot that yesterday 6pm BST? > > > On 27 August 2010 18:42, Serge Yuriev wrote: > >> Hello, >> >> Applied my patch on CentOS - success. >> Commented this on Jira >> >> On 27 August 2010 20:57, Michael Jerris wrote: >> > I think it is fine, but I have a recollection of trying this before and >> it >> > not solving the problem, but the problem I had was the same warning >> message >> > from bootstrap, but it did not actually cause it to fail, so I want to >> do >> > some testing on different platforms to confirm for sure. Anyone else >> > willing to test and comment on >> http://jira.freeswitch.org/browse/FSBUILD-300 >> > would be appreciated. >> >> -- >> With kind regards, >> Serge >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/aaa8f413/attachment.html From steveayre at gmail.com Fri Aug 27 11:59:19 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 27 Aug 2010 19:59:19 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> Message-ID: And you need to manually apply the patch on the Jira manually as it's not in git yet (until it's been tested). -Steve On 27 August 2010 19:57, Steven Ayre wrote: > Use git. > > > On 27 August 2010 19:27, lloyd thomas wrote: > >> "rebootstrap and ./configure" >> Did that already. >> Is there a newer snapshot that yesterday 6pm BST? >> >> >> On 27 August 2010 18:42, Serge Yuriev wrote: >> >>> Hello, >>> >>> Applied my patch on CentOS - success. >>> Commented this on Jira >>> >>> On 27 August 2010 20:57, Michael Jerris wrote: >>> > I think it is fine, but I have a recollection of trying this before and >>> it >>> > not solving the problem, but the problem I had was the same warning >>> message >>> > from bootstrap, but it did not actually cause it to fail, so I want to >>> do >>> > some testing on different platforms to confirm for sure. Anyone else >>> > willing to test and comment on >>> http://jira.freeswitch.org/browse/FSBUILD-300 >>> > would be appreciated. >>> >>> -- >>> With kind regards, >>> Serge >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/7db2dd65/attachment.html From lloydie.t at gmail.com Fri Aug 27 12:34:30 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Fri, 27 Aug 2010 20:34:30 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> Message-ID: OK, just updating to jaunty. will try after Thanks On 27 August 2010 19:59, Steven Ayre wrote: > And you need to manually apply the patch on the Jira manually as it's not > in git yet (until it's been tested). > > -Steve > > > > On 27 August 2010 19:57, Steven Ayre wrote: > >> Use git. >> >> >> On 27 August 2010 19:27, lloyd thomas wrote: >> >>> "rebootstrap and ./configure" >>> Did that already. >>> Is there a newer snapshot that yesterday 6pm BST? >>> >>> >>> On 27 August 2010 18:42, Serge Yuriev wrote: >>> >>>> Hello, >>>> >>>> Applied my patch on CentOS - success. >>>> Commented this on Jira >>>> >>>> On 27 August 2010 20:57, Michael Jerris wrote: >>>> > I think it is fine, but I have a recollection of trying this before >>>> and it >>>> > not solving the problem, but the problem I had was the same warning >>>> message >>>> > from bootstrap, but it did not actually cause it to fail, so I want to >>>> do >>>> > some testing on different platforms to confirm for sure. Anyone else >>>> > willing to test and comment on >>>> http://jira.freeswitch.org/browse/FSBUILD-300 >>>> > would be appreciated. >>>> >>>> -- >>>> With kind regards, >>>> Serge >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/6f341c33/attachment.html From d at d-man.org Fri Aug 27 14:43:41 2010 From: d at d-man.org (Darren Schreiber) Date: Fri, 27 Aug 2010 14:43:41 -0700 Subject: [Freeswitch-users] Become a FreeSWITCH NINJA - Signup for FreeSWITCH Training in New York City! Message-ID: <196C835FADC4D243A06AA032715AE72103FC@EXVMBX020-20.exch020.serverdata.net> Hi folks, After a very successful 1st training, I'm pleased to announce our second official FreeSWITCH Training in New York City! The training is a 3-day bootcamp where we'll dive deep into FreeSWITCH and all the various goodies FreeSWITCH has to offer. Scheduled from October 13th-15th in downtown Manhattan, you'll learn everything you've ever wanted to know to master FreeSWITCH, including: * Understanding configuration files and the default configuration * Call authentication and routing basics * Integration modules (mod_skypiax, mod_dingaling for Skype/GTalk/XMPP integration) * the FreeSWITCH event system * Load balancing and high availability * FreeSWITCH Internals * How to debug and troubleshoot FreeSWITCH * Building Custom C Modules * Advanced Modules Registration is open now and group discounts are available. Please contact me for more information! Hope to see you in New York! For more information, visit http://www.voipkb.com/ Sincerely, Darren Schreiber the 2600hz Project www.2600hz.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100827/02316bcd/attachment-0001.html From lloydie.t at gmail.com Fri Aug 27 18:11:01 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Sat, 28 Aug 2010 02:11:01 +0100 Subject: [Freeswitch-users] weird bootstrap error In-Reply-To: References: <978F084E-7807-4F95-96E6-47FC38F522CD@jerris.com> Message-ID: Bootstrap error fixed, but new problem arises.will create new thread On 27 August 2010 20:34, lloyd thomas wrote: > OK, just updating to jaunty. will try after > > Thanks > > > On 27 August 2010 19:59, Steven Ayre wrote: > >> And you need to manually apply the patch on the Jira manually as it's not >> in git yet (until it's been tested). >> >> -Steve >> >> >> >> On 27 August 2010 19:57, Steven Ayre wrote: >> >>> Use git. >>> >>> >>> On 27 August 2010 19:27, lloyd thomas wrote: >>> >>>> "rebootstrap and ./configure" >>>> Did that already. >>>> Is there a newer snapshot that yesterday 6pm BST? >>>> >>>> >>>> On 27 August 2010 18:42, Serge Yuriev wrote: >>>> >>>>> Hello, >>>>> >>>>> Applied my patch on CentOS - success. >>>>> Commented this on Jira >>>>> >>>>> On 27 August 2010 20:57, Michael Jerris wrote: >>>>> > I think it is fine, but I have a recollection of trying this before >>>>> and it >>>>> > not solving the problem, but the problem I had was the same warning >>>>> message >>>>> > from bootstrap, but it did not actually cause it to fail, so I want >>>>> to do >>>>> > some testing on different platforms to confirm for sure. Anyone else >>>>> > willing to test and comment on >>>>> http://jira.freeswitch.org/browse/FSBUILD-300 >>>>> > would be appreciated. >>>>> >>>>> -- >>>>> With kind regards, >>>>> Serge >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100828/4d391364/attachment.html From lloydie.t at gmail.com Fri Aug 27 18:14:23 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Sat, 28 Aug 2010 02:14:23 +0100 Subject: [Freeswitch-users] mod_spandsp error Message-ID: After my bootstrap problem, make seems to be failing with the following error. Can someone shed some light? libtool: link: ( cd ".libs" && rm -f "libspandsp.la" && ln -s "../ libspandsp.la" "libspandsp.la" ) Creating mod_spandsp.la /usr/bin/ld: cannot find -ljpeg collect2: ld returned 1 exit status quiet_libtool: link: gcc -shared .libs/mod_spandsp_la-mod_spandsp.o .libs/mod_spandsp_la-udptl.o .libs/mod_spandsp_la-mod_spandsp_fax.o .libs/mod_spandsp_la-mod_spandsp_dsp.o .libs/mod_spandsp_la-mod_spandsp_codecs.o -Wl,-rpath -Wl,/usr/src/freeswitch/.libs -Wl,-rpath -Wl,/usr/local/freeswitch/lib -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs -L/usr/src/freeswitch/libs/apr/.libs -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs -ljpeg /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt -lpthread /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -lm -lc -lncurses -pthread -Wl,-soname -Wl,mod_spandsp.so -o .libs/mod_spandsp.so make[4]: *** [mod_spandsp.la] Error 1 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100828/cb8251a5/attachment.html From brian at freeswitch.org Fri Aug 27 18:17:30 2010 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Aug 2010 20:17:30 -0500 Subject: [Freeswitch-users] mod_spandsp error In-Reply-To: References: Message-ID: <52E172B9-D906-4CD7-A2B2-26AE673EE935@freeswitch.org> install libjpeg-devel /b On Aug 27, 2010, at 8:14 PM, lloyd thomas wrote: > /usr/bin/ld: cannot find -ljpeg From sameer2k3t at gmail.com Fri Aug 27 18:24:32 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Sat, 28 Aug 2010 06:24:32 +0500 Subject: [Freeswitch-users] Skype help Message-ID: hi all i am trying to use multiple instances of same skype user, i want to run 20 instances skype version is the latest downloaded from skype website, i couldnot find skype 2.0.72 FS is the latest git my db files are located in /root/multi/01/skypeuser similarly /root/multi/02/skypeuser my multi.sh is: #start the fake X server on a given port /usr/bin/Xvfb :101 -screen 0 800x600x16 -nolisten tcp -ac & sleep 3 # start a Skype client instance that will connect to the X server above, and will login to the Skype network using the 'username password' you send to it on stdin. Here passwd5 would be the password and user5 the username su root -c "/bin/echo 'skypeuser passwd1'| DISPLAY=:101 /usr/bin/skype --dbpath=/root/multi/01 --pipelogin &" exit 0 but skype is unable to login... this user can not come online.. am i making any mistake in dbpath or somewhere else???? also when i kill the Xvfb it says FreeFontPath: FPE "unix/:7100" refcount is 2, should be 1; fixing. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100828/625748f7/attachment.html From freeswitch-list at puzzled.xs4all.nl Fri Aug 27 18:26:48 2010 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sat, 28 Aug 2010 03:26:48 +0200 Subject: [Freeswitch-users] mod_spandsp error In-Reply-To: References: Message-ID: <4C7865D8.8030803@puzzled.xs4all.nl> On 08/28/2010 03:14 AM, lloyd thomas wrote: > After my bootstrap problem, make seems to be failing with the following > error. Can someone shed some light? > > libtool: link: ( cd ".libs" && rm -f "libspandsp.la > " && ln -s "../libspandsp.la > " "libspandsp.la " ) > Creating mod_spandsp.la > /usr/bin/ld: cannot find -ljpeg ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Read this line? You need to install the appropriate devel package. Regards, Patrick From lloydie.t at gmail.com Fri Aug 27 18:31:16 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Sat, 28 Aug 2010 02:31:16 +0100 Subject: [Freeswitch-users] mod_spandsp error In-Reply-To: <52E172B9-D906-4CD7-A2B2-26AE673EE935@freeswitch.org> References: <52E172B9-D906-4CD7-A2B2-26AE673EE935@freeswitch.org> Message-ID: trying apt-get install libjpeg62-dev. let you know On 28 August 2010 02:17, Brian West wrote: > install libjpeg-devel > > /b > > On Aug 27, 2010, at 8:14 PM, lloyd thomas wrote: > > > /usr/bin/ld: cannot find -ljpeg > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100828/f2846d69/attachment.html From brokendash at gmail.com Fri Aug 27 19:31:58 2010 From: brokendash at gmail.com (broken dash) Date: Fri, 27 Aug 2010 21:31:58 -0500 Subject: [Freeswitch-users] playback options In-Reply-To: <51DD11C0-0EC2-47AF-A7B2-EEE3E909F547@jerris.com> References: <51DD11C0-0EC2-47AF-A7B2-EEE3E909F547@jerris.com> Message-ID: Thanks, never crossed my mind to handle it like that... :-) B, On Sat, Aug 21, 2010 at 1:10 PM, Michael Jerris wrote: > You could use mod_local_stream to always have this stream running and ready to go or adjust the buffer settings to your preference > > http://wiki.freeswitch.org/wiki/Mod_local_stream > http://wiki.freeswitch.org/wiki/Mod_shout > > > On Aug 19, 2010, at 9:02 PM, broken dash wrote: > >> I'm pulling shout cast streams into freeswitch using the playback >> action and I'm wondering if there is a variable like >> playback_delimeter that I could set within my dialplan/scripts that >> would essentially load up the stream and chop off a configurable >> amount of time before it essentially bridges up the audio to the >> caller? or perhaps there is a dialplan routine that could do this for >> me? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mustafa.pk at gmail.com Fri Aug 27 23:54:26 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Sat, 28 Aug 2010 11:54:26 +0500 Subject: [Freeswitch-users] Skype help In-Reply-To: References: Message-ID: TIP #1 you need to know what's going inside your fake X server, install x11vnc server, make it to serve your fake display server port e.g. :101, now connect with this vnc server using a vnc client on your desktop and see what's actually happening inside the X and what's the behavior of Skype client -- Visually. -m On Sat, Aug 28, 2010 at 6:24 AM, Sameer Khan wrote: > hi all > i am trying to use multiple instances of same skype user, i want to run 20 > instances > skype version is the latest downloaded from skype website, i couldnot find > skype 2.0.72 > FS is the latest git > > my db files are located in /root/multi/01/skypeuser > similarly /root/multi/02/skypeuser > > my multi.sh is: > > #start the fake X server on a given port > /usr/bin/Xvfb :101 -screen 0 800x600x16 -nolisten tcp -ac & > sleep 3 > > # start a Skype client instance that will connect to the X server above, > and will login to the Skype network using the 'username password' you send > to it on stdin. Here passwd5 would be the password and user5 the username > su root -c "/bin/echo 'skypeuser passwd1'| DISPLAY=:101 /usr/bin/skype > --dbpath=/root/multi/01 --pipelogin &" > > exit 0 > > but skype is unable to login... this user can not come online.. am i making > any mistake in dbpath or somewhere else???? > > also when i kill the Xvfb it says > > FreeFontPath: FPE "unix/:7100" refcount is 2, should be 1; fixing. > > thanks > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100828/d59954d1/attachment.html From lloydie.t at gmail.com Sat Aug 28 01:02:59 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Sat, 28 Aug 2010 09:02:59 +0100 Subject: [Freeswitch-users] mod_spandsp error In-Reply-To: <4C7865D8.8030803@puzzled.xs4all.nl> References: <4C7865D8.8030803@puzzled.xs4all.nl> Message-ID: apt-get install libjpeg62-dev worked for me on Juanty. Might want to include in te 'Ubuntu Quick Start' wiki On 28 August 2010 02:26, Patrick Lists wrote: > On 08/28/2010 03:14 AM, lloyd thomas wrote: > > After my bootstrap problem, make seems to be failing with the following > > error. Can someone shed some light? > > > > libtool: link: ( cd ".libs" && rm -f "libspandsp.la > > " && ln -s "../libspandsp.la > > " "libspandsp.la " ) > > Creating mod_spandsp.la > > /usr/bin/ld: cannot find -ljpeg > > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > Read this line? You need to install the appropriate devel package. > > Regards, > Patrick > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100828/c0d6d926/attachment.html From jim at k4gvo.com Sat Aug 28 01:53:24 2010 From: jim at k4gvo.com (Jim) Date: Sat, 28 Aug 2010 04:53:24 -0400 Subject: [Freeswitch-users] Automated Attendant? Message-ID: <4C78CE84.4070300@k4gvo.com> It's interesting the wiki goes to great lengths to explain the difference between an automated attendant and IVR and elsewhere in the wiki in the list of features Automated Attendant is shown, however there is no mention of usage/configuration etc. So I'm left to think that you have to roll your own Automated Attendant using IVR which brings to mind the question "why suggest there are two different things when they appear to be the same?" But my question really is, "is there an Automated Attendant in FS separate from IVR and if so can I get some details?" Thanks, Jim. From brokendash at gmail.com Sat Aug 28 03:08:37 2010 From: brokendash at gmail.com (broken dash) Date: Sat, 28 Aug 2010 05:08:37 -0500 Subject: [Freeswitch-users] Automated Attendant? In-Reply-To: <4C78CE84.4070300@k4gvo.com> References: <4C78CE84.4070300@k4gvo.com> Message-ID: One is simple, and the other could become ridiculously complex... ;-) Have you tried out any of the config GUI's for FS? What are you wishing to do with an attendent? http://wiki.freeswitch.org/wiki/IVR Cheers, Brian On Sat, Aug 28, 2010 at 3:53 AM, Jim wrote: > It's interesting the wiki goes to great lengths to explain the > difference between an automated attendant and ?IVR and elsewhere in the > wiki in the list of features Automated Attendant is shown, however there > is no mention of usage/configuration etc. > > So I'm left to think that you have to roll your own Automated Attendant > using IVR which brings to mind the question "why suggest there are two > different things when they appear to be the same?" > > But my question really is, "is there an Automated Attendant in FS > separate from IVR and if ?so can I get some details?" > > Thanks, > Jim. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jim at k4gvo.com Sat Aug 28 08:21:01 2010 From: jim at k4gvo.com (Jim) Date: Sat, 28 Aug 2010 11:21:01 -0400 Subject: [Freeswitch-users] Automated Attendant? In-Reply-To: References: <4C78CE84.4070300@k4gvo.com> Message-ID: <4C79295D.6080304@k4gvo.com> On 08/28/2010 06:08 AM, broken dash wrote: > One is simple, and the other could become ridiculously complex... ;-) > Have you tried out any of the config GUI's for FS? What are you > wishing to do with an attendent? > > http://wiki.freeswitch.org/wiki/IVR > > Cheers, > Brian > > > Disgustingly simple. I just want to have an option of leaving voice mail for one or another person (extension, user, whatever) or checking voicemail. I loaded bluebox and deleted it almost immediately. It really didn't bring anything to the table as far as I could see. I've spend 2 weeks getting my dialplan the way I wanted it and bluebox pretty much wiped the slate clean. I didn't feel like starting over. I was reading the wiki/IVR that's one place I quoted in my email. I just thought it strange that the author would go to great pains to explain that an automated attendant was significantly different than an IVR and when I went to find an automated attendant for FS, all I could find were IVR scripts. :) There's enough examples in the dialplan an the wiki for me to get started I think. If not, I can probably fall back on javascript. Programming doesn't scare me. I started doing it in 1965. :) Anyway I just want to be sure there wasn't an automated attendant function in FS that either I overlooked, or wasn't yet documented, given that an automated attendant ISN'T an IVR, or so says the wiki. Jim. From brokendash at gmail.com Sat Aug 28 10:28:38 2010 From: brokendash at gmail.com (broken dash) Date: Sat, 28 Aug 2010 12:28:38 -0500 Subject: [Freeswitch-users] Automated Attendant? In-Reply-To: <4C79295D.6080304@k4gvo.com> References: <4C78CE84.4070300@k4gvo.com> <4C79295D.6080304@k4gvo.com> Message-ID: I feel your pain... :-) I tried blue box last night and it seemed to suck from my perspective. right now I have fusionpbx installed by I find that even though I do some things within the web interface such as setup gateways etc. it also had some examples it populated into the FS scripts dir and I've been tweaking them from the console with vi. My problem is that I "thought" i had a decent understanding of JavaScript, which I did not.. hehe I keep hoping someone will create a visual configuration tool where you can piece together the logic/flow model for an IVR application but I haven't seen anything along those lines. So far I've spent several weeks looking/learning into the best way to go about creating my IVR apps an so far I'm leaning towards doing them in LUA. I would like to be able to use the Google data api stuff with my FS scripts but that is... further in you go... the deeper it all seems to get. :-) Cheers, Brian On Sat, Aug 28, 2010 at 10:21 AM, Jim wrote: > On 08/28/2010 06:08 AM, broken dash wrote: >> One is simple, and the other could become ridiculously complex... ;-) >> Have you tried out any of the config GUI's for FS? What are you >> wishing to do with an attendent? >> >> http://wiki.freeswitch.org/wiki/IVR >> >> Cheers, >> Brian >> >> >> > Disgustingly simple. ?I just want to have an option of leaving voice > mail for one or another person (extension, user, whatever) or checking > voicemail. > > I loaded bluebox and deleted it almost immediately. ?It really didn't > bring anything to the table as far as I could see. ?I've spend 2 weeks > getting my dialplan the way I wanted it and bluebox pretty much wiped > the slate clean. ?I didn't feel like starting over. > > I was reading the wiki/IVR that's one place I quoted in my email. ?I > just thought it strange that the author would go to great pains to > explain that an automated attendant was significantly different than an > IVR and when I went to find an automated attendant for FS, all I could > find were IVR scripts. :) > > There's enough examples in the dialplan an the wiki for me to get > started I think. ?If not, I can probably fall back on javascript. > Programming doesn't scare me. ?I started doing it in 1965. :) > > Anyway I just want to be sure there wasn't an automated attendant > function in FS that either I overlooked, or wasn't yet documented, given > that an automated attendant ISN'T an IVR, or so says the wiki. > > Jim. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From 12ukwn at gmail.com Sat Aug 28 11:05:11 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Sat, 28 Aug 2010 20:05:11 +0200 Subject: [Freeswitch-users] Automated Attendant? In-Reply-To: References: <4C78CE84.4070300@k4gvo.com> <4C79295D.6080304@k4gvo.com> Message-ID: <20100828200511.6da54835@anubis.defcon1> On Sat, 28 Aug 2010 12:28:38 -0500, broken dash wrote: > I feel your pain... :-) I tried blue box last night and it seemed to > suck from my perspective. right now I have fusionpbx installed by I > find that even though I do some things within the web interface such > as setup gateways etc. it also had some examples it populated into > the FS scripts dir and I've been tweaking them from the console with > vi. My problem is that I "thought" i had a decent understanding of > JavaScript, which I did not.. hehe I keep hoping someone will create > a visual configuration tool where you can piece together the > logic/flow model for an IVR application but I haven't seen anything > along those lines. So far I've spent several weeks looking/learning Me too, I tried them but when you want something specific... May be a light client isn't the solution. > into the best way to go about creating my IVR apps an so far I'm > leaning towards doing them in LUA. I would like to be able to use the > Google data api stuff with my FS scripts but that is... further in you > go... the deeper it all seems to get. :-) It seems to be the better choice because the wiki says between the same pieces of code written in javascript and LUA, LUA performed 50% more rapidly! (Quite normal, the LUA source is ridiculously small: 5.1.4 is only 1.8MB) -- JAPAN is a WONDERFUL planet -- I wonder if we'll ever reach their level of COMPARATIVE SHOPPING ... From gmaruzz at celliax.org Sat Aug 28 11:18:37 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 28 Aug 2010 20:18:37 +0200 Subject: [Freeswitch-users] Automated Attendant? In-Reply-To: <20100828200511.6da54835@anubis.defcon1> References: <4C78CE84.4070300@k4gvo.com> <4C79295D.6080304@k4gvo.com> <20100828200511.6da54835@anubis.defcon1> Message-ID: On Sat, Aug 28, 2010 at 8:05 PM, Jean-Yves F. Barbier <12ukwn at gmail.com> wrote: > > It seems to be the better choice because the wiki says between the same > pieces of code written in javascript and LUA, LUA performed 50% more > rapidly! > (Quite normal, the LUA source is ridiculously small: 5.1.4 is only 1.8MB) Also, LUA is very easy to use -giovanni > > -- > JAPAN is a WONDERFUL planet -- I wonder if we'll ever reach their level > of COMPARATIVE SHOPPING ... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From jaybinks at gmail.com Sat Aug 28 20:37:07 2010 From: jaybinks at gmail.com (Jay Binks) Date: Sun, 29 Aug 2010 13:37:07 +1000 Subject: [Freeswitch-users] mod_spandsp error In-Reply-To: References: <4C7865D8.8030803@puzzled.xs4all.nl> Message-ID: <4C80A48C-277D-4B52-A89A-7BFAD1FB0D85@gmail.com> Good idea, but remember wiki's are user updatable.. Would be great if you'd create a wiki account and update it yourself.. Jay On 28/08/2010, at 6:02 PM, lloyd thomas wrote: > apt-get install libjpeg62-dev worked for me on Juanty. Might want to include in te 'Ubuntu Quick Start' wiki > > On 28 August 2010 02:26, Patrick Lists wrote: > On 08/28/2010 03:14 AM, lloyd thomas wrote: > > After my bootstrap problem, make seems to be failing with the following > > error. Can someone shed some light? > > > > libtool: link: ( cd ".libs" && rm -f "libspandsp.la > > " && ln -s "../libspandsp.la > > " "libspandsp.la " ) > > Creating mod_spandsp.la > > /usr/bin/ld: cannot find -ljpeg > > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > Read this line? You need to install the appropriate devel package. > > Regards, > Patrick > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/34a885b1/attachment.html From rupa at rupa.com Sat Aug 28 21:07:06 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 28 Aug 2010 23:07:06 -0500 Subject: [Freeswitch-users] How to set .wav file as a voice-mail greeting? In-Reply-To: <4C77DB43.4060806@isptelecom.net> References: <4C7412F1.9000509@isptelecom.net> <4C773322.7030508@isptelecom.net> <4C77DB43.4060806@isptelecom.net> Message-ID: I didn't look closely enough. vm_prefs api is for retrieving the info, not setting it. You can either update the entry in the database directly or extend the voicemail api to support updating the information. If you aren't up to implementing the latter yourself (attach a patch to jira) maybe a bounty would get it done? On Fri, Aug 27, 2010 at 10:35 AM, Victor Chukalovskiy wrote: > Rupa, > > I see "vm_prefs" in the Database Schema section of Vocimeail WiKi. There is > a "greeting_path" variable. I guess this is what I need. > How one access FS database though? > > Thank you, V. > > > > > On -10/01/37 02:59 PM, Rupa Schomaker wrote: > > Have you looked at the vm_prefs api? > > On Thu, Aug 26, 2010 at 10:38 PM, Victor Chukalovskiy < > Victor at isptelecom.net> wrote: > >> Quick question: >> Is there an elegant way of using pre-recorder .wav file as a greeting >> for voice mail box of a user? >> My feeling is that No unless editing DB. But I'd be happy to learn I'm >> mistaken :-) >> >> Thank you, >> Victor >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -Rupa > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100828/0fa618af/attachment.html From dave.redmore at spigotsystems.com Sun Aug 29 00:01:15 2010 From: dave.redmore at spigotsystems.com (Dave Redmore) Date: Sun, 29 Aug 2010 02:01:15 -0500 (CDT) Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <18277729.641283065190545.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> Hello All, I ran into an issue today that has burned up most of my day troubleshooting. I have resolved the problem, but would really like to understand what caused it, or some of the internal Freeswitch plumbing that is at play so that I can learn something from all of this time I have invested. I have a Freeswitch server running that acts as a proxy to an account with an ITSP for doing T38 faxing. The Freeswitch server has a public IP address - there are four "users" who register simple FXS ATAs to my server and it then proxies to the ITSP using the "proxy_media" functionality. It has been working very well for the last 6 months or so. I have never had to deal with any NAT traversal issues - I just point the ATA to the IP to register and everything is great. Here is what the four users "looked" like - User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> Freeswitch Proxy User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> Freeswitch Proxy User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> Freeswitch Proxy User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> Freeswitch Proxy (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit on the same Comcast Gateway) As I said, this all worked perfectly without any need to "fiddle" with anything on any firewalls - worked right out of the box. So, today I changed out my IPCOP firewall for a pfsense firewall - and my HT-287 would no longer register. After much head-scratching, packet captures, etc. I found that I needed to set up a Static Port NAT for the port the HT-287 was using (5062) in order to get this to work. So, I see WHAT is happening, but I really want to know WHY it is happening. Here are the gory details: The sofia status of the profile looks like this - when the I have the Static Port NAT in place (details changed for security): _______________________________________________________________ Call-ID: 0e551b3c694a793c at 192.168.1.137 User: 8885554525 at 173.11.22.111 Contact: "user" Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 173.22.22.55 Port: 5060 Auth-User: 8885554525 Auth-Realm: 173.11.22.111 MWI-Account: 8885554525 at 173.11.22.111 Call-ID: 1716488819-5062-1 at 192.168.7.150 User: 8885554544 at 173.11.22.111 Contact: "user" Agent: Grandstream HT-502 V1.1B 1.0.1.63 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 98.255.0.11 Port: 5062 Auth-User: 8885554544 Auth-Realm: 173.11.22.111 MWI-Account: 8885554544 at 173.11.22.111 Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 User: 8885554549 at 173.11.22.111 Contact: "user" Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 173.11.22.99 Port: 5062 Auth-User: 8885554549 Auth-Realm: 173.11.22.111 MWI-Account: 8885554549 at 173.11.22.111 Call-ID: 1035241259-5060-1 at 10.1.10.150 User: 8885554547 at 173.11.22.111 Contact: "user" Agent: Grandstream HT-503 V1.1B 1.0.1.63 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 98.222.55.100 Port: 5060 Auth-User: 8885554547 Auth-Realm: 173.11.22.111 MWI-Account: 8885554547 at 173.11.22.111 ___________________________________________________________ The "User4" account is in red. The "Contact" field is substantially different and the "Status" indicates "Registered (UDP)", rather than "Registered (UDP-NAT)" as the others. When I do a packet capture on the external NIC interface (eth0) - I see the following when the HT-287 tries to register and the Static Port NAT is NOT in place: ___________________________________________________________________ Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Method: REGISTER Request-URI: sip:173.11.22.111:5090 Request-URI Host Part: 173.11.22.111 Request-URI Host Port: 5090 Message Header Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 Transport: UDP Sent-by Address: 10.8.11.149 Sent-by port: 5062 Branch: z9hG4bKda48f838c8689e41 From: ;tag=c8a0d452edc5ac4b SIP from address: sip:8885554549 at 173.11.22.111:5090 SIP tag: c8a0d452edc5ac4b To: Contact: Contact Binding: Supported: replaces, timer Call-ID: aa77d777bae71be6 at 10.8.11.149 CSeq: 100 REGISTER Sequence Number: 100 Method: REGISTER Expires: 3600 User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 _______________________________________________________________ When Freeswitch replies back with a "401 Unauthorized" - asking for further Auth - it replies back to port 5062 - so the packet never comes back (pfsense is looking for a packet back on port 11521 in this case). If I put the Static Port NAT in place - all is well, because the "Source" port shows as "5062" - the rest of the packet looks pretty much the same. Now, here is a packet coming from one of the other Users - this one comes through a DD-WRT router - here we see that the Source Port is 5060 : _________________________________________________________________ Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Method: REGISTER Request-URI: sip:173.11.22.111:5090 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b Transport: UDP Sent-by Address: 192.168.1.137 Branch: z9hG4bK665bc67a1c64292b From: "fax" ;tag=8dc68b35111c4261 To: Contact: Contact Binding: Call-ID: 0e551b3c694a793c at 192.168.1.137 CSeq: 503 REGISTER Sequence Number: 503 Method: REGISTER Expires: 3600 User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 ______________________________________________________________________ Here is one more packet coming from a Comcast/SMC Router - again, the source port is correct: ______________________________________________________________________ Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport Transport: UDP Sent-by Address: 10.1.10.150 Sent-by port: 5060 Branch: z9hG4bK58981045 RPort: rport From: ;tag=138706651 To: Call-ID: 1035241259-5060-1 at 10.1.10.150 CSeq: 79875 REGISTER Sequence Number: 79875 Method: REGISTER Contact: ;reg-id=1;+sip.instance="" Contact Binding: ;reg-id=1;+sip.instance="" Max-Forwards: 70 User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 Supported: path Expires: 300 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 ___________________________________________________________ So, here are my questions: - Why is the Sofia Status so much different for the registration coming through the pfSense firewall. It looks like it doesn't get tagged as being NAT'd and the "Contact" info is much less. - Do most modern routers automatically Static Port NAT any SIP traffic? Both DD-WRT and SMC routers appear to be doing this - and not just on a simple Port bases (UDP 5060 only), because one of these examples is on 5062. Are these "SIP aware" firewalls that are doing this automatically, as the IPCOP did before? - Is the extra "Contact" data in the last packet example different because it is a different UA (HT-503 rather than an HT-287) - Is Freeswitch not flagging the registration from my office (User4) as being NAT'd because it is coming in on the same subnet as the interface Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and pfsense is at 173.11.22.99)? Sorry for this terribly long posting - I'm just very curious to understand what is going on here, now that I have collected all this information. Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/6f7db29c/attachment-0001.html From david.ponzone at ipeva.fr Sun Aug 29 01:15:19 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Sun, 29 Aug 2010 10:15:19 +0200 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> References: <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: <1A8DB11D-897D-41FE-8359-E7E05233C200@ipeva.fr> Dave, quite quickly, it's obvious your FreeSWITCH is no longer able to detect that your HT-287 is behind NAT. One possiblity is that the rport is missing from the REGISTER. Perhaps your pfsense is messing with it ? So to start, I would recommend you take a trace when the packet enters pfsense and when it goes out to your proxy, and compare them to see any differences. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 29/08/2010 ? 09:01, Dave Redmore a ?crit : > Hello All, > > I ran into an issue today that has burned up most of my day > troubleshooting. I have resolved the problem, but would really like > to understand what caused it, or some of the internal Freeswitch > plumbing that is at play so that I can learn something from all of > this time I have invested. > > I have a Freeswitch server running that acts as a proxy to an > account with an ITSP for doing T38 faxing. The Freeswitch server > has a public IP address - there are four "users" who register simple > FXS ATAs to my server and it then proxies to the ITSP using the > "proxy_media" functionality. It has been working very well for the > last 6 months or so. I have never had to deal with any NAT > traversal issues - I just point the ATA to the IP to register and > everything is great. > > Here is what the four users "looked" like - > > User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> > Freeswitch Proxy > User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> > Freeswitch Proxy > User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet - > > Freeswitch Proxy > User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway > -> Freeswitch Proxy > > (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy > sit on the same Comcast Gateway) > > As I said, this all worked perfectly without any need to "fiddle" > with anything on any firewalls - worked right out of the box. > > So, today I changed out my IPCOP firewall for a pfsense firewall - > and my HT-287 would no longer register. > > After much head-scratching, packet captures, etc. I found that I > needed to set up a Static Port NAT for the port the HT-287 was using > (5062) in order to get this to work. > > So, I see WHAT is happening, but I really want to know WHY it is > happening. > > Here are the gory details: > > The sofia status of the profile looks like this - when the I have > the Static Port NAT in place (details changed for security): > > _______________________________________________________________ > Call-ID: 0e551b3c694a793c at 192.168.1.137 > User: 8885554525 at 173.11.22.111 > Contact: "user" > > Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 173.22.22.55 > Port: 5060 > Auth-User: 8885554525 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554525 at 173.11.22.111 > > Call-ID: 1716488819-5062-1 at 192.168.7.150 > User: 8885554544 at 173.11.22.111 > Contact: "user" ; fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> > Agent: Grandstream HT-502 V1.1B 1.0.1.63 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 98.255.0.11 > Port: 5062 > Auth-User: 8885554544 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554544 at 173.11.22.111 > > Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 > User: 8885554549 at 173.11.22.111 > Contact: "user" > Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 173.11.22.99 > Port: 5062 > Auth-User: 8885554549 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554549 at 173.11.22.111 > > Call-ID: 1035241259-5060-1 at 10.1.10.150 > User: 8885554547 at 173.11.22.111 > Contact: "user" _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> > Agent: Grandstream HT-503 V1.1B 1.0.1.63 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 98.222.55.100 > Port: 5060 > Auth-User: 8885554547 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554547 at 173.11.22.111 > ___________________________________________________________ > > The "User4" account is in red. The "Contact" field is substantially > different and the "Status" indicates "Registered (UDP)", rather than > "Registered (UDP-NAT)" as the others. > > When I do a packet capture on the external NIC interface (eth0) - I > see the following when the HT-287 tries to register and the Static > Port NAT is NOT in place: > > ___________________________________________________________________ > Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Method: REGISTER > Request-URI: sip:173.11.22.111:5090 > Request-URI Host Part: 173.11.22.111 > Request-URI Host Port: 5090 > Message Header > Via: SIP/2.0/UDP > 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 > Transport: UDP > Sent-by Address: 10.8.11.149 > Sent-by port: 5062 > Branch: z9hG4bKda48f838c8689e41 > From: ;tag=c8a0d452edc5ac4b > SIP from address: sip:8885554549 at 173.11.22.111:5090 > SIP tag: c8a0d452edc5ac4b > To: > Contact: > Contact Binding: > Supported: replaces, timer > Call-ID: aa77d777bae71be6 at 10.8.11.149 > CSeq: 100 REGISTER > Sequence Number: 100 > Method: REGISTER > Expires: 3600 > User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > _______________________________________________________________ > > When Freeswitch replies back with a "401 Unauthorized" - asking for > further Auth - it replies back to port 5062 - so the packet never > comes back (pfsense is looking for a packet back on port 11521 in > this case). > > If I put the Static Port NAT in place - all is well, because the > "Source" port shows as "5062" - the rest of the packet looks pretty > much the same. > > Now, here is a packet coming from one of the other Users - this one > comes through a DD-WRT router - here we see that the Source Port is > 5060 : > > _________________________________________________________________ > Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Method: REGISTER > Request-URI: sip:173.11.22.111:5090 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b > Transport: UDP > Sent-by Address: 192.168.1.137 > Branch: z9hG4bK665bc67a1c64292b > From: "fax" 8885554525 at 173.11.22.111:5090>;tag=8dc68b35111c4261 > To: > Contact: > Contact Binding: > Call-ID: 0e551b3c694a793c at 192.168.1.137 > CSeq: 503 REGISTER > Sequence Number: 503 > Method: REGISTER > Expires: 3600 > User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > ______________________________________________________________________ > > Here is one more packet coming from a Comcast/SMC Router - again, > the source port is correct: > > ______________________________________________________________________ > Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Message Header > Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport > Transport: UDP > Sent-by Address: 10.1.10.150 > Sent-by port: 5060 > Branch: z9hG4bK58981045 > RPort: rport > From: 8885554547 at 173.11.22.111:5090;user=phone>;tag=138706651 > To: > Call-ID: 1035241259-5060-1 at 10.1.10.150 > CSeq: 79875 REGISTER > Sequence Number: 79875 > Method: REGISTER > Contact: ;reg- > id=1;+sip.instance="" > Contact Binding: >;reg-id=1;+sip.instance=" >" > Max-Forwards: 70 > User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 > Supported: path > Expires: 300 > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, > INFO, REFER, UPDATE > Content-Length: 0 > ___________________________________________________________ > > So, here are my questions: > > - Why is the Sofia Status so much different for the registration > coming through the pfSense firewall. It looks like it doesn't get > tagged as being NAT'd and the "Contact" info is much less. > > - Do most modern routers automatically Static Port NAT any SIP > traffic? Both DD-WRT and SMC routers appear to be doing this - and > not just on a simple Port bases (UDP 5060 only), because one of > these examples is on 5062. Are these "SIP aware" firewalls that are > doing this automatically, as the IPCOP did before? > > - Is the extra "Contact" data in the last packet example different > because it is a different UA (HT-503 rather than an HT-287) > > - Is Freeswitch not flagging the registration from my office (User4) > as being NAT'd because it is coming in on the same subnet as the > interface Freeswitch received the packet on (Freeswitch is at > 173.11.22.111 and pfsense is at 173.11.22.99)? > > Sorry for this terribly long posting - I'm just very curious to > understand what is going on here, now that I have collected all this > information. > > Thanks, > > Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/da9cf23b/attachment-0001.html From brokendash at gmail.com Sun Aug 29 02:29:15 2010 From: brokendash at gmail.com (broken dash) Date: Sun, 29 Aug 2010 04:29:15 -0500 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <1A8DB11D-897D-41FE-8359-E7E05233C200@ipeva.fr> References: <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> <1A8DB11D-897D-41FE-8359-E7E05233C200@ipeva.fr> Message-ID: I came across this sipx troubleshooting faq talking about how pfsense's port radomization jacks things up and they go on to describe how you solved your problem. It seems almost certain that it's the cause, but surely the linux port randomization would be equally as problematic as the bsd version... http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration I wonder if u installed the freeswitch package on your pfsense firewall and configured it be a B2BUA if that would react in the same way... Cheers, Brian On Sun, Aug 29, 2010 at 3:15 AM, David Ponzone wrote: > Dave, > quite quickly, it's obvious your FreeSWITCH is no longer able to detect that > your HT-287 is behind NAT. > One possiblity is that the rport is missing from the REGISTER. > Perhaps your pfsense is messing with it ? > So to start, I would recommend you take a trace when the packet enters > pfsense and when it goes out to your proxy, and compare them to see any > differences. > David Ponzone ?Direction Technique > email: david.ponzone at ipeva.fr > tel: ? ? ?01 74 03 18 97 > gsm: ? 06 66 98 76 34 > Service Client?IPeva > tel: ? ? ?0811 46 26 26 > www.ipeva.fr? -? ?www.ipeva-studio.com > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autoris?e est interdite. Tout message ?lectronique est susceptible > d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. > > > > Le 29/08/2010 ? 09:01, Dave Redmore a ?crit : > > Hello All, > > I ran into an issue today that has burned up most of my day > troubleshooting.? I have resolved the problem, but would really like to > understand what caused it, or some of the internal Freeswitch plumbing that > is at play so that I can learn something from all of this time I have > invested. > > I have a Freeswitch server running that acts as a proxy to an account with > an ITSP for doing T38 faxing.? The Freeswitch server has a public IP address > - there are four "users" who register simple FXS ATAs to my server and it > then proxies to the ITSP using the "proxy_media" functionality.? It has been > working very well for the last 6 months or so.? I have never had to deal > with any NAT traversal issues - I just point the ATA to the IP to register > and everything is great. > > Here is what the four users "looked" like - > > User1 :? Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> Freeswitch > Proxy > User2 :? Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> Freeswitch > Proxy > User3 :? Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> > Freeswitch Proxy > User4 :? Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> > Freeswitch Proxy > > (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit on > the same Comcast Gateway) > > As I said, this all worked perfectly without any need to "fiddle" with > anything on any firewalls - worked right out of the box. > > So, today I changed out my IPCOP firewall for a pfsense firewall - and my > HT-287 would no longer register. > > After much head-scratching, packet captures, etc. I found that I needed to > set up a Static Port NAT for the port the HT-287 was using (5062) in order > to get this to work. > > So, I see WHAT is happening, but I really want to know WHY it is happening. > > Here are the gory details: > > The sofia status of the profile looks like this - when the I have the Static > Port NAT in place (details changed for security): > > _______________________________________________________________ > Call-ID:????????0e551b3c694a793c at 192.168.1.137 > User:???????????8885554525 at 173.11.22.111 > Contact:??????? "user" > > Agent:????????? Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Status:???????? Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) > Host:?????????? 173-11-22-111-illinois.hfc.comcastbusiness.net > IP:???????????? 173.22.22.55 > Port:?????????? 5060 > Auth-User:????? 8885554525 > Auth-Realm:???? 173.11.22.111 > MWI-Account:????8885554525 at 173.11.22.111 > > Call-ID:????????1716488819-5062-1 at 192.168.7.150 > User:???????????8885554544 at 173.11.22.111 > Contact:??????? "user" > fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> > Agent:????????? Grandstream HT-502? V1.1B 1.0.1.63 > Status:???????? Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) > Host:?????????? 173-11-22-111-illinois.hfc.comcastbusiness.net > IP:???????????? 98.255.0.11 > Port:?????????? 5062 > Auth-User:????? 8885554544 > Auth-Realm:???? 173.11.22.111 > MWI-Account:????8885554544 at 173.11.22.111 > > Call-ID:????????090ee80e1a0ec9ed at 10.8.11.149 > User:???????????8885554549 at 173.11.22.111 > Contact:??????? "user" > Agent:????????? Grandstream HT287 1.1.0.45 DevId 000b82127390 > Status:???????? Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) > Host:?????????? 173-11-22-111-illinois.hfc.comcastbusiness.net > IP:???????????? 173.11.22.99 > Port:?????????? 5062 > Auth-User:????? 8885554549 > Auth-Realm:???? 173.11.22.111 > MWI-Account:????8885554549 at 173.11.22.111 > > Call-ID:????????1035241259-5060-1 at 10.1.10.150 > User:???????????8885554547 at 173.11.22.111 > Contact:??????? "user" > _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> > Agent:????????? Grandstream HT-503? V1.1B 1.0.1.63 > Status:???????? Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) > Host:?????????? 173-11-22-111-illinois.hfc.comcastbusiness.net > IP:???????????? 98.222.55.100 > Port:?????????? 5060 > Auth-User:????? 8885554547 > Auth-Realm:???? 173.11.22.111 > MWI-Account:????8885554547 at 173.11.22.111 > ___________________________________________________________ > > The "User4" account is in red.? The "Contact" field is substantially > different and the "Status" indicates "Registered (UDP)", rather than > "Registered (UDP-NAT)" as the others. > > When I do a packet capture on the external NIC interface (eth0) - I see the > following when the HT-287 tries to register and the Static Port NAT is NOT > in place: > > ___________________________________________________________________ > Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 > (173.11.22.111) > User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) > Session Initiation Protocol > ??? Request-Line: REGISTER?sip:173.11.22.111:5090?SIP/2.0 > ??????? Method: REGISTER > ??????? Request-URI:?sip:173.11.22.111:5090 > ??????????? Request-URI Host Part: 173.11.22.111 > ??????????? Request-URI Host Port: 5090 > ??? Message Header > ??????? Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 > ??????????? Transport: UDP > ??????????? Sent-by Address: 10.8.11.149 > ??????????? Sent-by port: 5062 > ??????????? Branch: z9hG4bKda48f838c8689e41 > ??????? From: ;tag=c8a0d452edc5ac4b > ??????????? SIP from address:?sip:8885554549 at 173.11.22.111:5090 > ??????????? SIP tag: c8a0d452edc5ac4b > ??????? To: > ??????? Contact: > ??????????? Contact Binding: > ??????? Supported: replaces, timer > ??????? Call-ID:?aa77d777bae71be6 at 10.8.11.149 > ??????? CSeq: 100 REGISTER > ??????????? Sequence Number: 100 > ??????????? Method: REGISTER > ??????? Expires: 3600 > ??????? User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > ??????? Max-Forwards: 70 > ??????? Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > ??????? Content-Length: 0 > _______________________________________________________________ > > When Freeswitch replies back with a "401 Unauthorized" - asking for further > Auth - it replies back to port 5062 - so the packet never comes back > (pfsense is looking for a packet back on port 11521 in this case). > > If I put the Static Port NAT in place - all is well, because the "Source" > port shows as "5062" - the rest of the packet looks pretty much the same. > > Now, here is a packet coming from one of the other Users - this one comes > through a DD-WRT router - here we see that the Source Port is 5060 : > > _________________________________________________________________ > Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 > (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > ??? Request-Line: REGISTER?sip:173.11.22.111:5090?SIP/2.0 > ??????? Method: REGISTER > ??????? Request-URI:?sip:173.11.22.111:5090 > ??????? [Resent Packet: False] > ??? Message Header > ??????? Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b > ??????????? Transport: UDP > ??????????? Sent-by Address: 192.168.1.137 > ??????????? Branch: z9hG4bK665bc67a1c64292b > ??????? From: "fax" ;tag=8dc68b35111c4261 > ??????? To: > ??????? Contact: > ??????????? Contact Binding: > ??????? Call-ID:?0e551b3c694a793c at 192.168.1.137 > ??????? CSeq: 503 REGISTER > ??????????? Sequence Number: 503 > ??????????? Method: REGISTER > ??????? Expires: 3600 > ??????? User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > ??????? Max-Forwards: 70 > ??????? Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > ??????? Content-Length: 0 > ______________________________________________________________________ > > Here is one more packet coming from a Comcast/SMC Router - again, the source > port is correct: > > ______________________________________________________________________ > ?Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: 173.11.22.111 > (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > ??? Request-Line: REGISTER?sip:173.11.22.111:5090?SIP/2.0 > ??? Message Header > ??????? Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport > ??????????? Transport: UDP > ??????????? Sent-by Address: 10.1.10.150 > ??????????? Sent-by port: 5060 > ??????????? Branch: z9hG4bK58981045 > ??????????? RPort: rport > ??????? From: ;tag=138706651 > ??????? To: > ??????? Call-ID:?1035241259-5060-1 at 10.1.10.150 > ??????? CSeq: 79875 REGISTER > ??????????? Sequence Number: 79875 > ??????????? Method: REGISTER > ??????? Contact: > ;reg-id=1;+sip.instance="" > ??????????? Contact Binding: > ;reg-id=1;+sip.instance="" > ??????? Max-Forwards: 70 > ??????? User-Agent: Grandstream HT-503? V1.1B 1.0.1.63 > ??????? Supported: path > ??????? Expires: 300 > ??????? Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE > ??????? Content-Length: 0 > ___________________________________________________________ > > So, here are my questions: > > - Why is the Sofia Status so much different for the registration coming > through the pfSense firewall.? It looks like it doesn't get tagged as being > NAT'd and the "Contact" info is much less. > > - Do most modern routers automatically Static Port NAT any SIP traffic? > Both DD-WRT and SMC routers appear to be doing this - and not just on a > simple Port bases (UDP 5060 only), because one of these examples is on > 5062.? Are these "SIP aware" firewalls that are doing this automatically, > as? the IPCOP did before? > > - Is the extra "Contact" data in the last packet example different because > it is a different UA (HT-503 rather than an HT-287) > > - Is Freeswitch not flagging the registration from my office (User4) as > being NAT'd because it is coming in on the same subnet as the interface > Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and > pfsense is at 173.11.22.99)? > > Sorry for this terribly long posting - I'm just very curious to understand > what is going on here, now that I have collected all this information. > > Thanks, > > Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From david.ponzone at ipeva.fr Sun Aug 29 02:46:24 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Sun, 29 Aug 2010 11:46:24 +0200 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: References: <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> <1A8DB11D-897D-41FE-8359-E7E05233C200@ipeva.fr> Message-ID: <78EDAF2E-7CE8-4DF1-A999-325F0B64CC12@ipeva.fr> Brian, I am not sure. Dave said its device cannot register anymore. In your doc, it says the INVITE won't work because of the port mismatch with the REGISTER. BTW, what kind of crappy firewall does that... A firewall is supposed to keep the same source port during the lifetime of the translation, which can be a very long time if you send regular keepalives. To change it between a REGISTER and an INVITE, a firewall would need to have some kind of SIP ALG, and that's dodgy. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 29/08/2010 ? 11:29, broken dash a ?crit : > I came across this sipx troubleshooting faq talking about how > pfsense's port radomization jacks things up and they go on to describe > how you solved your problem. It seems almost certain that it's the > cause, but surely the linux port randomization would be equally as > problematic as the bsd version... > > http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration > > I wonder if u installed the freeswitch package on your pfsense > firewall and configured it be a B2BUA if that would react in the same > way... > > > > Cheers, > Brian > > > On Sun, Aug 29, 2010 at 3:15 AM, David Ponzone > wrote: >> Dave, >> quite quickly, it's obvious your FreeSWITCH is no longer able to >> detect that >> your HT-287 is behind NAT. >> One possiblity is that the rport is missing from the REGISTER. >> Perhaps your pfsense is messing with it ? >> So to start, I would recommend you take a trace when the packet >> enters >> pfsense and when it goes out to your proxy, and compare them to see >> any >> differences. >> David Ponzone Direction Technique >> email: david.ponzone at ipeva.fr >> tel: 01 74 03 18 97 >> gsm: 06 66 98 76 34 >> Service Client IPeva >> tel: 0811 46 26 26 >> www.ipeva.fr - www.ipeva-studio.com >> Ce message et toutes les pi?ces jointes sont confidentiels et >> ?tablis ? >> l'intention exclusive de ses destinataires. Toute utilisation ou >> diffusion >> non autoris?e est interdite. Tout message ?lectronique est >> susceptible >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce >> message s'il >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire >> de ce >> message, merci de le d?truire imm?diatement et d'avertir >> l'exp?diteur. >> >> >> >> Le 29/08/2010 ? 09:01, Dave Redmore a ?crit : >> >> Hello All, >> >> I ran into an issue today that has burned up most of my day >> troubleshooting. I have resolved the problem, but would really >> like to >> understand what caused it, or some of the internal Freeswitch >> plumbing that >> is at play so that I can learn something from all of this time I have >> invested. >> >> I have a Freeswitch server running that acts as a proxy to an >> account with >> an ITSP for doing T38 faxing. The Freeswitch server has a public >> IP address >> - there are four "users" who register simple FXS ATAs to my server >> and it >> then proxies to the ITSP using the "proxy_media" functionality. It >> has been >> working very well for the last 6 months or so. I have never had to >> deal >> with any NAT traversal issues - I just point the ATA to the IP to >> register >> and everything is great. >> >> Here is what the four users "looked" like - >> >> User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> >> Freeswitch >> Proxy >> User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> >> Freeswitch >> Proxy >> User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet >> -> >> Freeswitch Proxy >> User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast >> Gateway -> >> Freeswitch Proxy >> >> (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy >> sit on >> the same Comcast Gateway) >> >> As I said, this all worked perfectly without any need to "fiddle" >> with >> anything on any firewalls - worked right out of the box. >> >> So, today I changed out my IPCOP firewall for a pfsense firewall - >> and my >> HT-287 would no longer register. >> >> After much head-scratching, packet captures, etc. I found that I >> needed to >> set up a Static Port NAT for the port the HT-287 was using (5062) >> in order >> to get this to work. >> >> So, I see WHAT is happening, but I really want to know WHY it is >> happening. >> >> Here are the gory details: >> >> The sofia status of the profile looks like this - when the I have >> the Static >> Port NAT in place (details changed for security): >> >> _______________________________________________________________ >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> User: 8885554525 at 173.11.22.111 >> Contact: "user" >> > > >> Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 173.22.22.55 >> Port: 5060 >> Auth-User: 8885554525 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554525 at 173.11.22.111 >> >> Call-ID: 1716488819-5062-1 at 192.168.7.150 >> User: 8885554544 at 173.11.22.111 >> Contact: "user" >> > fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> >> Agent: Grandstream HT-502 V1.1B 1.0.1.63 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 98.255.0.11 >> Port: 5062 >> Auth-User: 8885554544 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554544 at 173.11.22.111 >> >> Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 >> User: 8885554549 at 173.11.22.111 >> Contact: "user" >> Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 173.11.22.99 >> Port: 5062 >> Auth-User: 8885554549 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554549 at 173.11.22.111 >> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> User: 8885554547 at 173.11.22.111 >> Contact: "user" >> > _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> >> Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 98.222.55.100 >> Port: 5060 >> Auth-User: 8885554547 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554547 at 173.11.22.111 >> ___________________________________________________________ >> >> The "User4" account is in red. The "Contact" field is substantially >> different and the "Status" indicates "Registered (UDP)", rather than >> "Registered (UDP-NAT)" as the others. >> >> When I do a packet capture on the external NIC interface (eth0) - I >> see the >> following when the HT-287 tries to register and the Static Port NAT >> is NOT >> in place: >> >> ___________________________________________________________________ >> Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: >> 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 >> (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Method: REGISTER >> Request-URI: sip:173.11.22.111:5090 >> Request-URI Host Part: 173.11.22.111 >> Request-URI Host Port: 5090 >> Message Header >> Via: SIP/2.0/UDP >> 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 >> Transport: UDP >> Sent-by Address: 10.8.11.149 >> Sent-by port: 5062 >> Branch: z9hG4bKda48f838c8689e41 >> From: > 8885554549 at 173.11.22.111:5090>;tag=c8a0d452edc5ac4b >> SIP from address: sip:8885554549 at 173.11.22.111:5090 >> SIP tag: c8a0d452edc5ac4b >> To: >> Contact: >> Contact Binding: >> Supported: replaces, timer >> Call-ID: aa77d777bae71be6 at 10.8.11.149 >> CSeq: 100 REGISTER >> Sequence Number: 100 >> Method: REGISTER >> Expires: 3600 >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> Max-Forwards: 70 >> Allow: >> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> Content-Length: 0 >> _______________________________________________________________ >> >> When Freeswitch replies back with a "401 Unauthorized" - asking for >> further >> Auth - it replies back to port 5062 - so the packet never comes back >> (pfsense is looking for a packet back on port 11521 in this case). >> >> If I put the Static Port NAT in place - all is well, because the >> "Source" >> port shows as "5062" - the rest of the packet looks pretty much the >> same. >> >> Now, here is a packet coming from one of the other Users - this one >> comes >> through a DD-WRT router - here we see that the Source Port is 5060 : >> >> _________________________________________________________________ >> Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: >> 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Method: REGISTER >> Request-URI: sip:173.11.22.111:5090 >> [Resent Packet: False] >> Message Header >> Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b >> Transport: UDP >> Sent-by Address: 192.168.1.137 >> Branch: z9hG4bK665bc67a1c64292b >> From: "fax" > 8885554525 at 173.11.22.111:5090>;tag=8dc68b35111c4261 >> To: >> Contact: >> Contact Binding: >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> CSeq: 503 REGISTER >> Sequence Number: 503 >> Method: REGISTER >> Expires: 3600 >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> Max-Forwards: 70 >> Allow: >> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> Content-Length: 0 >> ______________________________________________________________________ >> >> Here is one more packet coming from a Comcast/SMC Router - again, >> the source >> port is correct: >> >> ______________________________________________________________________ >> Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: >> 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Message Header >> Via: SIP/2.0/UDP >> 10.1.10.150:5060;branch=z9hG4bK58981045;rport >> Transport: UDP >> Sent-by Address: 10.1.10.150 >> Sent-by port: 5060 >> Branch: z9hG4bK58981045 >> RPort: rport >> From: > 8885554547 at 173.11.22.111:5090;user=phone>;tag=138706651 >> To: >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> CSeq: 79875 REGISTER >> Sequence Number: 79875 >> Method: REGISTER >> Contact: >> ;reg- >> id=1;+sip.instance="" >> Contact Binding: >> ;reg- >> id=1;+sip.instance="" >> Max-Forwards: 70 >> User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> Supported: path >> Expires: 300 >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, >> NOTIFY, INFO, >> REFER, UPDATE >> Content-Length: 0 >> ___________________________________________________________ >> >> So, here are my questions: >> >> - Why is the Sofia Status so much different for the registration >> coming >> through the pfSense firewall. It looks like it doesn't get tagged >> as being >> NAT'd and the "Contact" info is much less. >> >> - Do most modern routers automatically Static Port NAT any SIP >> traffic? >> Both DD-WRT and SMC routers appear to be doing this - and not just >> on a >> simple Port bases (UDP 5060 only), because one of these examples is >> on >> 5062. Are these "SIP aware" firewalls that are doing this >> automatically, >> as the IPCOP did before? >> >> - Is the extra "Contact" data in the last packet example different >> because >> it is a different UA (HT-503 rather than an HT-287) >> >> - Is Freeswitch not flagging the registration from my office >> (User4) as >> being NAT'd because it is coming in on the same subnet as the >> interface >> Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and >> pfsense is at 173.11.22.99)? >> >> Sorry for this terribly long posting - I'm just very curious to >> understand >> what is going on here, now that I have collected all this >> information. >> >> Thanks, >> >> Dave >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/231dbbd7/attachment-0001.html From tgraziano at myitdepartment.net Sun Aug 29 04:40:29 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Sun, 29 Aug 2010 07:40:29 -0400 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> References: <18277729.641283065190545.JavaMail.root@zimbra1.spigotsystems.com> <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: Ipcop has a similar setting to pfsense. You probably missed it. MOST FIREWALLS do not use static port NAT. The default rules for pfsense (and packages) for port 5060 should be removed. On your outbound rule for your LAN static port nat needs to be enabled. Once you do that recreate the nat rules AND remove the siproxd package by default. This is really a pfsense firewall question, it is clear static port was not enabled so the source port was re-written because that is what MOST firewalls do by default. On 8/29/10, Dave Redmore wrote: > Hello All, > > I ran into an issue today that has burned up most of my day troubleshooting. > I have resolved the problem, but would really like to understand what caused > it, or some of the internal Freeswitch plumbing that is at play so that I > can learn something from all of this time I have invested. > > I have a Freeswitch server running that acts as a proxy to an account with > an ITSP for doing T38 faxing. The Freeswitch server has a public IP address > - there are four "users" who register simple FXS ATAs to my server and it > then proxies to the ITSP using the "proxy_media" functionality. It has been > working very well for the last 6 months or so. I have never had to deal with > any NAT traversal issues - I just point the ATA to the IP to register and > everything is great. > > Here is what the four users "looked" like - > > User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> Freeswitch > Proxy > User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> Freeswitch > Proxy > User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> > Freeswitch Proxy > User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> > Freeswitch Proxy > > (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit on > the same Comcast Gateway) > > As I said, this all worked perfectly without any need to "fiddle" with > anything on any firewalls - worked right out of the box. > > So, today I changed out my IPCOP firewall for a pfsense firewall - and my > HT-287 would no longer register. > > After much head-scratching, packet captures, etc. I found that I needed to > set up a Static Port NAT for the port the HT-287 was using (5062) in order > to get this to work. > > So, I see WHAT is happening, but I really want to know WHY it is happening. > > Here are the gory details: > > The sofia status of the profile looks like this - when the I have the Static > Port NAT in place (details changed for security): > > _______________________________________________________________ > Call-ID: 0e551b3c694a793c at 192.168.1.137 > User: 8885554525 at 173.11.22.111 > Contact: "user" > > Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 173.22.22.55 > Port: 5060 > Auth-User: 8885554525 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554525 at 173.11.22.111 > > Call-ID: 1716488819-5062-1 at 192.168.7.150 > User: 8885554544 at 173.11.22.111 > Contact: "user" fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> > Agent: Grandstream HT-502 V1.1B 1.0.1.63 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 98.255.0.11 > Port: 5062 > Auth-User: 8885554544 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554544 at 173.11.22.111 > > Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 > User: 8885554549 at 173.11.22.111 > Contact: "user" > Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 173.11.22.99 > Port: 5062 > Auth-User: 8885554549 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554549 at 173.11.22.111 > > Call-ID: 1035241259-5060-1 at 10.1.10.150 > User: 8885554547 at 173.11.22.111 > Contact: "user" _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> > Agent: Grandstream HT-503 V1.1B 1.0.1.63 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 98.222.55.100 > Port: 5060 > Auth-User: 8885554547 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554547 at 173.11.22.111 > ___________________________________________________________ > > The "User4" account is in red. The "Contact" field is substantially > different and the "Status" indicates "Registered (UDP)", rather than > "Registered (UDP-NAT)" as the others. > > When I do a packet capture on the external NIC interface (eth0) - I see the > following when the HT-287 tries to register and the Static Port NAT is NOT > in place: > > ___________________________________________________________________ > Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 > (173.11.22.111) > User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Method: REGISTER > Request-URI: sip:173.11.22.111:5090 > Request-URI Host Part: 173.11.22.111 > Request-URI Host Port: 5090 > Message Header > Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 > Transport: UDP > Sent-by Address: 10.8.11.149 > Sent-by port: 5062 > Branch: z9hG4bKda48f838c8689e41 > From: ;tag=c8a0d452edc5ac4b > SIP from address: sip:8885554549 at 173.11.22.111:5090 > SIP tag: c8a0d452edc5ac4b > To: > Contact: > Contact Binding: > Supported: replaces, timer > Call-ID: aa77d777bae71be6 at 10.8.11.149 > CSeq: 100 REGISTER > Sequence Number: 100 > Method: REGISTER > Expires: 3600 > User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > _______________________________________________________________ > > When Freeswitch replies back with a "401 Unauthorized" - asking for further > Auth - it replies back to port 5062 - so the packet never comes back > (pfsense is looking for a packet back on port 11521 in this case). > > If I put the Static Port NAT in place - all is well, because the "Source" > port shows as "5062" - the rest of the packet looks pretty much the same. > > Now, here is a packet coming from one of the other Users - this one comes > through a DD-WRT router - here we see that the Source Port is 5060 : > > _________________________________________________________________ > Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 > (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Method: REGISTER > Request-URI: sip:173.11.22.111:5090 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b > Transport: UDP > Sent-by Address: 192.168.1.137 > Branch: z9hG4bK665bc67a1c64292b > From: "fax" ;tag=8dc68b35111c4261 > To: > Contact: > Contact Binding: > Call-ID: 0e551b3c694a793c at 192.168.1.137 > CSeq: 503 REGISTER > Sequence Number: 503 > Method: REGISTER > Expires: 3600 > User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > ______________________________________________________________________ > > Here is one more packet coming from a Comcast/SMC Router - again, the source > port is correct: > > ______________________________________________________________________ > Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: 173.11.22.111 > (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Message Header > Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport > Transport: UDP > Sent-by Address: 10.1.10.150 > Sent-by port: 5060 > Branch: z9hG4bK58981045 > RPort: rport > From: ;tag=138706651 > To: > Call-ID: 1035241259-5060-1 at 10.1.10.150 > CSeq: 79875 REGISTER > Sequence Number: 79875 > Method: REGISTER > Contact: > ;reg-id=1;+sip.instance="" > Contact Binding: > ;reg-id=1;+sip.instance="" > Max-Forwards: 70 > User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 > Supported: path > Expires: 300 > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, > UPDATE > Content-Length: 0 > ___________________________________________________________ > > So, here are my questions: > > - Why is the Sofia Status so much different for the registration coming > through the pfSense firewall. It looks like it doesn't get tagged as being > NAT'd and the "Contact" info is much less. > > - Do most modern routers automatically Static Port NAT any SIP traffic? Both > DD-WRT and SMC routers appear to be doing this - and not just on a simple > Port bases (UDP 5060 only), because one of these examples is on 5062. Are > these "SIP aware" firewalls that are doing this automatically, as the IPCOP > did before? > > - Is the extra "Contact" data in the last packet example different because > it is a different UA (HT-503 rather than an HT-287) > > - Is Freeswitch not flagging the registration from my office (User4) as > being NAT'd because it is coming in on the same subnet as the interface > Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and > pfsense is at 173.11.22.99)? > > Sorry for this terribly long posting - I'm just very curious to understand > what is going on here, now that I have collected all this information. > > Thanks, > > Dave > > > -- Sent from my mobile device ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. From tgraziano at myitdepartment.net Sun Aug 29 05:11:26 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Sun, 29 Aug 2010 08:11:26 -0400 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: References: <18277729.641283065190545.JavaMail.root@zimbra1.spigotsystems.com> <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: Yeah, the sipxroxd is in the installed packages on his build. Remove the intsalled package and make sure the default rule for outgoing traffic is set for manual/static nat, not automatic. http://blog.myitdepartment.net/?p=37 On Sun, Aug 29, 2010 at 7:40 AM, Tony Graziano wrote: > Ipcop has a similar setting to pfsense. You probably missed it. > > MOST FIREWALLS do not use static port NAT. The default rules for > pfsense (and packages) for port 5060 should be removed. > > On your outbound rule for your LAN static port nat needs to be > enabled. Once you do that recreate the nat rules AND remove the > siproxd package by default. > > This is really a pfsense firewall question, it is clear static port > was not enabled so the source port was re-written because that is what > MOST firewalls do by default. > > On 8/29/10, Dave ?Redmore wrote: >> Hello All, >> >> I ran into an issue today that has burned up most of my day troubleshooting. >> I have resolved the problem, but would really like to understand what caused >> it, or some of the internal Freeswitch plumbing that is at play so that I >> can learn something from all of this time I have invested. >> >> I have a Freeswitch server running that acts as a proxy to an account with >> an ITSP for doing T38 faxing. The Freeswitch server has a public IP address >> - there are four "users" who register simple FXS ATAs to my server and it >> then proxies to the ITSP using the "proxy_media" functionality. It has been >> working very well for the last 6 months or so. I have never had to deal with >> any NAT traversal issues - I just point the ATA to the IP to register and >> everything is great. >> >> Here is what the four users "looked" like - >> >> User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> Freeswitch >> Proxy >> User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> Freeswitch >> Proxy >> User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> >> Freeswitch Proxy >> User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> >> Freeswitch Proxy >> >> (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit on >> the same Comcast Gateway) >> >> As I said, this all worked perfectly without any need to "fiddle" with >> anything on any firewalls - worked right out of the box. >> >> So, today I changed out my IPCOP firewall for a pfsense firewall - and my >> HT-287 would no longer register. >> >> After much head-scratching, packet captures, etc. I found that I needed to >> set up a Static Port NAT for the port the HT-287 was using (5062) in order >> to get this to work. >> >> So, I see WHAT is happening, but I really want to know WHY it is happening. >> >> Here are the gory details: >> >> The sofia status of the profile looks like this - when the I have the Static >> Port NAT in place (details changed for security): >> >> _______________________________________________________________ >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> User: 8885554525 at 173.11.22.111 >> Contact: "user" >> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 173.22.22.55 >> Port: 5060 >> Auth-User: 8885554525 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554525 at 173.11.22.111 >> >> Call-ID: 1716488819-5062-1 at 192.168.7.150 >> User: 8885554544 at 173.11.22.111 >> Contact: "user" > fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> >> Agent: Grandstream HT-502 V1.1B 1.0.1.63 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 98.255.0.11 >> Port: 5062 >> Auth-User: 8885554544 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554544 at 173.11.22.111 >> >> Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 >> User: 8885554549 at 173.11.22.111 >> Contact: "user" >> Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 173.11.22.99 >> Port: 5062 >> Auth-User: 8885554549 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554549 at 173.11.22.111 >> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> User: 8885554547 at 173.11.22.111 >> Contact: "user" > _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> >> Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 98.222.55.100 >> Port: 5060 >> Auth-User: 8885554547 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554547 at 173.11.22.111 >> ___________________________________________________________ >> >> The "User4" account is in red. The "Contact" field is substantially >> different and the "Status" indicates "Registered (UDP)", rather than >> "Registered (UDP-NAT)" as the others. >> >> When I do a packet capture on the external NIC interface (eth0) - I see the >> following when the HT-287 tries to register and the Static Port NAT is NOT >> in place: >> >> ___________________________________________________________________ >> Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Method: REGISTER >> Request-URI: sip:173.11.22.111:5090 >> Request-URI Host Part: 173.11.22.111 >> Request-URI Host Port: 5090 >> Message Header >> Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 >> Transport: UDP >> Sent-by Address: 10.8.11.149 >> Sent-by port: 5062 >> Branch: z9hG4bKda48f838c8689e41 >> From: ;tag=c8a0d452edc5ac4b >> SIP from address: sip:8885554549 at 173.11.22.111:5090 >> SIP tag: c8a0d452edc5ac4b >> To: >> Contact: >> Contact Binding: >> Supported: replaces, timer >> Call-ID: aa77d777bae71be6 at 10.8.11.149 >> CSeq: 100 REGISTER >> Sequence Number: 100 >> Method: REGISTER >> Expires: 3600 >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> Max-Forwards: 70 >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> Content-Length: 0 >> _______________________________________________________________ >> >> When Freeswitch replies back with a "401 Unauthorized" - asking for further >> Auth - it replies back to port 5062 - so the packet never comes back >> (pfsense is looking for a packet back on port 11521 in this case). >> >> If I put the Static Port NAT in place - all is well, because the "Source" >> port shows as "5062" - the rest of the packet looks pretty much the same. >> >> Now, here is a packet coming from one of the other Users - this one comes >> through a DD-WRT router - here we see that the Source Port is 5060 : >> >> _________________________________________________________________ >> Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Method: REGISTER >> Request-URI: sip:173.11.22.111:5090 >> [Resent Packet: False] >> Message Header >> Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b >> Transport: UDP >> Sent-by Address: 192.168.1.137 >> Branch: z9hG4bK665bc67a1c64292b >> From: "fax" ;tag=8dc68b35111c4261 >> To: >> Contact: >> Contact Binding: >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> CSeq: 503 REGISTER >> Sequence Number: 503 >> Method: REGISTER >> Expires: 3600 >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> Max-Forwards: 70 >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> Content-Length: 0 >> ______________________________________________________________________ >> >> Here is one more packet coming from a Comcast/SMC Router - again, the source >> port is correct: >> >> ______________________________________________________________________ >> Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Message Header >> Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport >> Transport: UDP >> Sent-by Address: 10.1.10.150 >> Sent-by port: 5060 >> Branch: z9hG4bK58981045 >> RPort: rport >> From: ;tag=138706651 >> To: >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> CSeq: 79875 REGISTER >> Sequence Number: 79875 >> Method: REGISTER >> Contact: >> ;reg-id=1;+sip.instance="" >> Contact Binding: >> ;reg-id=1;+sip.instance="" >> Max-Forwards: 70 >> User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> Supported: path >> Expires: 300 >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, >> UPDATE >> Content-Length: 0 >> ___________________________________________________________ >> >> So, here are my questions: >> >> - Why is the Sofia Status so much different for the registration coming >> through the pfSense firewall. It looks like it doesn't get tagged as being >> NAT'd and the "Contact" info is much less. >> >> - Do most modern routers automatically Static Port NAT any SIP traffic? Both >> DD-WRT and SMC routers appear to be doing this - and not just on a simple >> Port bases (UDP 5060 only), because one of these examples is on 5062. Are >> these "SIP aware" firewalls that are doing this automatically, as the IPCOP >> did before? >> >> - Is the extra "Contact" data in the last packet example different because >> it is a different UA (HT-503 rather than an HT-287) >> >> - Is Freeswitch not flagging the registration from my office (User4) as >> being NAT'd because it is coming in on the same subnet as the interface >> Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and >> pfsense is at 173.11.22.99)? >> >> Sorry for this terribly long posting - I'm just very curious to understand >> what is going on here, now that I have collected all this information. >> >> Thanks, >> >> Dave >> >> >> > > -- > Sent from my mobile device > > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. From ktngl at yahoo.co.uk Sun Aug 29 05:36:36 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Sun, 29 Aug 2010 12:36:36 +0000 (GMT) Subject: [Freeswitch-users] Esl rubymod get variable Message-ID: <782838.95773.qm@web29207.mail.ird.yahoo.com> How can the value of a variable be retrieved in esl ruby -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/1f742331/attachment.html From dujinfang at gmail.com Sun Aug 29 05:57:13 2010 From: dujinfang at gmail.com (Seven Du) Date: Sun, 29 Aug 2010 20:57:13 +0800 Subject: [Freeswitch-users] Esl rubymod get variable In-Reply-To: <782838.95773.qm@web29207.mail.ird.yahoo.com> References: <782838.95773.qm@web29207.mail.ird.yahoo.com> Message-ID: while e = conn.recvEvent name = e.getHeader("Event-Name") var = e.getHeader("variable_blah") puts e.serialize end On Sun, Aug 29, 2010 at 8:36 PM, Nigel Kent wrote: > How can the value of a variable be retrieved in esl ruby > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/92cbf0ed/attachment-0001.html From ktngl at yahoo.co.uk Sun Aug 29 06:40:06 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Sun, 29 Aug 2010 13:40:06 +0000 (GMT) Subject: [Freeswitch-users] Esl rubymod get variable In-Reply-To: Message-ID: <293295.39046.qm@web29219.mail.ird.yahoo.com> That is when an event occurs. I am wanting to get current value of a custom set variable (like application get) example custom variable ivrflag is set to 0 @con.execute("set", "ivrflag=0") Then later on? it may be set to 1. Now I want to check the current state. What would be the syntax to get the current value of ivrflag --- On Sun, 29/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 12:57 while e = conn.recvEvent?? ? ? ? ? ?name = e.getHeader("Event-Name")?? ? ? ?var = e.getHeader("variable_blah")?? ? ? ?puts e.serializeend On Sun, Aug 29, 2010 at 8:36 PM, Nigel Kent wrote: How can the value of a variable be retrieved in esl ruby _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/2796750f/attachment.html From dave.redmore at spigotsystems.com Sun Aug 29 08:26:13 2010 From: dave.redmore at spigotsystems.com (Dave Redmore) Date: Sun, 29 Aug 2010 10:26:13 -0500 (CDT) Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <18775622.01283094676121.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: <16828278.21283095573108.JavaMail.root@zimbra1.spigotsystems.com> Confusion abounds here - sorry if I am being obtuse... A few points on all this - 1 - sipproxd is NOT installed on the firewall 2 - I am confused by the source port randomization issue. I think that what pfsense does by default is randomize the source port translations, rather than using the same source port translations for all connections from an internal host. This is completely different from the issue of telling pfsense to not change the source port at all - i.e. create a Static Port NAT. 3 - One of the things that I find most confusing about what I saw/see in the packet captures is that I EXPECTED to see non-SIP ports as the source port for the registration requests. What we commonly call NAT is more accurately described as PAT (Port Address Translation) - it functions by translating the source port of requests in and out of the firewall. It is one of the FEW things that I like about Cisco is that they more accurately use the terms NAT and PAT. 4 - So, that brings me back to why am I NOT seeing random source ports - why is Freeswitch NOT tagging my connection from pfsense as being NAT'd? Dave To start receiving Spigot Network's once a month newsletter filled with interesting technology news and great offers click SUBSCRIBE ----- Original Message ----- From: "Tony Graziano" To: "FreeSWITCH Users Help" Sent: Sunday, August 29, 2010 7:11:26 AM GMT -06:00 US/Canada Central Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... Yeah, the sipxroxd is in the installed packages on his build. Remove the intsalled package and make sure the default rule for outgoing traffic is set for manual/static nat, not automatic. http://blog.myitdepartment.net/?p=37 On Sun, Aug 29, 2010 at 7:40 AM, Tony Graziano wrote: > Ipcop has a similar setting to pfsense. You probably missed it. > > MOST FIREWALLS do not use static port NAT. The default rules for > pfsense (and packages) for port 5060 should be removed. > > On your outbound rule for your LAN static port nat needs to be > enabled. Once you do that recreate the nat rules AND remove the > siproxd package by default. > > This is really a pfsense firewall question, it is clear static port > was not enabled so the source port was re-written because that is what > MOST firewalls do by default. > > On 8/29/10, Dave Redmore wrote: >> Hello All, >> >> I ran into an issue today that has burned up most of my day troubleshooting. >> I have resolved the problem, but would really like to understand what caused >> it, or some of the internal Freeswitch plumbing that is at play so that I >> can learn something from all of this time I have invested. >> >> I have a Freeswitch server running that acts as a proxy to an account with >> an ITSP for doing T38 faxing. The Freeswitch server has a public IP address >> - there are four "users" who register simple FXS ATAs to my server and it >> then proxies to the ITSP using the "proxy_media" functionality. It has been >> working very well for the last 6 months or so. I have never had to deal with >> any NAT traversal issues - I just point the ATA to the IP to register and >> everything is great. >> >> Here is what the four users "looked" like - >> >> User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> Freeswitch >> Proxy >> User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> Freeswitch >> Proxy >> User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> >> Freeswitch Proxy >> User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> >> Freeswitch Proxy >> >> (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit on >> the same Comcast Gateway) >> >> As I said, this all worked perfectly without any need to "fiddle" with >> anything on any firewalls - worked right out of the box. >> >> So, today I changed out my IPCOP firewall for a pfsense firewall - and my >> HT-287 would no longer register. >> >> After much head-scratching, packet captures, etc. I found that I needed to >> set up a Static Port NAT for the port the HT-287 was using (5062) in order >> to get this to work. >> >> So, I see WHAT is happening, but I really want to know WHY it is happening. >> >> Here are the gory details: >> >> The sofia status of the profile looks like this - when the I have the Static >> Port NAT in place (details changed for security): >> >> _______________________________________________________________ >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> User: 8885554525 at 173.11.22.111 >> Contact: "user" >> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 173.22.22.55 >> Port: 5060 >> Auth-User: 8885554525 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554525 at 173.11.22.111 >> >> Call-ID: 1716488819-5062-1 at 192.168.7.150 >> User: 8885554544 at 173.11.22.111 >> Contact: "user" > fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> >> Agent: Grandstream HT-502 V1.1B 1.0.1.63 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 98.255.0.11 >> Port: 5062 >> Auth-User: 8885554544 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554544 at 173.11.22.111 >> >> Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 >> User: 8885554549 at 173.11.22.111 >> Contact: "user" >> Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 173.11.22.99 >> Port: 5062 >> Auth-User: 8885554549 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554549 at 173.11.22.111 >> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> User: 8885554547 at 173.11.22.111 >> Contact: "user" > _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> >> Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 98.222.55.100 >> Port: 5060 >> Auth-User: 8885554547 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554547 at 173.11.22.111 >> ___________________________________________________________ >> >> The "User4" account is in red. The "Contact" field is substantially >> different and the "Status" indicates "Registered (UDP)", rather than >> "Registered (UDP-NAT)" as the others. >> >> When I do a packet capture on the external NIC interface (eth0) - I see the >> following when the HT-287 tries to register and the Static Port NAT is NOT >> in place: >> >> ___________________________________________________________________ >> Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Method: REGISTER >> Request-URI: sip:173.11.22.111:5090 >> Request-URI Host Part: 173.11.22.111 >> Request-URI Host Port: 5090 >> Message Header >> Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 >> Transport: UDP >> Sent-by Address: 10.8.11.149 >> Sent-by port: 5062 >> Branch: z9hG4bKda48f838c8689e41 >> From: ;tag=c8a0d452edc5ac4b >> SIP from address: sip:8885554549 at 173.11.22.111:5090 >> SIP tag: c8a0d452edc5ac4b >> To: >> Contact: >> Contact Binding: >> Supported: replaces, timer >> Call-ID: aa77d777bae71be6 at 10.8.11.149 >> CSeq: 100 REGISTER >> Sequence Number: 100 >> Method: REGISTER >> Expires: 3600 >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> Max-Forwards: 70 >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> Content-Length: 0 >> _______________________________________________________________ >> >> When Freeswitch replies back with a "401 Unauthorized" - asking for further >> Auth - it replies back to port 5062 - so the packet never comes back >> (pfsense is looking for a packet back on port 11521 in this case). >> >> If I put the Static Port NAT in place - all is well, because the "Source" >> port shows as "5062" - the rest of the packet looks pretty much the same. >> >> Now, here is a packet coming from one of the other Users - this one comes >> through a DD-WRT router - here we see that the Source Port is 5060 : >> >> _________________________________________________________________ >> Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Method: REGISTER >> Request-URI: sip:173.11.22.111:5090 >> [Resent Packet: False] >> Message Header >> Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b >> Transport: UDP >> Sent-by Address: 192.168.1.137 >> Branch: z9hG4bK665bc67a1c64292b >> From: "fax" ;tag=8dc68b35111c4261 >> To: >> Contact: >> Contact Binding: >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> CSeq: 503 REGISTER >> Sequence Number: 503 >> Method: REGISTER >> Expires: 3600 >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> Max-Forwards: 70 >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> Content-Length: 0 >> ______________________________________________________________________ >> >> Here is one more packet coming from a Comcast/SMC Router - again, the source >> port is correct: >> >> ______________________________________________________________________ >> Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> Message Header >> Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport >> Transport: UDP >> Sent-by Address: 10.1.10.150 >> Sent-by port: 5060 >> Branch: z9hG4bK58981045 >> RPort: rport >> From: ;tag=138706651 >> To: >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> CSeq: 79875 REGISTER >> Sequence Number: 79875 >> Method: REGISTER >> Contact: >> ;reg-id=1;+sip.instance="" >> Contact Binding: >> ;reg-id=1;+sip.instance="" >> Max-Forwards: 70 >> User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> Supported: path >> Expires: 300 >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, >> UPDATE >> Content-Length: 0 >> ___________________________________________________________ >> >> So, here are my questions: >> >> - Why is the Sofia Status so much different for the registration coming >> through the pfSense firewall. It looks like it doesn't get tagged as being >> NAT'd and the "Contact" info is much less. >> >> - Do most modern routers automatically Static Port NAT any SIP traffic? Both >> DD-WRT and SMC routers appear to be doing this - and not just on a simple >> Port bases (UDP 5060 only), because one of these examples is on 5062. Are >> these "SIP aware" firewalls that are doing this automatically, as the IPCOP >> did before? >> >> - Is the extra "Contact" data in the last packet example different because >> it is a different UA (HT-503 rather than an HT-287) >> >> - Is Freeswitch not flagging the registration from my office (User4) as >> being NAT'd because it is coming in on the same subnet as the interface >> Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and >> pfsense is at 173.11.22.99)? >> >> Sorry for this terribly long posting - I'm just very curious to understand >> what is going on here, now that I have collected all this information. >> >> Thanks, >> >> Dave >> >> >> > > -- > Sent from my mobile device > > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/347314cb/attachment-0001.html From tgraziano at myitdepartment.net Sun Aug 29 08:53:05 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Sun, 29 Aug 2010 11:53:05 -0400 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <16828278.21283095573108.JavaMail.root@zimbra1.spigotsystems.com> References: <18775622.01283094676121.JavaMail.root@zimbra1.spigotsystems.com> <16828278.21283095573108.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: If pfsense and your FS install are on the same subnet then "something" must be sitting in between to randomize the ports. In the 1.2.x install of pfsense, siproxd is installed by default. There is also a default rule for port 5060 that is in pfsense. I suggest removing the filters AND rules and starting from scratch. I believe this is also the default, and undesired installation, when installing a sip system behind it, even for the beta snapshots of pfsense 2.0. Please check your installed packages. Please remove any rules you did not create for the network and start over. What you are describing sounds to me the siproxd IS/WAS installed. It picks up on anything on the LAN and randomizes the port. It's a very common thing with pfsense. On Sun, Aug 29, 2010 at 11:26 AM, Dave Redmore < dave.redmore at spigotsystems.com> wrote: > Confusion abounds here - sorry if I am being obtuse... > > A few points on all this - > > 1 - sipproxd is NOT installed on the firewall > > 2 - I am confused by the source port randomization issue. I think that > what pfsense does by default is randomize the source port translations, > rather than using the same source port translations for all connections from > an internal host. This is completely different from the issue of telling > pfsense to not change the source port at all - i.e. create a Static Port > NAT. > > 3 - One of the things that I find most confusing about what I saw/see in > the packet captures is that I EXPECTED to see non-SIP ports as the source > port for the registration requests. What we commonly call NAT is more > accurately described as PAT (Port Address Translation) - it functions by > translating the source port of requests in and out of the firewall. It is > one of the FEW things that I like about Cisco is that they more accurately > use the terms NAT and PAT. > > 4 - So, that brings me back to why am I NOT seeing random source ports - > why is Freeswitch NOT tagging my connection from pfsense as being NAT'd? > > Dave > > > > To start receiving Spigot Network's once a month newsletter filled with > interesting technology news and great offers click SUBSCRIBE > > > ----- Original Message ----- > From: "Tony Graziano" > To: "FreeSWITCH Users Help" > Sent: Sunday, August 29, 2010 7:11:26 AM GMT -06:00 US/Canada Central > Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... > > Yeah, the sipxroxd is in the installed packages on his build. Remove > the intsalled package and make sure the default rule for outgoing > traffic is set for manual/static nat, not automatic. > > http://blog.myitdepartment.net/?p=37 > > On Sun, Aug 29, 2010 at 7:40 AM, Tony Graziano > wrote: > > Ipcop has a similar setting to pfsense. You probably missed it. > > > > MOST FIREWALLS do not use static port NAT. The default rules for > > pfsense (and packages) for port 5060 should be removed. > > > > On your outbound rule for your LAN static port nat needs to be > > enabled. Once you do that recreate the nat rules AND remove the > > siproxd package by default. > > > > This is really a pfsense firewall question, it is clear static port > > was not enabled so the source port was re-written because that is what > > MOST firewalls do by default. > > > > On 8/29/10, Dave Redmore wrote: > >> Hello All, > >> > >> I ran into an issue today that has burned up most of my day > troubleshooting. > >> I have resolved the problem, but would really like to understand what > caused > >> it, or some of the internal Freeswitch plumbing that is at play so that > I > >> can learn something from all of this time I have invested. > >> > >> I have a Freeswitch server running that acts as a proxy to an account > with > >> an ITSP for doing T38 faxing. The Freeswitch server has a public IP > address > >> - there are four "users" who register simple FXS ATAs to my server and > it > >> then proxies to the ITSP using the "proxy_media" functionality. It has > been > >> working very well for the last 6 months or so. I have never had to deal > with > >> any NAT traversal issues - I just point the ATA to the IP to register > and > >> everything is great. > >> > >> Here is what the four users "looked" like - > >> > >> User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> > Freeswitch > >> Proxy > >> User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> > Freeswitch > >> Proxy > >> User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> > >> Freeswitch Proxy > >> User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> > >> Freeswitch Proxy > >> > >> (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit > on > >> the same Comcast Gateway) > >> > >> As I said, this all worked perfectly without any need to "fiddle" with > >> anything on any firewalls - worked right out of the box. > >> > >> So, today I changed out my IPCOP firewall for a pfsense firewall - and > my > >> HT-287 would no longer register. > >> > >> After much head-scratching, packet captures, etc. I found that I needed > to > >> set up a Static Port NAT for the port the HT-287 was using (5062) in > order > >> to get this to work. > >> > >> So, I see WHAT is happening, but I really want to know WHY it is > happening. > >> > >> Here are the gory details: > >> > >> The sofia status of the profile looks like this - when the I have the > Static > >> Port NAT in place (details changed for security): > >> > >> _______________________________________________________________ > >> Call-ID: 0e551b3c694a793c at 192.168.1.137 > >> User: 8885554525 at 173.11.22.111 > >> Contact: "user" > >> > ;fs_nat=yes;fs_path=sip%3A8885554525%40173.22.22.55%3A5060> > >> Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) > >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > >> IP: 173.22.22.55 > >> Port: 5060 > >> Auth-User: 8885554525 > >> Auth-Realm: 173.11.22.111 > >> MWI-Account: 8885554525 at 173.11.22.111 > >> > >> Call-ID: 1716488819-5062-1 at 192.168.7.150 > >> User: 8885554544 at 173.11.22.111 > >> Contact: "user" ;user=phone;fs_nat=yes; > >> fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> > >> Agent: Grandstream HT-502 V1.1B 1.0.1.63 > >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) > >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > >> IP: 98.255.0.11 > >> Port: 5062 > >> Auth-User: 8885554544 > >> Auth-Realm: 173.11.22.111 > >> MWI-Account: 8885554544 at 173.11.22.111 > >> > >> Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 > >> User: 8885554549 at 173.11.22.111 > >> Contact: "user" > >> Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > >> Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) > >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > >> IP: 173.11.22.99 > >> Port: 5062 > >> Auth-User: 8885554549 > >> Auth-Realm: 173.11.22.111 > >> MWI-Account: 8885554549 at 173.11.22.111 > >> > >> Call-ID: 1035241259-5060-1 at 10.1.10.150 > >> User: 8885554547 at 173.11.22.111 > >> Contact: "user" ;user=phone;fs_nat=yes;fs > >> _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> > >> Agent: Grandstream HT-503 V1.1B 1.0.1.63 > >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) > >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > >> IP: 98.222.55.100 > >> Port: 5060 > >> Auth-User: 8885554547 > >> Auth-Realm: 173.11.22.111 > >> MWI-Account: 8885554547 at 173.11.22.111 > >> ___________________________________________________________ > >> > >> The "User4" account is in red. The "Contact" field is substantially > >> different and the "Status" indicates "Registered (UDP)", rather than > >> "Registered (UDP-NAT)" as the others. > >> > >> When I do a packet capture on the external NIC interface (eth0) - I see > the > >> following when the HT-287 tries to register and the Static Port NAT is > NOT > >> in place: > >> > >> ___________________________________________________________________ > >> Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 > >> (173.11.22.111) > >> User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) > >> Session Initiation Protocol > >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > >> Method: REGISTER > >> Request-URI: sip:173.11.22.111:5090 > >> Request-URI Host Part: 173.11.22.111 > >> Request-URI Host Port: 5090 > >> Message Header > >> Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 > >> Transport: UDP > >> Sent-by Address: 10.8.11.149 > >> Sent-by port: 5062 > >> Branch: z9hG4bKda48f838c8689e41 > >> From: ;tag=c8a0d452edc5ac4b > >> SIP from address: sip:8885554549 at 173.11.22.111:5090 > >> SIP tag: c8a0d452edc5ac4b > >> To: > >> Contact: > >> Contact Binding: > >> Supported: replaces, timer > >> Call-ID: aa77d777bae71be6 at 10.8.11.149 > >> CSeq: 100 REGISTER > >> Sequence Number: 100 > >> Method: REGISTER > >> Expires: 3600 > >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > >> Max-Forwards: 70 > >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > >> Content-Length: 0 > >> _______________________________________________________________ > >> > >> When Freeswitch replies back with a "401 Unauthorized" - asking for > further > >> Auth - it replies back to port 5062 - so the packet never comes back > >> (pfsense is looking for a packet back on port 11521 in this case). > >> > >> If I put the Static Port NAT in place - all is well, because the > "Source" > >> port shows as "5062" - the rest of the packet looks pretty much the > same. > >> > >> Now, here is a packet coming from one of the other Users - this one > comes > >> through a DD-WRT router - here we see that the Source Port is 5060 : > >> > >> _________________________________________________________________ > >> Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 > >> (173.11.22.111) > >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > >> Session Initiation Protocol > >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > >> Method: REGISTER > >> Request-URI: sip:173.11.22.111:5090 > >> [Resent Packet: False] > >> Message Header > >> Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b > >> Transport: UDP > >> Sent-by Address: 192.168.1.137 > >> Branch: z9hG4bK665bc67a1c64292b > >> From: "fax" ;tag=8dc68b35111c4261 > >> To: > >> Contact: > > > >> Contact Binding: > > > >> Call-ID: 0e551b3c694a793c at 192.168.1.137 > >> CSeq: 503 REGISTER > >> Sequence Number: 503 > >> Method: REGISTER > >> Expires: 3600 > >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > >> Max-Forwards: 70 > >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > >> Content-Length: 0 > >> ______________________________________________________________________ > >> > >> Here is one more packet coming from a Comcast/SMC Router - again, the > source > >> port is correct: > >> > >> ______________________________________________________________________ > >> Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: > 173.11.22.111 > >> (173.11.22.111) > >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > >> Session Initiation Protocol > >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > >> Message Header > >> Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport > >> Transport: UDP > >> Sent-by Address: 10.1.10.150 > >> Sent-by port: 5060 > >> Branch: z9hG4bK58981045 > >> RPort: rport > >> From: ;tag=138706651 > >> To: > >> Call-ID: 1035241259-5060-1 at 10.1.10.150 > >> CSeq: 79875 REGISTER > >> Sequence Number: 79875 > >> Method: REGISTER > >> Contact: > >> ;user=phone>;reg-id=1;+sip.instance="" > >> Contact Binding: > >> ;user=phone>;reg-id=1;+sip.instance="" > >> Max-Forwards: 70 > >> User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 > >> Supported: path > >> Expires: 300 > >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, > >> UPDATE > >> Content-Length: 0 > >> ___________________________________________________________ > >> > >> So, here are my questions: > >> > >> - Why is the Sofia Status so much different for the registration coming > >> through the pfSense firewall. It looks like it doesn't get tagged as > being > >> NAT'd and the "Contact" info is much less. > >> > >> - Do most modern routers automatically Static Port NAT any SIP traffic? > Both > >> DD-WRT and SMC routers appear to be doing this - and not just on a > simple > >> Port bases (UDP 5060 only), because one of these examples is on 5062. > Are > >> these "SIP aware" firewalls that are doing this automatically, as the > IPCOP > >> did before? > >> > >> - Is the extra "Contact" data in the last packet example different > because > >> it is a different UA (HT-503 rather than an HT-287) > >> > >> - Is Freeswitch not flagging the registration from my office (User4) as > >> being NAT'd because it is coming in on the same subnet as the interface > >> Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and > >> pfsense is at 173.11.22.99)? > >> > >> Sorry for this terribly long posting - I'm just very curious to > understand > >> what is going on here, now that I have collected all this information. > >> > >> Thanks, > >> > >> Dave > >> > >> > >> > > > > -- > > Sent from my mobile device > > > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > sip: tgraziano at voice.myitdepartment.net > > Fax: 434.984.8431 > > > > Email: tgraziano at myitdepartment.net > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > sip: helpdesk at voice.myitdepartment.net > > Fax: 434.984.8427 > > > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > > > Why do mathematicians always confuse Halloween and Christmas? > > Because 31 Oct = 25 Dec. > > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/f21a1aeb/attachment-0001.html From dujinfang at gmail.com Sun Aug 29 09:18:56 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 30 Aug 2010 00:18:56 +0800 Subject: [Freeswitch-users] Esl rubymod get variable In-Reply-To: <293295.39046.qm@web29219.mail.ird.yahoo.com> References: <293295.39046.qm@web29219.mail.ird.yahoo.com> Message-ID: you either can wait for a new event coming, or use the uuid_dump API to get everything. @con.execute("uuid_dump", uuid).getBody On Sun, Aug 29, 2010 at 9:40 PM, Nigel Kent wrote: > That is when an event occurs. I am wanting to get current value of a custom > set variable (like application get) > > example custom variable ivrflag is set to 0 > > @con.execute("set", "ivrflag=0") > Then later on it may be set to 1. > > Now I want to check the current state. What would be the syntax to get the > current value of ivrflag > > > > --- On *Sun, 29/8/10, Seven Du * wrote: > > > From: Seven Du > Subject: Re: [Freeswitch-users] Esl rubymod get variable > To: "FreeSWITCH Users Help" > Date: Sunday, 29 August, 2010, 12:57 > > > while e = conn.recvEvent > name = e.getHeader("Event-Name") > var = e.getHeader("variable_blah") > puts e.serialize > end > > > On Sun, Aug 29, 2010 at 8:36 PM, Nigel Kent > > wrote: > > How can the value of a variable be retrieved in esl ruby > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > -----Inline Attachment Follows----- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/48fb33c1/attachment.html From ktngl at yahoo.co.uk Sun Aug 29 09:38:35 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Sun, 29 Aug 2010 16:38:35 +0000 (GMT) Subject: [Freeswitch-users] Esl rubymod get variable In-Reply-To: Message-ID: <750018.49968.qm@web29213.mail.ird.yahoo.com> Thanks for that tip. I want to find out as well how to retrieve the? value with the opposite of the 'set' command which is how the value was added in the first place. Is a get command not available ?. --- On Sun, 29/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 16:18 you either can wait for a new event coming, or use the uuid_dump API to get everything. @con.execute("uuid_dump", uuid).getBody On Sun, Aug 29, 2010 at 9:40 PM, Nigel Kent wrote: That is when an event occurs. I am wanting to get current value of a custom set variable (like application get) example custom variable ivrflag is set to 0 @con.execute("set", "ivrflag=0") Then later on? it may be set to 1. Now I want to check the current state. What would be the syntax to get the current value of ivrflag --- On Sun, 29/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 12:57 while e = conn.recvEvent?? ? ? ? ? ?name = e.getHeader("Event-Name")?? ? ? ?var = e.getHeader("variable_blah")?? ? ? ?puts e.serializeend On Sun, Aug 29, 2010 at 8:36 PM, Nigel Kent wrote: How can the value of a variable be retrieved in esl ruby _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/591f33c3/attachment.html From jonas.gauffin at gmail.com Sun Aug 29 09:55:49 2010 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Sun, 29 Aug 2010 18:55:49 +0200 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION In-Reply-To: References: Message-ID: I've done some debugging and found the problem (but not the cause). switch_core_session_perform_receive_message fails to switch_core_session_read_lock the session (error code: 720258). Should I post it as a jira issue? What does 720258 mean? On Fri, Aug 27, 2010 at 1:36 PM, Jonas Gauffin wrote: > Excellent information thanks. The server is a dedicated voip server and > nothing else is running on it (and therefore the port should be free). > > I've talked to my voip provider and they were kind enough to give me their > traces. > > Freeswitch get's their SDP on the bleg (their software have modified the > trace a bit): > > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP 212.247.E.E;rport=5060;branch=z9hG4bK52XcXtpQBp3BH > From: "blablabla" ;tag=rccD75cXD61Kr > To: > >;tag=686577264 > Call-ID: c3e8e0e8-2c3c-122e-479c-1fc6e9408ca4 > CSeq: 1106961 INVITE > Contact: > Record-Route: > Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE > Content-Type: application/sdp > Content-Length: 169 > > v=0 > o=- 46541649 0 IN IP4 130.244.x.x > s=Cisco SDP 0 > c=IN IP4 130.244.x.x > t=0 0 > m=audio 18928 RTP/AVP 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > which makes FS cancel the call leg: > > CANCEL sip:0771221122 at voipprovider.se SIP/2.0 > Via: SIP/2.0/UDP 212.247.E.E;rport;branch=z9hG4bK52XcXtpQBp3BH > Max-Forwards: 68 > From: "blablabla" ;tag=rccD75cXD61Kr > To: > > Call-ID: c3e8e0e8-2c3c-122e-479c-1fc6e9408ca4 > CSeq: 1106961 CANCEL > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > > > On Fri, Aug 27, 2010 at 9:57 AM, Steven Ayre wrote: > >> Yes, G.729a and G.729b are incorrect and your device is at fault... only >> other phones of the same type would probably recognise it without issue. >> >> PCMA should be used in this case though. >> >> RTP payload numbers are spread through multiple RFCs. Anything in the >> 96-127 range is dynamic and the codec is determined from the matching rtpmap >> line, any of the static numbers don't need a rtpmap line to work. IANA >> oversees assignment of the static numbers and they have the full list: >> http://www.iana.org/assignments/rtp-parameters >> >> As you can see 0=PCMU, 8=PCMA, and G.729 should use 18. Support of annex B >> is specified in the fmtp parameter, not the codec name - e.g. "a=fmtp:18 >> annexb=no". Annex A never needs to be specified as it can be read normally >> by plain G.729, so it's just up to the implementation on whether it wants to >> save quality or cpu when encoding. >> >> Do you have any other applications running which would also be using the >> RTP port range? A call will fail if it tries to use a port that's already in >> use, perhaps with that message. FS should avoid using ports it's already >> using, but can't know about any other programs on the system. >> >> -Steve >> >> >> >> On 27 August 2010 08:03, Jonas Gauffin wrote: >> >>> It doesn't happen every time and it's on a production system with a bit >>> of volume. therefore a bit hard to get SIP traces. I'll try if Anthony >>> really needs them. >>> >>> FS do say this: >>> 2010-08-27 07:14:10.758750 [DEBUG] switch_ivr_originate.c:3111sofia/external/ >>> 0700123456 at 212.151.Y.Y:5060 Media Establishment Failed >>> >>> In which RFC are codec names defined? rfc3551 defines "G729" but no >>> "G.729A" or "G.729B". But as you say, shouldn't FS use PCMA in any case? >>> >>> >>> >>> On Fri, Aug 27, 2010 at 8:43 AM, Steven Ayre wrote: >>> >>>> Ordinarily it'd mean there was a problem with the codecs (e.g. needing >>>> to use an unsupported codec or transcode a codec that only works as a >>>> passthrough one). >>>> >>>> Looks like it should have gone through with PCMA (8) though. Can you >>>> repeat the call with sip trace on? Perhaps the incompatible destination >>>> comes from an endpoint. >>>> >>>> 'sofia profile siptrace on' from the CLI, replace on with >>>> off to turn it off again. >>>> >>>> -Steve >>>> >>>> >>>> >>>> On 27 August 2010 07:26, Jonas Gauffin wrote: >>>> >>>>> Hello, >>>>> >>>>> Why do this call result in INCOMPATIBLE_DESTINATION? >>>>> >>>>> http://pastebin.freeswitch.org/13736 >>>>> >>>>> Regards, >>>>> Jonas >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/08d2a314/attachment-0001.html From mrene_lists at avgs.ca Sun Aug 29 10:31:58 2010 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 29 Aug 2010 13:31:58 -0400 Subject: [Freeswitch-users] Esl rubymod get variable In-Reply-To: <750018.49968.qm@web29213.mail.ird.yahoo.com> References: <750018.49968.qm@web29213.mail.ird.yahoo.com> Message-ID: <0B59088C-7817-4E1F-9BC7-A87799975829@avgs.ca> uuid_getvar http://wiki.freeswitch.org/wiki/Mod_commands#uuid_getvar Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2010-08-29, at 12:38 PM, Nigel Kent wrote: > Thanks for that tip. > > I want to find out as well how to retrieve the value with the opposite of the 'set' command which is how the value was added in the first place. > > Is a get command not available ?. > > > > --- On Sun, 29/8/10, Seven Du wrote: > > From: Seven Du > Subject: Re: [Freeswitch-users] Esl rubymod get variable > To: "FreeSWITCH Users Help" > Date: Sunday, 29 August, 2010, 16:18 > > you either can wait for a new event coming, or use the uuid_dump API to get everything. > > @con.execute("uuid_dump", uuid).getBody > > On Sun, Aug 29, 2010 at 9:40 PM, Nigel Kent wrote: > That is when an event occurs. I am wanting to get current value of a custom set variable (like application get) > > example custom variable ivrflag is set to 0 > > @con.execute("set", "ivrflag=0") > Then later on it may be set to 1. > > Now I want to check the current state. What would be the syntax to get the current value of ivrflag > > > > --- On Sun, 29/8/10, Seven Du wrote: > > From: Seven Du > Subject: Re: [Freeswitch-users] Esl rubymod get variable > To: "FreeSWITCH Users Help" > Date: Sunday, 29 August, 2010, 12:57 > > > while e = conn.recvEvent > name = e.getHeader("Event-Name") > var = e.getHeader("variable_blah") > puts e.serialize > end > > > On Sun, Aug 29, 2010 at 8:36 PM, Nigel Kent wrote: > How can the value of a variable be retrieved in esl ruby > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > -----Inline Attachment Follows----- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/ec1273a7/attachment.html From dave.redmore at spigotsystems.com Sun Aug 29 11:00:31 2010 From: dave.redmore at spigotsystems.com (Dave Redmore) Date: Sun, 29 Aug 2010 13:00:31 -0500 (CDT) Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <25121409.71283104421357.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: <13431520.91283104831264.JavaMail.root@zimbra1.spigotsystems.com> Thanks for your help/thoughts Tony. I just confirmed that siproxd is not installed - SSH'd in and checked running processes to be sure. Like I said earlier - I can make it work - I have disabled the automatic outbound NAT and set up AON (Advanced Outbound Nat). It is just that I want to understand what I am seeing so that I can learn from all this. Here is one thing - If Freeswitch flagged my pfsense connection as being behind NAT - would it then compensate for the Source port being 11521 (per the packet capture in the original email)? Am I totally wrong in thinking that it would be "normal" to see packets with the source port changed for users behind generic NAT firewalls? I might try hooking the IPCOP box back up and doing a capture of that, so I can see what is different between the IPCOP (worked "out of the box") vs. the pfsense. Dave ----- Original Message ----- From: "Tony Graziano" To: "FreeSWITCH Users Help" Sent: Sunday, August 29, 2010 10:53:05 AM GMT -06:00 US/Canada Central Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... If pfsense and your FS install are on the same subnet then "something" must be sitting in between to randomize the ports. In the 1.2.x install of pfsense, siproxd is installed by default. There is also a default rule for port 5060 that is in pfsense. I suggest removing the filters AND rules and starting from scratch. I believe this is also the default, and undesired installation, when installing a sip system behind it, even for the beta snapshots of pfsense 2.0. Please check your installed packages. Please remove any rules you did not create for the network and start over. What you are describing sounds to me the siproxd IS/WAS installed. It picks up on anything on the LAN and randomizes the port. It's a very common thing with pfsense. On Sun, Aug 29, 2010 at 11:26 AM, Dave Redmore < dave.redmore at spigotsystems.com > wrote: Confusion abounds here - sorry if I am being obtuse... A few points on all this - 1 - sipproxd is NOT installed on the firewall 2 - I am confused by the source port randomization issue. I think that what pfsense does by default is randomize the source port translations, rather than using the same source port translations for all connections from an internal host. This is completely different from the issue of telling pfsense to not change the source port at all - i.e. create a Static Port NAT. 3 - One of the things that I find most confusing about what I saw/see in the packet captures is that I EXPECTED to see non-SIP ports as the source port for the registration requests. What we commonly call NAT is more accurately described as PAT (Port Address Translation) - it functions by translating the source port of requests in and out of the firewall. It is one of the FEW things that I like about Cisco is that they more accurately use the terms NAT and PAT. 4 - So, that brings me back to why am I NOT seeing random source ports - why is Freeswitch NOT tagging my connection from pfsense as being NAT'd? Dave To start receiving Spigot Network's once a month newsletter filled with interesting technology news and great offers click SUBSCRIBE ----- Original Message ----- From: "Tony Graziano" < tgraziano at myitdepartment.net > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Sent: Sunday, August 29, 2010 7:11:26 AM GMT -06:00 US/Canada Central Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... Yeah, the sipxroxd is in the installed packages on his build. Remove the intsalled package and make sure the default rule for outgoing traffic is set for manual/static nat, not automatic. http://blog.myitdepartment.net/?p=37 On Sun, Aug 29, 2010 at 7:40 AM, Tony Graziano < tgraziano at myitdepartment.net > wrote: > Ipcop has a similar setting to pfsense. You probably missed it. > > MOST FIREWALLS do not use static port NAT. The default rules for > pfsense (and packages) for port 5060 should be removed. > > On your outbound rule for your LAN static port nat needs to be > enabled. Once you do that recreate the nat rules AND remove the > siproxd package by default. > > This is really a pfsense firewall question, it is clear static port > was not enabled so the source port was re-written because that is what > MOST firewalls do by default. > > On 8/29/10, Dave Redmore < dave.redmore at spigotsystems.com > wrote: >> Hello All, >> >> I ran into an issue today that has burned up most of my day troubleshooting. >> I have resolved the problem, but would really like to understand what caused >> it, or some of the internal Freeswitch plumbing that is at play so that I >> can learn something from all of this time I have invested. >> >> I have a Freeswitch server running that acts as a proxy to an account with >> an ITSP for doing T38 faxing. The Freeswitch server has a public IP address >> - there are four "users" who register simple FXS ATAs to my server and it >> then proxies to the ITSP using the "proxy_media" functionality. It has been >> working very well for the last 6 months or so. I have never had to deal with >> any NAT traversal issues - I just point the ATA to the IP to register and >> everything is great. >> >> Here is what the four users "looked" like - >> >> User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> Freeswitch >> Proxy >> User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> Freeswitch >> Proxy >> User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> >> Freeswitch Proxy >> User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> >> Freeswitch Proxy >> >> (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit on >> the same Comcast Gateway) >> >> As I said, this all worked perfectly without any need to "fiddle" with >> anything on any firewalls - worked right out of the box. >> >> So, today I changed out my IPCOP firewall for a pfsense firewall - and my >> HT-287 would no longer register. >> >> After much head-scratching, packet captures, etc. I found that I needed to >> set up a Static Port NAT for the port the HT-287 was using (5062) in order >> to get this to work. >> >> So, I see WHAT is happening, but I really want to know WHY it is happening. >> >> Here are the gory details: >> >> The sofia status of the profile looks like this - when the I have the Static >> Port NAT in place (details changed for security): >> >> _______________________________________________________________ >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> User: 8885554525 at 173.11.22.111 >> Contact: "user" >> < sip:8885554525 at 192.168.1.137 ;fs_nat=yes;fs_path=sip%3A8885554525%40173.22.22.55%3A5060> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 173.22.22.55 >> Port: 5060 >> Auth-User: 8885554525 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554525 at 173.11.22.111 >> >> Call-ID: 1716488819-5062-1 at 192.168.7.150 >> User: 8885554544 at 173.11.22.111 >> Contact: "user" > fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> >> Agent: Grandstream HT-502 V1.1B 1.0.1.63 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 98.255.0.11 >> Port: 5062 >> Auth-User: 8885554544 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554544 at 173.11.22.111 >> >> Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 >> User: 8885554549 at 173.11.22.111 >> Contact: "user" < sip:8885554549 at 10.8.11.149:5062 > >> Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 173.11.22.99 >> Port: 5062 >> Auth-User: 8885554549 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554549 at 173.11.22.111 >> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> User: 8885554547 at 173.11.22.111 >> Contact: "user" > _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> >> Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> IP: 98.222.55.100 >> Port: 5060 >> Auth-User: 8885554547 >> Auth-Realm: 173.11.22.111 >> MWI-Account: 8885554547 at 173.11.22.111 >> ___________________________________________________________ >> >> The "User4" account is in red. The "Contact" field is substantially >> different and the "Status" indicates "Registered (UDP)", rather than >> "Registered (UDP-NAT)" as the others. >> >> When I do a packet capture on the external NIC interface (eth0) - I see the >> following when the HT-287 tries to register and the Static Port NAT is NOT >> in place: >> >> ___________________________________________________________________ >> Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip: 173.11.22.111:5090 SIP/2.0 >> Method: REGISTER >> Request-URI: sip: 173.11.22.111:5090 >> Request-URI Host Part: 173.11.22.111 >> Request-URI Host Port: 5090 >> Message Header >> Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 >> Transport: UDP >> Sent-by Address: 10.8.11.149 >> Sent-by port: 5062 >> Branch: z9hG4bKda48f838c8689e41 >> From: < sip:8885554549 at 173.11.22.111:5090 >;tag=c8a0d452edc5ac4b >> SIP from address: sip:8885554549 at 173.11.22.111:5090 >> SIP tag: c8a0d452edc5ac4b >> To: < sip:8885554549 at 173.11.22.111:5090 > >> Contact: < sip:88855564549 at 10.8.11.149:5062 > >> Contact Binding: < sip:8885554549 at 10.8.11.149:5062 > >> Supported: replaces, timer >> Call-ID: aa77d777bae71be6 at 10.8.11.149 >> CSeq: 100 REGISTER >> Sequence Number: 100 >> Method: REGISTER >> Expires: 3600 >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> Max-Forwards: 70 >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> Content-Length: 0 >> _______________________________________________________________ >> >> When Freeswitch replies back with a "401 Unauthorized" - asking for further >> Auth - it replies back to port 5062 - so the packet never comes back >> (pfsense is looking for a packet back on port 11521 in this case). >> >> If I put the Static Port NAT in place - all is well, because the "Source" >> port shows as "5062" - the rest of the packet looks pretty much the same. >> >> Now, here is a packet coming from one of the other Users - this one comes >> through a DD-WRT router - here we see that the Source Port is 5060 : >> >> _________________________________________________________________ >> Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip: 173.11.22.111:5090 SIP/2.0 >> Method: REGISTER >> Request-URI: sip: 173.11.22.111:5090 >> [Resent Packet: False] >> Message Header >> Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b >> Transport: UDP >> Sent-by Address: 192.168.1.137 >> Branch: z9hG4bK665bc67a1c64292b >> From: "fax" < sip:8885554525 at 173.11.22.111:5090 >;tag=8dc68b35111c4261 >> To: < sip:8156564525 at 173.15.28.101:5090 > >> Contact: < sip:8885554525 at 192.168.1.137 > >> Contact Binding: < sip:8885554525 at 192.168.1.137 > >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> CSeq: 503 REGISTER >> Sequence Number: 503 >> Method: REGISTER >> Expires: 3600 >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> Max-Forwards: 70 >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> Content-Length: 0 >> ______________________________________________________________________ >> >> Here is one more packet coming from a Comcast/SMC Router - again, the source >> port is correct: >> >> ______________________________________________________________________ >> Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: 173.11.22.111 >> (173.11.22.111) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> Session Initiation Protocol >> Request-Line: REGISTER sip: 173.11.22.111:5090 SIP/2.0 >> Message Header >> Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport >> Transport: UDP >> Sent-by Address: 10.1.10.150 >> Sent-by port: 5060 >> Branch: z9hG4bK58981045 >> RPort: rport >> From: ;tag=138706651 >> To: >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> CSeq: 79875 REGISTER >> Sequence Number: 79875 >> Method: REGISTER >> Contact: >> ;reg-id=1;+sip.instance="" >> Contact Binding: >> ;reg-id=1;+sip.instance="" >> Max-Forwards: 70 >> User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> Supported: path >> Expires: 300 >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, >> UPDATE >> Content-Length: 0 >> ___________________________________________________________ >> >> So, here are my questions: >> >> - Why is the Sofia Status so much different for the registration coming >> through the pfSense firewall. It looks like it doesn't get tagged as being >> NAT'd and the "Contact" info is much less. >> >> - Do most modern routers automatically Static Port NAT any SIP traffic? Both >> DD-WRT and SMC routers appear to be doing this - and not just on a simple >> Port bases (UDP 5060 only), because one of these examples is on 5062. Are >> these "SIP aware" firewalls that are doing this automatically, as the IPCOP >> did before? >> >> - Is the extra "Contact" data in the last packet example different because >> it is a different UA (HT-503 rather than an HT-287) >> >> - Is Freeswitch not flagging the registration from my office (User4) as >> being NAT'd because it is coming in on the same subnet as the interface >> Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and >> pfsense is at 173.11.22.99)? >> >> Sorry for this terribly long posting - I'm just very curious to understand >> what is going on here, now that I have collected all this information. >> >> Thanks, >> >> Dave >> >> >> > > -- > Sent from my mobile device > > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/30416a9d/attachment-0001.html From tgraziano at myitdepartment.net Sun Aug 29 11:48:39 2010 From: tgraziano at myitdepartment.net (Tony Graziano) Date: Sun, 29 Aug 2010 14:48:39 -0400 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <13431520.91283104831264.JavaMail.root@zimbra1.spigotsystems.com> References: <25121409.71283104421357.JavaMail.root@zimbra1.spigotsystems.com> <13431520.91283104831264.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: ANY generic firewall does not have any type of symmetric nat turned on by default. MANY consumer based routers are also INCAPABLE of doing so (MANUAL). The default use for pfsense is AUTOMATIC, but at least it has an easy way to do MANUAL (AON), easier, IMO than IPTABLES. On Sun, Aug 29, 2010 at 2:00 PM, Dave Redmore < dave.redmore at spigotsystems.com> wrote: > Thanks for your help/thoughts Tony. > > I just confirmed that siproxd is not installed - SSH'd in and checked > running processes to be sure. > > Like I said earlier - I can make it work - I have disabled the automatic > outbound NAT and set up AON (Advanced Outbound Nat). It is just that I want > to understand what I am seeing so that I can learn from all this. > > Here is one thing - If Freeswitch flagged my pfsense connection as being > behind NAT - would it then compensate for the Source port being 11521 (per > the packet capture in the original email)? Am I totally wrong in thinking > that it would be "normal" to see packets with the source port changed for > users behind generic NAT firewalls? > > I might try hooking the IPCOP box back up and doing a capture of that, so I > can see what is different between the IPCOP (worked "out of the box") vs. > the pfsense. > > Dave > > > ----- Original Message ----- > From: "Tony Graziano" > To: "FreeSWITCH Users Help" > Sent: Sunday, August 29, 2010 10:53:05 AM GMT -06:00 US/Canada Central > Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... > > If pfsense and your FS install are on the same subnet then "something" must > be sitting in between to randomize the ports. In the 1.2.x install of > pfsense, siproxd is installed by default. There is also a default rule for > port 5060 that is in pfsense. I suggest removing the filters AND rules and > starting from scratch. > > I believe this is also the default, and undesired installation, when > installing a sip system behind it, even for the beta snapshots of pfsense > 2.0. Please check your installed packages. Please remove any rules you did > not create for the network and start over. What you are describing sounds to > me the siproxd IS/WAS installed. It picks up on anything on the LAN and > randomizes the port. It's a very common thing with pfsense. > > On Sun, Aug 29, 2010 at 11:26 AM, Dave Redmore < > dave.redmore at spigotsystems.com> wrote: > >> Confusion abounds here - sorry if I am being obtuse... >> >> A few points on all this - >> >> 1 - sipproxd is NOT installed on the firewall >> >> 2 - I am confused by the source port randomization issue. I think that >> what pfsense does by default is randomize the source port translations, >> rather than using the same source port translations for all connections from >> an internal host. This is completely different from the issue of telling >> pfsense to not change the source port at all - i.e. create a Static Port >> NAT. >> >> 3 - One of the things that I find most confusing about what I saw/see in >> the packet captures is that I EXPECTED to see non-SIP ports as the source >> port for the registration requests. What we commonly call NAT is more >> accurately described as PAT (Port Address Translation) - it functions by >> translating the source port of requests in and out of the firewall. It is >> one of the FEW things that I like about Cisco is that they more accurately >> use the terms NAT and PAT. >> >> 4 - So, that brings me back to why am I NOT seeing random source ports - >> why is Freeswitch NOT tagging my connection from pfsense as being NAT'd? >> >> Dave >> >> >> >> To start receiving Spigot Network's once a month newsletter filled with >> interesting technology news and great offers click SUBSCRIBE >> >> >> ----- Original Message ----- >> From: "Tony Graziano" >> To: "FreeSWITCH Users Help" >> Sent: Sunday, August 29, 2010 7:11:26 AM GMT -06:00 US/Canada Central >> Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... >> >> Yeah, the sipxroxd is in the installed packages on his build. Remove >> the intsalled package and make sure the default rule for outgoing >> traffic is set for manual/static nat, not automatic. >> >> http://blog.myitdepartment.net/?p=37 >> >> On Sun, Aug 29, 2010 at 7:40 AM, Tony Graziano >> wrote: >> > Ipcop has a similar setting to pfsense. You probably missed it. >> > >> > MOST FIREWALLS do not use static port NAT. The default rules for >> > pfsense (and packages) for port 5060 should be removed. >> > >> > On your outbound rule for your LAN static port nat needs to be >> > enabled. Once you do that recreate the nat rules AND remove the >> > siproxd package by default. >> > >> > This is really a pfsense firewall question, it is clear static port >> > was not enabled so the source port was re-written because that is what >> > MOST firewalls do by default. >> > >> > On 8/29/10, Dave Redmore wrote: >> >> Hello All, >> >> >> >> I ran into an issue today that has burned up most of my day >> troubleshooting. >> >> I have resolved the problem, but would really like to understand what >> caused >> >> it, or some of the internal Freeswitch plumbing that is at play so that >> I >> >> can learn something from all of this time I have invested. >> >> >> >> I have a Freeswitch server running that acts as a proxy to an account >> with >> >> an ITSP for doing T38 faxing. The Freeswitch server has a public IP >> address >> >> - there are four "users" who register simple FXS ATAs to my server and >> it >> >> then proxies to the ITSP using the "proxy_media" functionality. It has >> been >> >> working very well for the last 6 months or so. I have never had to deal >> with >> >> any NAT traversal issues - I just point the ATA to the IP to register >> and >> >> everything is great. >> >> >> >> Here is what the four users "looked" like - >> >> >> >> User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> >> Freeswitch >> >> Proxy >> >> User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> >> Freeswitch >> >> Proxy >> >> User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> >> >> Freeswitch Proxy >> >> User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> >> >> Freeswitch Proxy >> >> >> >> (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit >> on >> >> the same Comcast Gateway) >> >> >> >> As I said, this all worked perfectly without any need to "fiddle" with >> >> anything on any firewalls - worked right out of the box. >> >> >> >> So, today I changed out my IPCOP firewall for a pfsense firewall - and >> my >> >> HT-287 would no longer register. >> >> >> >> After much head-scratching, packet captures, etc. I found that I needed >> to >> >> set up a Static Port NAT for the port the HT-287 was using (5062) in >> order >> >> to get this to work. >> >> >> >> So, I see WHAT is happening, but I really want to know WHY it is >> happening. >> >> >> >> Here are the gory details: >> >> >> >> The sofia status of the profile looks like this - when the I have the >> Static >> >> Port NAT in place (details changed for security): >> >> >> >> _______________________________________________________________ >> >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> >> User: 8885554525 at 173.11.22.111 >> >> Contact: "user" >> >> >> ;fs_nat=yes;fs_path=sip%3A8885554525%40173.22.22.55%3A5060> >> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) >> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> >> IP: 173.22.22.55 >> >> Port: 5060 >> >> Auth-User: 8885554525 >> >> Auth-Realm: 173.11.22.111 >> >> MWI-Account: 8885554525 at 173.11.22.111 >> >> >> >> Call-ID: 1716488819-5062-1 at 192.168.7.150 >> >> User: 8885554544 at 173.11.22.111 >> >> Contact: "user" > ;user=phone;fs_nat=yes; >> >> fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> >> >> Agent: Grandstream HT-502 V1.1B 1.0.1.63 >> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) >> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> >> IP: 98.255.0.11 >> >> Port: 5062 >> >> Auth-User: 8885554544 >> >> Auth-Realm: 173.11.22.111 >> >> MWI-Account: 8885554544 at 173.11.22.111 >> >> >> >> Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 >> >> User: 8885554549 at 173.11.22.111 >> >> Contact: "user" >> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> >> Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) >> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> >> IP: 173.11.22.99 >> >> Port: 5062 >> >> Auth-User: 8885554549 >> >> Auth-Realm: 173.11.22.111 >> >> MWI-Account: 8885554549 at 173.11.22.111 >> >> >> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> >> User: 8885554547 at 173.11.22.111 >> >> Contact: "user" > ;user=phone;fs_nat=yes;fs >> >> _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> >> >> Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) >> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >> >> IP: 98.222.55.100 >> >> Port: 5060 >> >> Auth-User: 8885554547 >> >> Auth-Realm: 173.11.22.111 >> >> MWI-Account: 8885554547 at 173.11.22.111 >> >> ___________________________________________________________ >> >> >> >> The "User4" account is in red. The "Contact" field is substantially >> >> different and the "Status" indicates "Registered (UDP)", rather than >> >> "Registered (UDP-NAT)" as the others. >> >> >> >> When I do a packet capture on the external NIC interface (eth0) - I see >> the >> >> following when the HT-287 tries to register and the Static Port NAT is >> NOT >> >> in place: >> >> >> >> ___________________________________________________________________ >> >> Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 >> >> (173.11.22.111) >> >> User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) >> >> Session Initiation Protocol >> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> >> Method: REGISTER >> >> Request-URI: sip:173.11.22.111:5090 >> >> Request-URI Host Part: 173.11.22.111 >> >> Request-URI Host Port: 5090 >> >> Message Header >> >> Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 >> >> Transport: UDP >> >> Sent-by Address: 10.8.11.149 >> >> Sent-by port: 5062 >> >> Branch: z9hG4bKda48f838c8689e41 >> >> From: ;tag=c8a0d452edc5ac4b >> >> SIP from address: sip:8885554549 at 173.11.22.111:5090 >> >> SIP tag: c8a0d452edc5ac4b >> >> To: >> >> Contact: >> >> Contact Binding: >> >> Supported: replaces, timer >> >> Call-ID: aa77d777bae71be6 at 10.8.11.149 >> >> CSeq: 100 REGISTER >> >> Sequence Number: 100 >> >> Method: REGISTER >> >> Expires: 3600 >> >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >> >> Max-Forwards: 70 >> >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> >> Content-Length: 0 >> >> _______________________________________________________________ >> >> >> >> When Freeswitch replies back with a "401 Unauthorized" - asking for >> further >> >> Auth - it replies back to port 5062 - so the packet never comes back >> >> (pfsense is looking for a packet back on port 11521 in this case). >> >> >> >> If I put the Static Port NAT in place - all is well, because the >> "Source" >> >> port shows as "5062" - the rest of the packet looks pretty much the >> same. >> >> >> >> Now, here is a packet coming from one of the other Users - this one >> comes >> >> through a DD-WRT router - here we see that the Source Port is 5060 : >> >> >> >> _________________________________________________________________ >> >> Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 >> >> (173.11.22.111) >> >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> >> Session Initiation Protocol >> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> >> Method: REGISTER >> >> Request-URI: sip:173.11.22.111:5090 >> >> [Resent Packet: False] >> >> Message Header >> >> Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b >> >> Transport: UDP >> >> Sent-by Address: 192.168.1.137 >> >> Branch: z9hG4bK665bc67a1c64292b >> >> From: "fax" ;tag=8dc68b35111c4261 >> >> To: >> >> Contact: >> > >> >> Contact Binding: >> > >> >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >> >> CSeq: 503 REGISTER >> >> Sequence Number: 503 >> >> Method: REGISTER >> >> Expires: 3600 >> >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >> >> Max-Forwards: 70 >> >> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >> >> Content-Length: 0 >> >> ______________________________________________________________________ >> >> >> >> Here is one more packet coming from a Comcast/SMC Router - again, the >> source >> >> port is correct: >> >> >> >> ______________________________________________________________________ >> >> Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: >> 173.11.22.111 >> >> (173.11.22.111) >> >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >> >> Session Initiation Protocol >> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >> >> Message Header >> >> Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport >> >> Transport: UDP >> >> Sent-by Address: 10.1.10.150 >> >> Sent-by port: 5060 >> >> Branch: z9hG4bK58981045 >> >> RPort: rport >> >> From: ;tag=138706651 >> >> To: >> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >> >> CSeq: 79875 REGISTER >> >> Sequence Number: 79875 >> >> Method: REGISTER >> >> Contact: >> >> > ;user=phone>;reg-id=1;+sip.instance="" >> >> Contact Binding: >> >> > ;user=phone>;reg-id=1;+sip.instance="" >> >> Max-Forwards: 70 >> >> User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 >> >> Supported: path >> >> Expires: 300 >> >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, >> REFER, >> >> UPDATE >> >> Content-Length: 0 >> >> ___________________________________________________________ >> >> >> >> So, here are my questions: >> >> >> >> - Why is the Sofia Status so much different for the registration coming >> >> through the pfSense firewall. It looks like it doesn't get tagged as >> being >> >> NAT'd and the "Contact" info is much less. >> >> >> >> - Do most modern routers automatically Static Port NAT any SIP traffic? >> Both >> >> DD-WRT and SMC routers appear to be doing this - and not just on a >> simple >> >> Port bases (UDP 5060 only), because one of these examples is on 5062. >> Are >> >> these "SIP aware" firewalls that are doing this automatically, as the >> IPCOP >> >> did before? >> >> >> >> - Is the extra "Contact" data in the last packet example different >> because >> >> it is a different UA (HT-503 rather than an HT-287) >> >> >> >> - Is Freeswitch not flagging the registration from my office (User4) as >> >> being NAT'd because it is coming in on the same subnet as the interface >> >> Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and >> >> pfsense is at 173.11.22.99)? >> >> >> >> Sorry for this terribly long posting - I'm just very curious to >> understand >> >> what is going on here, now that I have collected all this information. >> >> >> >> Thanks, >> >> >> >> Dave >> >> >> >> >> >> >> > >> > -- >> > Sent from my mobile device >> > >> > ====================== >> > Tony Graziano, Manager >> > Telephone: 434.984.8430 >> > sip: tgraziano at voice.myitdepartment.net >> > Fax: 434.984.8431 >> > >> > Email: tgraziano at myitdepartment.net >> > >> > LAN/Telephony/Security and Control Systems Helpdesk: >> > Telephone: 434.984.8426 >> > sip: helpdesk at voice.myitdepartment.net >> > Fax: 434.984.8427 >> > >> > Helpdesk Contract Customers: >> > http://www.myitdepartment.net/gethelp/ >> > >> > Why do mathematicians always confuse Halloween and Christmas? >> > Because 31 Oct = 25 Dec. >> > >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgraziano at voice.myitdepartment.net >> Fax: 434.984.8431 >> >> Email: tgraziano at myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpdesk at voice.myitdepartment.net >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> Because 31 Oct = 25 Dec. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > > _______________________________________________ FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano at voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano at myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk at voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/614c9fee/attachment-0001.html From mthakershi at gmail.com Sun Aug 29 12:22:03 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Sun, 29 Aug 2010 14:22:03 -0500 Subject: [Freeswitch-users] gTalk new phone calls feature Message-ID: Hello, As you know, Google introduced new "Call US/Canada" free from gMail. Is it possible to make outgoing calls from FreeSwitch to USA number using gTalk/gMail? Thank you for help/guidance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/80c090fb/attachment.html From brokendash at gmail.com Sun Aug 29 14:16:11 2010 From: brokendash at gmail.com (broken dash) Date: Sun, 29 Aug 2010 16:16:11 -0500 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: References: <25121409.71283104421357.JavaMail.root@zimbra1.spigotsystems.com> <13431520.91283104831264.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: Yeah, I read more into this last night and your 100% right on about most firewalls not supporting this out of the box...You should hookup your IPCOP box and unload the ip_conntrack_sip & ip_nat_sip modules to see if it breaks in a similar fashion. Brian On Sun, Aug 29, 2010 at 1:48 PM, Tony Graziano wrote: > ANY generic firewall does not have any type of symmetric nat turned on by > default. > > MANY consumer based routers are also INCAPABLE of doing so (MANUAL). > > The default use for pfsense is AUTOMATIC, but at least it has an easy way > to do MANUAL (AON), easier, IMO than IPTABLES. > > > > On Sun, Aug 29, 2010 at 2:00 PM, Dave Redmore < > dave.redmore at spigotsystems.com> wrote: > >> Thanks for your help/thoughts Tony. >> >> I just confirmed that siproxd is not installed - SSH'd in and checked >> running processes to be sure. >> >> Like I said earlier - I can make it work - I have disabled the automatic >> outbound NAT and set up AON (Advanced Outbound Nat). It is just that I want >> to understand what I am seeing so that I can learn from all this. >> >> Here is one thing - If Freeswitch flagged my pfsense connection as being >> behind NAT - would it then compensate for the Source port being 11521 (per >> the packet capture in the original email)? Am I totally wrong in thinking >> that it would be "normal" to see packets with the source port changed for >> users behind generic NAT firewalls? >> >> I might try hooking the IPCOP box back up and doing a capture of that, so >> I can see what is different between the IPCOP (worked "out of the box") vs. >> the pfsense. >> >> Dave >> >> >> ----- Original Message ----- >> From: "Tony Graziano" >> To: "FreeSWITCH Users Help" >> Sent: Sunday, August 29, 2010 10:53:05 AM GMT -06:00 US/Canada Central >> Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... >> >> If pfsense and your FS install are on the same subnet then "something" >> must be sitting in between to randomize the ports. In the 1.2.x install of >> pfsense, siproxd is installed by default. There is also a default rule for >> port 5060 that is in pfsense. I suggest removing the filters AND rules and >> starting from scratch. >> >> I believe this is also the default, and undesired installation, when >> installing a sip system behind it, even for the beta snapshots of pfsense >> 2.0. Please check your installed packages. Please remove any rules you did >> not create for the network and start over. What you are describing sounds to >> me the siproxd IS/WAS installed. It picks up on anything on the LAN and >> randomizes the port. It's a very common thing with pfsense. >> >> On Sun, Aug 29, 2010 at 11:26 AM, Dave Redmore < >> dave.redmore at spigotsystems.com> wrote: >> >>> Confusion abounds here - sorry if I am being obtuse... >>> >>> A few points on all this - >>> >>> 1 - sipproxd is NOT installed on the firewall >>> >>> 2 - I am confused by the source port randomization issue. I think that >>> what pfsense does by default is randomize the source port translations, >>> rather than using the same source port translations for all connections from >>> an internal host. This is completely different from the issue of telling >>> pfsense to not change the source port at all - i.e. create a Static Port >>> NAT. >>> >>> 3 - One of the things that I find most confusing about what I saw/see in >>> the packet captures is that I EXPECTED to see non-SIP ports as the source >>> port for the registration requests. What we commonly call NAT is more >>> accurately described as PAT (Port Address Translation) - it functions by >>> translating the source port of requests in and out of the firewall. It is >>> one of the FEW things that I like about Cisco is that they more accurately >>> use the terms NAT and PAT. >>> >>> 4 - So, that brings me back to why am I NOT seeing random source ports - >>> why is Freeswitch NOT tagging my connection from pfsense as being NAT'd? >>> >>> Dave >>> >>> >>> >>> To start receiving Spigot Network's once a month newsletter filled with >>> interesting technology news and great offers click SUBSCRIBE >>> >>> >>> ----- Original Message ----- >>> From: "Tony Graziano" >>> To: "FreeSWITCH Users Help" >>> Sent: Sunday, August 29, 2010 7:11:26 AM GMT -06:00 US/Canada Central >>> Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... >>> >>> Yeah, the sipxroxd is in the installed packages on his build. Remove >>> the intsalled package and make sure the default rule for outgoing >>> traffic is set for manual/static nat, not automatic. >>> >>> http://blog.myitdepartment.net/?p=37 >>> >>> On Sun, Aug 29, 2010 at 7:40 AM, Tony Graziano >>> wrote: >>> > Ipcop has a similar setting to pfsense. You probably missed it. >>> > >>> > MOST FIREWALLS do not use static port NAT. The default rules for >>> > pfsense (and packages) for port 5060 should be removed. >>> > >>> > On your outbound rule for your LAN static port nat needs to be >>> > enabled. Once you do that recreate the nat rules AND remove the >>> > siproxd package by default. >>> > >>> > This is really a pfsense firewall question, it is clear static port >>> > was not enabled so the source port was re-written because that is what >>> > MOST firewalls do by default. >>> > >>> > On 8/29/10, Dave Redmore wrote: >>> >> Hello All, >>> >> >>> >> I ran into an issue today that has burned up most of my day >>> troubleshooting. >>> >> I have resolved the problem, but would really like to understand what >>> caused >>> >> it, or some of the internal Freeswitch plumbing that is at play so >>> that I >>> >> can learn something from all of this time I have invested. >>> >> >>> >> I have a Freeswitch server running that acts as a proxy to an account >>> with >>> >> an ITSP for doing T38 faxing. The Freeswitch server has a public IP >>> address >>> >> - there are four "users" who register simple FXS ATAs to my server and >>> it >>> >> then proxies to the ITSP using the "proxy_media" functionality. It has >>> been >>> >> working very well for the last 6 months or so. I have never had to >>> deal with >>> >> any NAT traversal issues - I just point the ATA to the IP to register >>> and >>> >> everything is great. >>> >> >>> >> Here is what the four users "looked" like - >>> >> >>> >> User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> >>> Freeswitch >>> >> Proxy >>> >> User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> >>> Freeswitch >>> >> Proxy >>> >> User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> >>> >> Freeswitch Proxy >>> >> User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> >>> >> Freeswitch Proxy >>> >> >>> >> (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy >>> sit on >>> >> the same Comcast Gateway) >>> >> >>> >> As I said, this all worked perfectly without any need to "fiddle" with >>> >> anything on any firewalls - worked right out of the box. >>> >> >>> >> So, today I changed out my IPCOP firewall for a pfsense firewall - and >>> my >>> >> HT-287 would no longer register. >>> >> >>> >> After much head-scratching, packet captures, etc. I found that I >>> needed to >>> >> set up a Static Port NAT for the port the HT-287 was using (5062) in >>> order >>> >> to get this to work. >>> >> >>> >> So, I see WHAT is happening, but I really want to know WHY it is >>> happening. >>> >> >>> >> Here are the gory details: >>> >> >>> >> The sofia status of the profile looks like this - when the I have the >>> Static >>> >> Port NAT in place (details changed for security): >>> >> >>> >> _______________________________________________________________ >>> >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >>> >> User: 8885554525 at 173.11.22.111 >>> >> Contact: "user" >>> >> >>> ;fs_nat=yes;fs_path=sip%3A8885554525%40173.22.22.55%3A5060> >>> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >>> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) >>> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >>> >> IP: 173.22.22.55 >>> >> Port: 5060 >>> >> Auth-User: 8885554525 >>> >> Auth-Realm: 173.11.22.111 >>> >> MWI-Account: 8885554525 at 173.11.22.111 >>> >> >>> >> Call-ID: 1716488819-5062-1 at 192.168.7.150 >>> >> User: 8885554544 at 173.11.22.111 >>> >> Contact: "user" >> ;user=phone;fs_nat=yes; >>> >> fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> >>> >> Agent: Grandstream HT-502 V1.1B 1.0.1.63 >>> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) >>> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >>> >> IP: 98.255.0.11 >>> >> Port: 5062 >>> >> Auth-User: 8885554544 >>> >> Auth-Realm: 173.11.22.111 >>> >> MWI-Account: 8885554544 at 173.11.22.111 >>> >> >>> >> Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 >>> >> User: 8885554549 at 173.11.22.111 >>> >> Contact: "user" >>> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >>> >> Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) >>> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >>> >> IP: 173.11.22.99 >>> >> Port: 5062 >>> >> Auth-User: 8885554549 >>> >> Auth-Realm: 173.11.22.111 >>> >> MWI-Account: 8885554549 at 173.11.22.111 >>> >> >>> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >>> >> User: 8885554547 at 173.11.22.111 >>> >> Contact: "user" >> ;user=phone;fs_nat=yes;fs >>> >> _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> >>> >> Agent: Grandstream HT-503 V1.1B 1.0.1.63 >>> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) >>> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net >>> >> IP: 98.222.55.100 >>> >> Port: 5060 >>> >> Auth-User: 8885554547 >>> >> Auth-Realm: 173.11.22.111 >>> >> MWI-Account: 8885554547 at 173.11.22.111 >>> >> ___________________________________________________________ >>> >> >>> >> The "User4" account is in red. The "Contact" field is substantially >>> >> different and the "Status" indicates "Registered (UDP)", rather than >>> >> "Registered (UDP-NAT)" as the others. >>> >> >>> >> When I do a packet capture on the external NIC interface (eth0) - I >>> see the >>> >> following when the HT-287 tries to register and the Static Port NAT is >>> NOT >>> >> in place: >>> >> >>> >> ___________________________________________________________________ >>> >> Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: >>> 173.11.22.111 >>> >> (173.11.22.111) >>> >> User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) >>> >> Session Initiation Protocol >>> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >>> >> Method: REGISTER >>> >> Request-URI: sip:173.11.22.111:5090 >>> >> Request-URI Host Part: 173.11.22.111 >>> >> Request-URI Host Port: 5090 >>> >> Message Header >>> >> Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 >>> >> Transport: UDP >>> >> Sent-by Address: 10.8.11.149 >>> >> Sent-by port: 5062 >>> >> Branch: z9hG4bKda48f838c8689e41 >>> >> From: ;tag=c8a0d452edc5ac4b >>> >> SIP from address: sip:8885554549 at 173.11.22.111:5090 >>> >> SIP tag: c8a0d452edc5ac4b >>> >> To: >>> >> Contact: >>> >> Contact Binding: >>> >> Supported: replaces, timer >>> >> Call-ID: aa77d777bae71be6 at 10.8.11.149 >>> >> CSeq: 100 REGISTER >>> >> Sequence Number: 100 >>> >> Method: REGISTER >>> >> Expires: 3600 >>> >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 >>> >> Max-Forwards: 70 >>> >> Allow: >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >>> >> Content-Length: 0 >>> >> _______________________________________________________________ >>> >> >>> >> When Freeswitch replies back with a "401 Unauthorized" - asking for >>> further >>> >> Auth - it replies back to port 5062 - so the packet never comes back >>> >> (pfsense is looking for a packet back on port 11521 in this case). >>> >> >>> >> If I put the Static Port NAT in place - all is well, because the >>> "Source" >>> >> port shows as "5062" - the rest of the packet looks pretty much the >>> same. >>> >> >>> >> Now, here is a packet coming from one of the other Users - this one >>> comes >>> >> through a DD-WRT router - here we see that the Source Port is 5060 : >>> >> >>> >> _________________________________________________________________ >>> >> Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: >>> 173.11.22.111 >>> >> (173.11.22.111) >>> >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >>> >> Session Initiation Protocol >>> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >>> >> Method: REGISTER >>> >> Request-URI: sip:173.11.22.111:5090 >>> >> [Resent Packet: False] >>> >> Message Header >>> >> Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b >>> >> Transport: UDP >>> >> Sent-by Address: 192.168.1.137 >>> >> Branch: z9hG4bK665bc67a1c64292b >>> >> From: "fax" ;tag=8dc68b35111c4261 >>> >> To: >>> >> Contact: >>> > >>> >> Contact Binding: >>> > >>> >> Call-ID: 0e551b3c694a793c at 192.168.1.137 >>> >> CSeq: 503 REGISTER >>> >> Sequence Number: 503 >>> >> Method: REGISTER >>> >> Expires: 3600 >>> >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 >>> >> Max-Forwards: 70 >>> >> Allow: >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE >>> >> Content-Length: 0 >>> >> ______________________________________________________________________ >>> >> >>> >> Here is one more packet coming from a Comcast/SMC Router - again, the >>> source >>> >> port is correct: >>> >> >>> >> ______________________________________________________________________ >>> >> Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: >>> 173.11.22.111 >>> >> (173.11.22.111) >>> >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) >>> >> Session Initiation Protocol >>> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 >>> >> Message Header >>> >> Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport >>> >> Transport: UDP >>> >> Sent-by Address: 10.1.10.150 >>> >> Sent-by port: 5060 >>> >> Branch: z9hG4bK58981045 >>> >> RPort: rport >>> >> From: ;tag=138706651 >>> >> To: >>> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 >>> >> CSeq: 79875 REGISTER >>> >> Sequence Number: 79875 >>> >> Method: REGISTER >>> >> Contact: >>> >> >> ;user=phone>;reg-id=1;+sip.instance="" >>> >> Contact Binding: >>> >> >> ;user=phone>;reg-id=1;+sip.instance="" >>> >> Max-Forwards: 70 >>> >> User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 >>> >> Supported: path >>> >> Expires: 300 >>> >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, >>> REFER, >>> >> UPDATE >>> >> Content-Length: 0 >>> >> ___________________________________________________________ >>> >> >>> >> So, here are my questions: >>> >> >>> >> - Why is the Sofia Status so much different for the registration >>> coming >>> >> through the pfSense firewall. It looks like it doesn't get tagged as >>> being >>> >> NAT'd and the "Contact" info is much less. >>> >> >>> >> - Do most modern routers automatically Static Port NAT any SIP >>> traffic? Both >>> >> DD-WRT and SMC routers appear to be doing this - and not just on a >>> simple >>> >> Port bases (UDP 5060 only), because one of these examples is on 5062. >>> Are >>> >> these "SIP aware" firewalls that are doing this automatically, as the >>> IPCOP >>> >> did before? >>> >> >>> >> - Is the extra "Contact" data in the last packet example different >>> because >>> >> it is a different UA (HT-503 rather than an HT-287) >>> >> >>> >> - Is Freeswitch not flagging the registration from my office (User4) >>> as >>> >> being NAT'd because it is coming in on the same subnet as the >>> interface >>> >> Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and >>> >> pfsense is at 173.11.22.99)? >>> >> >>> >> Sorry for this terribly long posting - I'm just very curious to >>> understand >>> >> what is going on here, now that I have collected all this information. >>> >> >>> >> Thanks, >>> >> >>> >> Dave >>> >> >>> >> >>> >> >>> > >>> > -- >>> > Sent from my mobile device >>> > >>> > ====================== >>> > Tony Graziano, Manager >>> > Telephone: 434.984.8430 >>> > sip: tgraziano at voice.myitdepartment.net >>> > Fax: 434.984.8431 >>> > >>> > Email: tgraziano at myitdepartment.net >>> > >>> > LAN/Telephony/Security and Control Systems Helpdesk: >>> > Telephone: 434.984.8426 >>> > sip: helpdesk at voice.myitdepartment.net >>> > Fax: 434.984.8427 >>> > >>> > Helpdesk Contract Customers: >>> > http://www.myitdepartment.net/gethelp/ >>> > >>> > Why do mathematicians always confuse Halloween and Christmas? >>> > Because 31 Oct = 25 Dec. >>> > >>> >>> >>> >>> -- >>> ====================== >>> Tony Graziano, Manager >>> Telephone: 434.984.8430 >>> sip: tgraziano at voice.myitdepartment.net >>> Fax: 434.984.8431 >>> >>> Email: tgraziano at myitdepartment.net >>> >>> LAN/Telephony/Security and Control Systems Helpdesk: >>> Telephone: 434.984.8426 >>> sip: helpdesk at voice.myitdepartment.net >>> Fax: 434.984.8427 >>> >>> Helpdesk Contract Customers: >>> http://www.myitdepartment.net/gethelp/ >>> >>> Why do mathematicians always confuse Halloween and Christmas? >>> Because 31 Oct = 25 Dec. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgraziano at voice.myitdepartment.net >> Fax: 434.984.8431 >> >> Email: tgraziano at myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpdesk at voice.myitdepartment.net >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> Because 31 Oct = 25 Dec. >> >> >> _______________________________________________ FreeSWITCH-users mailing >> list FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgraziano at voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgraziano at myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpdesk at voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100829/b4fc95ad/attachment-0001.html From sos at sokhapkin.dyndns.org Sun Aug 29 14:26:04 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sun, 29 Aug 2010 17:26:04 -0400 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: References: <25121409.71283104421357.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: <201008291726.04502.sos@sokhapkin.dyndns.org> To my experience ip_conntrack_sip & ip_nat_sip modules break SIP in many ways... On Sunday 29 August 2010, broken dash wrote: > Yeah, I read more into this last night and your 100% right on about most > firewalls not supporting this out of the box...You should hookup your IPCOP > box and unload the ip_conntrack_sip & ip_nat_sip modules to see if it > breaks in a similar fashion. > > Brian > > > On Sun, Aug 29, 2010 at 1:48 PM, Tony Graziano > > > wrote: > > > > ANY generic firewall does not have any type of symmetric nat turned on by > > default. > > > > MANY consumer based routers are also INCAPABLE of doing so (MANUAL). > > > > The default use for pfsense is AUTOMATIC, but at least it has an easy way > > to do MANUAL (AON), easier, IMO than IPTABLES. > > > > > > > > On Sun, Aug 29, 2010 at 2:00 PM, Dave Redmore < > > > > dave.redmore at spigotsystems.com> wrote: > >> Thanks for your help/thoughts Tony. > >> > >> I just confirmed that siproxd is not installed - SSH'd in and checked > >> running processes to be sure. > >> > >> Like I said earlier - I can make it work - I have disabled the automatic > >> outbound NAT and set up AON (Advanced Outbound Nat). It is just that I > >> want to understand what I am seeing so that I can learn from all this. > >> > >> Here is one thing - If Freeswitch flagged my pfsense connection as being > >> behind NAT - would it then compensate for the Source port being 11521 > >> (per the packet capture in the original email)? Am I totally wrong in > >> thinking that it would be "normal" to see packets with the source port > >> changed for users behind generic NAT firewalls? > >> > >> I might try hooking the IPCOP box back up and doing a capture of that, > >> so I can see what is different between the IPCOP (worked "out of the > >> box") vs. the pfsense. > >> > >> Dave > >> > >> > >> ----- Original Message ----- > >> From: "Tony Graziano" > >> To: "FreeSWITCH Users Help" > >> Sent: Sunday, August 29, 2010 10:53:05 AM GMT -06:00 US/Canada Central > >> Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... > >> > >> If pfsense and your FS install are on the same subnet then "something" > >> must be sitting in between to randomize the ports. In the 1.2.x install > >> of pfsense, siproxd is installed by default. There is also a default > >> rule for port 5060 that is in pfsense. I suggest removing the filters > >> AND rules and starting from scratch. > >> > >> I believe this is also the default, and undesired installation, when > >> installing a sip system behind it, even for the beta snapshots of > >> pfsense 2.0. Please check your installed packages. Please remove any > >> rules you did not create for the network and start over. What you are > >> describing sounds to me the siproxd IS/WAS installed. It picks up on > >> anything on the LAN and randomizes the port. It's a very common thing > >> with pfsense. > >> > >> On Sun, Aug 29, 2010 at 11:26 AM, Dave Redmore < > >> > >> dave.redmore at spigotsystems.com> wrote: > >>> Confusion abounds here - sorry if I am being obtuse... > >>> > >>> A few points on all this - > >>> > >>> 1 - sipproxd is NOT installed on the firewall > >>> > >>> 2 - I am confused by the source port randomization issue. I think that > >>> what pfsense does by default is randomize the source port translations, > >>> rather than using the same source port translations for all connections > >>> from an internal host. This is completely different from the issue of > >>> telling pfsense to not change the source port at all - i.e. create a > >>> Static Port NAT. > >>> > >>> 3 - One of the things that I find most confusing about what I saw/see > >>> in the packet captures is that I EXPECTED to see non-SIP ports as the > >>> source port for the registration requests. What we commonly call NAT > >>> is more accurately described as PAT (Port Address Translation) - it > >>> functions by translating the source port of requests in and out of the > >>> firewall. It is one of the FEW things that I like about Cisco is that > >>> they more accurately use the terms NAT and PAT. > >>> > >>> 4 - So, that brings me back to why am I NOT seeing random source ports > >>> - why is Freeswitch NOT tagging my connection from pfsense as being > >>> NAT'd? > >>> > >>> Dave > >>> > >>> > >>> > >>> To start receiving Spigot Network's once a month newsletter filled with > >>> interesting technology news and great offers click > >>> SUBSCRIBE > >>> > >>> > >>> ----- Original Message ----- > >>> From: "Tony Graziano" > >>> To: "FreeSWITCH Users Help" > >>> Sent: Sunday, August 29, 2010 7:11:26 AM GMT -06:00 US/Canada Central > >>> Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... > >>> > >>> Yeah, the sipxroxd is in the installed packages on his build. Remove > >>> the intsalled package and make sure the default rule for outgoing > >>> traffic is set for manual/static nat, not automatic. > >>> > >>> http://blog.myitdepartment.net/?p=37 > >>> > >>> On Sun, Aug 29, 2010 at 7:40 AM, Tony Graziano > >>> > >>> wrote: > >>> > Ipcop has a similar setting to pfsense. You probably missed it. > >>> > > >>> > MOST FIREWALLS do not use static port NAT. The default rules for > >>> > pfsense (and packages) for port 5060 should be removed. > >>> > > >>> > On your outbound rule for your LAN static port nat needs to be > >>> > enabled. Once you do that recreate the nat rules AND remove the > >>> > siproxd package by default. > >>> > > >>> > This is really a pfsense firewall question, it is clear static port > >>> > was not enabled so the source port was re-written because that is > >>> > what MOST firewalls do by default. > >>> > > >>> > On 8/29/10, Dave Redmore wrote: > >>> >> Hello All, > >>> >> > >>> >> I ran into an issue today that has burned up most of my day > >>> > >>> troubleshooting. > >>> > >>> >> I have resolved the problem, but would really like to understand > >>> >> what > >>> > >>> caused > >>> > >>> >> it, or some of the internal Freeswitch plumbing that is at play so > >>> > >>> that I > >>> > >>> >> can learn something from all of this time I have invested. > >>> >> > >>> >> I have a Freeswitch server running that acts as a proxy to an > >>> >> account > >>> > >>> with > >>> > >>> >> an ITSP for doing T38 faxing. The Freeswitch server has a public IP > >>> > >>> address > >>> > >>> >> - there are four "users" who register simple FXS ATAs to my server > >>> >> and > >>> > >>> it > >>> > >>> >> then proxies to the ITSP using the "proxy_media" functionality. It > >>> >> has > >>> > >>> been > >>> > >>> >> working very well for the last 6 months or so. I have never had to > >>> > >>> deal with > >>> > >>> >> any NAT traversal issues - I just point the ATA to the IP to > >>> >> register > >>> > >>> and > >>> > >>> >> everything is great. > >>> >> > >>> >> Here is what the four users "looked" like - > >>> >> > >>> >> User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> > >>> > >>> Freeswitch > >>> > >>> >> Proxy > >>> >> User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> > >>> > >>> Freeswitch > >>> > >>> >> Proxy > >>> >> User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet > >>> >> -> Freeswitch Proxy > >>> >> User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway > >>> >> -> Freeswitch Proxy > >>> >> > >>> >> (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy > >>> > >>> sit on > >>> > >>> >> the same Comcast Gateway) > >>> >> > >>> >> As I said, this all worked perfectly without any need to "fiddle" > >>> >> with anything on any firewalls - worked right out of the box. > >>> >> > >>> >> So, today I changed out my IPCOP firewall for a pfsense firewall - > >>> >> and > >>> > >>> my > >>> > >>> >> HT-287 would no longer register. > >>> >> > >>> >> After much head-scratching, packet captures, etc. I found that I > >>> > >>> needed to > >>> > >>> >> set up a Static Port NAT for the port the HT-287 was using (5062) in > >>> > >>> order > >>> > >>> >> to get this to work. > >>> >> > >>> >> So, I see WHAT is happening, but I really want to know WHY it is > >>> > >>> happening. > >>> > >>> >> Here are the gory details: > >>> >> > >>> >> The sofia status of the profile looks like this - when the I have > >>> >> the > >>> > >>> Static > >>> > >>> >> Port NAT in place (details changed for security): > >>> >> > >>> >> _______________________________________________________________ > >>> >> Call-ID: 0e551b3c694a793c at 192.168.1.137 > >>> >> User: 8885554525 at 173.11.22.111 > >>> >> Contact: "user" > >>> >> > >>> > >>> ;fs_nat=yes;fs_path=sip%3A8885554525%40173.22.22.55%3A5060> > >>> > >>> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > >>> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) > >>> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > >>> >> IP: 173.22.22.55 > >>> >> Port: 5060 > >>> >> Auth-User: 8885554525 > >>> >> Auth-Realm: 173.11.22.111 > >>> >> MWI-Account: 8885554525 at 173.11.22.111 > >>> >> > >>> >> Call-ID: 1716488819-5062-1 at 192.168.7.150 > >>> >> User: 8885554544 at 173.11.22.111 > >>> >> Contact: "user" >>> > >>> ;user=phone;fs_nat=yes; > >>> > >>> >> fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> > >>> >> Agent: Grandstream HT-502 V1.1B 1.0.1.63 > >>> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) > >>> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > >>> >> IP: 98.255.0.11 > >>> >> Port: 5062 > >>> >> Auth-User: 8885554544 > >>> >> Auth-Realm: 173.11.22.111 > >>> >> MWI-Account: 8885554544 at 173.11.22.111 > >>> >> > >>> >> Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 > >>> >> User: 8885554549 at 173.11.22.111 > >>> >> Contact: "user" > >>> >> Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > >>> >> Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) > >>> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > >>> >> IP: 173.11.22.99 > >>> >> Port: 5062 > >>> >> Auth-User: 8885554549 > >>> >> Auth-Realm: 173.11.22.111 > >>> >> MWI-Account: 8885554549 at 173.11.22.111 > >>> >> > >>> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 > >>> >> User: 8885554547 at 173.11.22.111 > >>> >> Contact: "user" >>> > >>> ;user=phone;fs_nat=yes;fs > >>> > >>> >> _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> > >>> >> Agent: Grandstream HT-503 V1.1B 1.0.1.63 > >>> >> Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) > >>> >> Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > >>> >> IP: 98.222.55.100 > >>> >> Port: 5060 > >>> >> Auth-User: 8885554547 > >>> >> Auth-Realm: 173.11.22.111 > >>> >> MWI-Account: 8885554547 at 173.11.22.111 > >>> >> ___________________________________________________________ > >>> >> > >>> >> The "User4" account is in red. The "Contact" field is substantially > >>> >> different and the "Status" indicates "Registered (UDP)", rather than > >>> >> "Registered (UDP-NAT)" as the others. > >>> >> > >>> >> When I do a packet capture on the external NIC interface (eth0) - I > >>> > >>> see the > >>> > >>> >> following when the HT-287 tries to register and the Static Port NAT > >>> >> is > >>> > >>> NOT > >>> > >>> >> in place: > >>> >> > >>> >> ___________________________________________________________________ > >>> >> Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: > >>> > >>> 173.11.22.111 > >>> > >>> >> (173.11.22.111) > >>> >> User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 > >>> >> (5090) Session Initiation Protocol > >>> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > >>> >> Method: REGISTER > >>> >> Request-URI: sip:173.11.22.111:5090 > >>> >> Request-URI Host Part: 173.11.22.111 > >>> >> Request-URI Host Port: 5090 > >>> >> Message Header > >>> >> Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 > >>> >> Transport: UDP > >>> >> Sent-by Address: 10.8.11.149 > >>> >> Sent-by port: 5062 > >>> >> Branch: z9hG4bKda48f838c8689e41 > >>> >> From: ;tag=c8a0d452edc5ac4b > >>> >> SIP from address: sip:8885554549 at 173.11.22.111:5090 > >>> >> SIP tag: c8a0d452edc5ac4b > >>> >> To: > >>> >> Contact: > >>> >> Contact Binding: > >>> >> Supported: replaces, timer > >>> >> Call-ID: aa77d777bae71be6 at 10.8.11.149 > >>> >> CSeq: 100 REGISTER > >>> >> Sequence Number: 100 > >>> >> Method: REGISTER > >>> >> Expires: 3600 > >>> >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > >>> >> Max-Forwards: 70 > >>> >> Allow: > >>> > >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > >>> > >>> >> Content-Length: 0 > >>> >> _______________________________________________________________ > >>> >> > >>> >> When Freeswitch replies back with a "401 Unauthorized" - asking for > >>> > >>> further > >>> > >>> >> Auth - it replies back to port 5062 - so the packet never comes back > >>> >> (pfsense is looking for a packet back on port 11521 in this case). > >>> >> > >>> >> If I put the Static Port NAT in place - all is well, because the > >>> > >>> "Source" > >>> > >>> >> port shows as "5062" - the rest of the packet looks pretty much the > >>> > >>> same. > >>> > >>> >> Now, here is a packet coming from one of the other Users - this one > >>> > >>> comes > >>> > >>> >> through a DD-WRT router - here we see that the Source Port is 5060 : > >>> >> > >>> >> _________________________________________________________________ > >>> >> Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: > >>> > >>> 173.11.22.111 > >>> > >>> >> (173.11.22.111) > >>> >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > >>> >> Session Initiation Protocol > >>> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > >>> >> Method: REGISTER > >>> >> Request-URI: sip:173.11.22.111:5090 > >>> >> [Resent Packet: False] > >>> >> Message Header > >>> >> Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b > >>> >> Transport: UDP > >>> >> Sent-by Address: 192.168.1.137 > >>> >> Branch: z9hG4bK665bc67a1c64292b > >>> >> From: "fax" ;tag=8dc68b35111c4261 > >>> >> To: > >>> >> Contact: > >>> >> > >>> >> > >>> >> Contact Binding: > >>> >> > >>> >> > >>> >> Call-ID: 0e551b3c694a793c at 192.168.1.137 > >>> >> CSeq: 503 REGISTER > >>> >> Sequence Number: 503 > >>> >> Method: REGISTER > >>> >> Expires: 3600 > >>> >> User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > >>> >> Max-Forwards: 70 > >>> >> Allow: > >>> > >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > >>> > >>> >> Content-Length: 0 > >>> >> ____________________________________________________________________ > >>> >>__ > >>> >> > >>> >> Here is one more packet coming from a Comcast/SMC Router - again, > >>> >> the > >>> > >>> source > >>> > >>> >> port is correct: > >>> >> > >>> >> ____________________________________________________________________ > >>> >>__ Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: > >>> > >>> 173.11.22.111 > >>> > >>> >> (173.11.22.111) > >>> >> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > >>> >> Session Initiation Protocol > >>> >> Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > >>> >> Message Header > >>> >> Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport > >>> >> Transport: UDP > >>> >> Sent-by Address: 10.1.10.150 > >>> >> Sent-by port: 5060 > >>> >> Branch: z9hG4bK58981045 > >>> >> RPort: rport > >>> >> From: ;tag=138706651 > >>> >> To: > >>> >> Call-ID: 1035241259-5060-1 at 10.1.10.150 > >>> >> CSeq: 79875 REGISTER > >>> >> Sequence Number: 79875 > >>> >> Method: REGISTER > >>> >> Contact: > >>> >> >>> > >>> ;user=phone>;reg-id=1;+sip.instance=" >>>000B821F9A84>" > >>> > >>> >> Contact Binding: > >>> >> >>> > >>> ;user=phone>;reg-id=1;+sip.instance=" >>>000B821F9A84>" > >>> > >>> >> Max-Forwards: 70 > >>> >> User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 > >>> >> Supported: path > >>> >> Expires: 300 > >>> >> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > >>> > >>> REFER, > >>> > >>> >> UPDATE > >>> >> Content-Length: 0 > >>> >> ___________________________________________________________ > >>> >> > >>> >> So, here are my questions: > >>> >> > >>> >> - Why is the Sofia Status so much different for the registration > >>> > >>> coming > >>> > >>> >> through the pfSense firewall. It looks like it doesn't get tagged as > >>> > >>> being > >>> > >>> >> NAT'd and the "Contact" info is much less. > >>> >> > >>> >> - Do most modern routers automatically Static Port NAT any SIP > >>> > >>> traffic? Both > >>> > >>> >> DD-WRT and SMC routers appear to be doing this - and not just on a > >>> > >>> simple > >>> > >>> >> Port bases (UDP 5060 only), because one of these examples is on > >>> >> 5062. > >>> > >>> Are > >>> > >>> >> these "SIP aware" firewalls that are doing this automatically, as > >>> >> the > >>> > >>> IPCOP > >>> > >>> >> did before? > >>> >> > >>> >> - Is the extra "Contact" data in the last packet example different > >>> > >>> because > >>> > >>> >> it is a different UA (HT-503 rather than an HT-287) > >>> >> > >>> >> - Is Freeswitch not flagging the registration from my office (User4) > >>> > >>> as > >>> > >>> >> being NAT'd because it is coming in on the same subnet as the > >>> > >>> interface > >>> > >>> >> Freeswitch received the packet on (Freeswitch is at 173.11.22.111 > >>> >> and pfsense is at 173.11.22.99)? > >>> >> > >>> >> Sorry for this terribly long posting - I'm just very curious to > >>> > >>> understand > >>> > >>> >> what is going on here, now that I have collected all this > >>> >> information. > >>> >> > >>> >> Thanks, > >>> >> > >>> >> Dave > >>> > > >>> > -- > >>> > Sent from my mobile device > >>> > > >>> > ====================== > >>> > Tony Graziano, Manager > >>> > Telephone: 434.984.8430 > >>> > sip: tgraziano at voice.myitdepartment.net > >>> > Fax: 434.984.8431 > >>> > > >>> > Email: tgraziano at myitdepartment.net > >>> > > >>> > LAN/Telephony/Security and Control Systems Helpdesk: > >>> > Telephone: 434.984.8426 > >>> > sip: helpdesk at voice.myitdepartment.net > >>> > Fax: 434.984.8427 > >>> > > >>> > Helpdesk Contract Customers: > >>> > http://www.myitdepartment.net/gethelp/ > >>> > > >>> > Why do mathematicians always confuse Halloween and Christmas? > >>> > Because 31 Oct = 25 Dec. > >>> > >>> -- > >>> ====================== > >>> Tony Graziano, Manager > >>> Telephone: 434.984.8430 > >>> sip: tgraziano at voice.myitdepartment.net > >>> Fax: 434.984.8431 > >>> > >>> Email: tgraziano at myitdepartment.net > >>> > >>> LAN/Telephony/Security and Control Systems Helpdesk: > >>> Telephone: 434.984.8426 > >>> sip: helpdesk at voice.myitdepartment.net > >>> Fax: 434.984.8427 > >>> > >>> Helpdesk Contract Customers: > >>> http://www.myitdepartment.net/gethelp/ > >>> > >>> Why do mathematicians always confuse Halloween and Christmas? > >>> Because 31 Oct = 25 Dec. > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > >>>s http://www.freeswitch.org > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > >>>s http://www.freeswitch.org > >> > >> -- > >> ====================== > >> Tony Graziano, Manager > >> Telephone: 434.984.8430 > >> sip: tgraziano at voice.myitdepartment.net > >> Fax: 434.984.8431 > >> > >> Email: tgraziano at myitdepartment.net > >> > >> LAN/Telephony/Security and Control Systems Helpdesk: > >> Telephone: 434.984.8426 > >> sip: helpdesk at voice.myitdepartment.net > >> Fax: 434.984.8427 > >> > >> Helpdesk Contract Customers: > >> http://www.myitdepartment.net/gethelp/ > >> > >> Why do mathematicians always confuse Halloween and Christmas? > >> Because 31 Oct = 25 Dec. > >> > >> > >> _______________________________________________ FreeSWITCH-users mailing > >> list FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE > >>: http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > sip: tgraziano at voice.myitdepartment.net > > Fax: 434.984.8431 > > > > Email: tgraziano at myitdepartment.net > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > sip: helpdesk at voice.myitdepartment.net > > Fax: 434.984.8427 > > > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > > > Why do mathematicians always confuse Halloween and Christmas? > > Because 31 Oct = 25 Dec. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > From david.ponzone at ipeva.fr Sun Aug 29 15:06:36 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 30 Aug 2010 00:06:36 +0200 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> References: <10182070.661283065275853.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: Dave, I misread your mail the first time and did not see you sent traces. I think there are some interesting things in those. For the packet coming from your HT-287 without the static port NAT: Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 -> rport is missing For the packet coming from the one behind a DD-WRT: rport is missing too, but the source port in the Via matches the source port of the packet, so it works, the same way it works with your pfSense if you add the static port NAT. But why FS manages to guess it's behind NAT eludes me, but the NAT detecting algo in FS is clearly complex. For the packet coming from the HT-503: I think you made a mistake. You said at the beginning of your mail that it is behind DD-WRT, but before the trace, you say "one more packet coming from Comcast/SMC". Anyway, this one is interesting. rport is there, so HT-503 is rport-capable (but HT-287 is not). I would check if a configuration or a firmware upgrade could enable rport on the HT-287. Also, you wonder if modern routers have some automatic static NAT. Actually, no, but what they do quite often is to not change the source port of the packet if this port is available on the external interface. For instance, if your device sends a packet from port 5060, and this port is free on the external side of the router, it will preserve 5060. Then if a second device sends a packet from port 5060, as it is already used, it will use a random port. You end up having a pseudo static NAT behaviour for the first device on your network. I saw that on business routers from Draytek, Funkwerk and others. So perhaps your ipcop was doing that ? Wild guess: if you add a second phone with the same source port 5060 behind a DD-WRT router, I am pretty sure you will have issues with its registration. About your question "is FreeSWITCH not tagging the device as behind nat because it is on the same subnet as pfSense ?". That's quite possible. I think there are places in FS conf where you define what is local. I think it's the special keyword localnet.auto. But I really think the end of all isues is rport. If rport is not available on your device, you can still force rport on FS side: David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 29/08/2010 ? 09:01, Dave Redmore a ?crit : > Hello All, > > I ran into an issue today that has burned up most of my day > troubleshooting. I have resolved the problem, but would really like > to understand what caused it, or some of the internal Freeswitch > plumbing that is at play so that I can learn something from all of > this time I have invested. > > I have a Freeswitch server running that acts as a proxy to an > account with an ITSP for doing T38 faxing. The Freeswitch server > has a public IP address - there are four "users" who register simple > FXS ATAs to my server and it then proxies to the ITSP using the > "proxy_media" functionality. It has been working very well for the > last 6 months or so. I have never had to deal with any NAT > traversal issues - I just point the ATA to the IP to register and > everything is great. > > Here is what the four users "looked" like - > > User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> > Freeswitch Proxy > User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> > Freeswitch Proxy > User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet - > > Freeswitch Proxy > User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway > -> Freeswitch Proxy > > (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy > sit on the same Comcast Gateway) > > As I said, this all worked perfectly without any need to "fiddle" > with anything on any firewalls - worked right out of the box. > > So, today I changed out my IPCOP firewall for a pfsense firewall - > and my HT-287 would no longer register. > > After much head-scratching, packet captures, etc. I found that I > needed to set up a Static Port NAT for the port the HT-287 was using > (5062) in order to get this to work. > > So, I see WHAT is happening, but I really want to know WHY it is > happening. > > Here are the gory details: > > The sofia status of the profile looks like this - when the I have > the Static Port NAT in place (details changed for security): > > _______________________________________________________________ > Call-ID: 0e551b3c694a793c at 192.168.1.137 > User: 8885554525 at 173.11.22.111 > Contact: "user" > > Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 173.22.22.55 > Port: 5060 > Auth-User: 8885554525 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554525 at 173.11.22.111 > > Call-ID: 1716488819-5062-1 at 192.168.7.150 > User: 8885554544 at 173.11.22.111 > Contact: "user" ; fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> > Agent: Grandstream HT-502 V1.1B 1.0.1.63 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 98.255.0.11 > Port: 5062 > Auth-User: 8885554544 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554544 at 173.11.22.111 > > Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 > User: 8885554549 at 173.11.22.111 > Contact: "user" > Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 173.11.22.99 > Port: 5062 > Auth-User: 8885554549 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554549 at 173.11.22.111 > > Call-ID: 1035241259-5060-1 at 10.1.10.150 > User: 8885554547 at 173.11.22.111 > Contact: "user" _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> > Agent: Grandstream HT-503 V1.1B 1.0.1.63 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 98.222.55.100 > Port: 5060 > Auth-User: 8885554547 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554547 at 173.11.22.111 > ___________________________________________________________ > > The "User4" account is in red. The "Contact" field is substantially > different and the "Status" indicates "Registered (UDP)", rather than > "Registered (UDP-NAT)" as the others. > > When I do a packet capture on the external NIC interface (eth0) - I > see the following when the HT-287 tries to register and the Static > Port NAT is NOT in place: > > ___________________________________________________________________ > Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Method: REGISTER > Request-URI: sip:173.11.22.111:5090 > Request-URI Host Part: 173.11.22.111 > Request-URI Host Port: 5090 > Message Header > Via: SIP/2.0/UDP > 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 > Transport: UDP > Sent-by Address: 10.8.11.149 > Sent-by port: 5062 > Branch: z9hG4bKda48f838c8689e41 > From: ;tag=c8a0d452edc5ac4b > SIP from address: sip:8885554549 at 173.11.22.111:5090 > SIP tag: c8a0d452edc5ac4b > To: > Contact: > Contact Binding: > Supported: replaces, timer > Call-ID: aa77d777bae71be6 at 10.8.11.149 > CSeq: 100 REGISTER > Sequence Number: 100 > Method: REGISTER > Expires: 3600 > User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > _______________________________________________________________ > > When Freeswitch replies back with a "401 Unauthorized" - asking for > further Auth - it replies back to port 5062 - so the packet never > comes back (pfsense is looking for a packet back on port 11521 in > this case). > > If I put the Static Port NAT in place - all is well, because the > "Source" port shows as "5062" - the rest of the packet looks pretty > much the same. > > Now, here is a packet coming from one of the other Users - this one > comes through a DD-WRT router - here we see that the Source Port is > 5060 : > > _________________________________________________________________ > Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Method: REGISTER > Request-URI: sip:173.11.22.111:5090 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b > Transport: UDP > Sent-by Address: 192.168.1.137 > Branch: z9hG4bK665bc67a1c64292b > From: "fax" 8885554525 at 173.11.22.111:5090>;tag=8dc68b35111c4261 > To: > Contact: > Contact Binding: > Call-ID: 0e551b3c694a793c at 192.168.1.137 > CSeq: 503 REGISTER > Sequence Number: 503 > Method: REGISTER > Expires: 3600 > User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > ______________________________________________________________________ > > Here is one more packet coming from a Comcast/SMC Router - again, > the source port is correct: > > ______________________________________________________________________ > Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Message Header > Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport > Transport: UDP > Sent-by Address: 10.1.10.150 > Sent-by port: 5060 > Branch: z9hG4bK58981045 > RPort: rport > From: 8885554547 at 173.11.22.111:5090;user=phone>;tag=138706651 > To: > Call-ID: 1035241259-5060-1 at 10.1.10.150 > CSeq: 79875 REGISTER > Sequence Number: 79875 > Method: REGISTER > Contact: ;reg- > id=1;+sip.instance="" > Contact Binding: >;reg-id=1;+sip.instance=" >" > Max-Forwards: 70 > User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 > Supported: path > Expires: 300 > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, > INFO, REFER, UPDATE > Content-Length: 0 > ___________________________________________________________ > > So, here are my questions: > > - Why is the Sofia Status so much different for the registration > coming through the pfSense firewall. It looks like it doesn't get > tagged as being NAT'd and the "Contact" info is much less. > > - Do most modern routers automatically Static Port NAT any SIP > traffic? Both DD-WRT and SMC routers appear to be doing this - and > not just on a simple Port bases (UDP 5060 only), because one of > these examples is on 5062. Are these "SIP aware" firewalls that are > doing this automatically, as the IPCOP did before? > > - Is the extra "Contact" data in the last packet example different > because it is a different UA (HT-503 rather than an HT-287) > > - Is Freeswitch not flagging the registration from my office (User4) > as being NAT'd because it is coming in on the same subnet as the > interface Freeswitch received the packet on (Freeswitch is at > 173.11.22.111 and pfsense is at 173.11.22.99)? > > Sorry for this terribly long posting - I'm just very curious to > understand what is going on here, now that I have collected all this > information. > > Thanks, > > Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/467e751e/attachment-0001.html From ktngl at yahoo.co.uk Sun Aug 29 17:57:49 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Mon, 30 Aug 2010 00:57:49 +0000 (GMT) Subject: [Freeswitch-users] Esl rubymod get variable In-Reply-To: <0B59088C-7817-4E1F-9BC7-A87799975829@avgs.ca> Message-ID: <685568.62090.qm@web29211.mail.ird.yahoo.com> I tried a few things as suggested but I am not getting any where val = @con.execute("uuid_getvar", "#{@uuid} ivrflag") or val = @con.execute("get", "ivrflag") does not get returned with the value of? ivrflag. What is the correct way to retrieve channel variable data? --- On Sun, 29/8/10, Mathieu Rene wrote: From: Mathieu Rene Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 17:31 uuid_getvarhttp://wiki.freeswitch.org/wiki/Mod_commands#uuid_getvar Mathieu ReneAvant-Garde Solutions IncOffice: + 1 (514) 664-1044 x100Cell: +1 (514) 664-1044 x200mrene at avgs.ca On 2010-08-29, at 12:38 PM, Nigel Kent wrote: Thanks for that tip. I want to find out as well how to retrieve the? value with the opposite of the 'set' command which is how the value was added in the first place. Is a get command not available ?. --- On Sun, 29/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 16:18 you either can wait for a new event coming, or use the uuid_dump API to get everything. @con.execute("uuid_dump", uuid).getBody On Sun, Aug 29, 2010 at 9:40 PM, Nigel Kent wrote: That is when an event occurs. I am wanting to get current value of a custom set variable (like application get) example custom variable ivrflag is set to 0 @con.execute("set", "ivrflag=0") Then later on? it may be set to 1. Now I want to check the current state. What would be the syntax to get the current value of ivrflag --- On Sun, 29/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 12:57 while e = conn.recvEvent?? ? ? ? ? ?name = e.getHeader("Event-Name")?? ? ? ?var = e.getHeader("variable_blah")?? ? ? ?puts e.serializeend On Sun, Aug 29, 2010 at 8:36 PM, Nigel Kent wrote: How can the value of a variable be retrieved in esl ruby _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/da4fa753/attachment.html From dujinfang at gmail.com Sun Aug 29 18:50:34 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 30 Aug 2010 09:50:34 +0800 Subject: [Freeswitch-users] Esl rubymod get variable In-Reply-To: <685568.62090.qm@web29211.mail.ird.yahoo.com> References: <0B59088C-7817-4E1F-9BC7-A87799975829@avgs.ca> <685568.62090.qm@web29211.mail.ird.yahoo.com> Message-ID: @con.api("uuid_getvar", "#@uuid} variable_ivrflag") On Mon, Aug 30, 2010 at 8:57 AM, Nigel Kent wrote: > I tried a few things as suggested but I am not getting any where > > val = @con.execute("uuid_getvar", "#{@uuid} ivrflag") > or > val = @con.execute("get", "ivrflag") > does not get returned with the value of ivrflag. > > What is the correct way to retrieve channel variable data? > > --- On *Sun, 29/8/10, Mathieu Rene * wrote: > > > From: Mathieu Rene > > Subject: Re: [Freeswitch-users] Esl rubymod get variable > To: "FreeSWITCH Users Help" > Date: Sunday, 29 August, 2010, 17:31 > > > uuid_getvar > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_getvar > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 2010-08-29, at 12:38 PM, Nigel Kent wrote: > > Thanks for that tip. > > I want to find out as well how to retrieve the value with the opposite of > the 'set' command which is how the value was added in the first place. > > Is a get command not available ?. > > > > --- On *Sun, 29/8/10, Seven Du > >* wrote: > > > From: Seven Du > > > Subject: Re: [Freeswitch-users] Esl rubymod get variable > To: "FreeSWITCH Users Help" > > > Date: Sunday, 29 August, 2010, 16:18 > > you either can wait for a new event coming, or use the uuid_dump API to get > everything. > > @con.execute("uuid_dump", uuid).getBody > > On Sun, Aug 29, 2010 at 9:40 PM, Nigel Kent wrote: > > That is when an event occurs. I am wanting to get current value of a custom > set variable (like application get) > > example custom variable ivrflag is set to 0 > > @con.execute("set", "ivrflag=0") > Then later on it may be set to 1. > > Now I want to check the current state. What would be the syntax to get the > current value of ivrflag > > > > --- On *Sun, 29/8/10, Seven Du * wrote: > > > From: Seven Du > Subject: Re: [Freeswitch-users] Esl rubymod get variable > To: "FreeSWITCH Users Help" > Date: Sunday, 29 August, 2010, 12:57 > > > while e = conn.recvEvent > name = e.getHeader("Event-Name") > var = e.getHeader("variable_blah") > puts e.serialize > end > > > On Sun, Aug 29, 2010 at 8:36 PM, Nigel Kent > > wrote: > > How can the value of a variable be retrieved in esl ruby > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > -----Inline Attachment Follows----- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/f07e3b49/attachment-0001.html From covici at ccs.covici.com Sun Aug 29 22:11:56 2010 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Mon, 30 Aug 2010 01:11:56 -0400 Subject: [Freeswitch-users] freetdm config problem Message-ID: <20381.1283145116@ccs.covici.com> Hi. I was trying to config freetdm, but it complained that the name was not provided and would not load the module. Here is the freetdm.xml followed by the freetdm.conf. I have an fxs port on channel 1 and an fxo port on channel 4 -- using dahdi. Thanks for any assistance. [span zt-fxs] name => FreeTDM-fxs number => 1 fxo-channel => 4 [span zt-fxo] name => FreeTDM-fxo number => 2 fxs-channel => 1 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From sameer2k3t at gmail.com Mon Aug 30 00:06:16 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Mon, 30 Aug 2010 12:06:16 +0500 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: Hi dear Giovani, i need to use multiple instances of same skype username, i googled but i could not find skype 2.0.72. on latest skype it doesnt work can you please provide me if you have it with you... please upload it somewhere and provide the link and also add link to wiki page so it will be helpful for everyone. ' Thanks On Mon, Aug 23, 2010 at 9:26 PM, Giovanni Maruzzelli wrote: > As I yet told you, you have to find someone that has it, and that has > it in the static build. > > But this is a problem only if you use multiple skype interfaces, I believe. > > Anyway, I'll contact you off list > > -giovanni > > On Sun, Aug 22, 2010 at 4:06 PM, Shamun toha md > wrote: > > Hi, Which skype to use? Isn't that 2.0.72 version ancient skype? IF > 2.0.72 > > to be used, where can we download this for Fedora/CentOS? Any idea > guys!!! > > > > Which Skype Client to use on Linux > > > > Use the static build of the stable Skype client (2.0.72). > > > > Don't use the build for your distro, neither the 'dynamic build'. > > > > Don't use the beta Skype client or more recent "stable" (2.0.72 is the > one > > you want to use) if you want to have multiple channels with the same > > skypename. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/55475d48/attachment.html From ktngl at yahoo.co.uk Mon Aug 30 01:46:46 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Mon, 30 Aug 2010 08:46:46 +0000 (GMT) Subject: [Freeswitch-users] Esl rubymod get variable In-Reply-To: Message-ID: <432566.83356.qm@web29217.mail.ird.yahoo.com> that worked thanks --- On Mon, 30/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Monday, 30 August, 2010, 1:50 @con.api("uuid_getvar", "#@uuid} variable_ivrflag") On Mon, Aug 30, 2010 at 8:57 AM, Nigel Kent wrote: I tried a few things as suggested but I am not getting any where val = @con.execute("uuid_getvar", "#{@uuid} ivrflag") or val = @con.execute("get", "ivrflag") does not get returned with the value of? ivrflag. What is the correct way to retrieve channel variable data? --- On Sun, 29/8/10, Mathieu Rene wrote: From: Mathieu Rene Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 17:31 uuid_getvarhttp://wiki.freeswitch.org/wiki/Mod_commands#uuid_getvar Mathieu ReneAvant-Garde Solutions IncOffice: + 1 (514) 664-1044 x100Cell: +1 (514) 664-1044 x200mrene at avgs.ca On 2010-08-29, at 12:38 PM, Nigel Kent wrote: Thanks for that tip. I want to find out as well how to retrieve the? value with the opposite of the 'set' command which is how the value was added in the first place. Is a get command not available ?. --- On Sun, 29/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 16:18 you either can wait for a new event coming, or use the uuid_dump API to get everything. @con.execute("uuid_dump", uuid).getBody On Sun, Aug 29, 2010 at 9:40 PM, Nigel Kent wrote: That is when an event occurs. I am wanting to get current value of a custom set variable (like application get) example custom variable ivrflag is set to 0 @con.execute("set", "ivrflag=0") Then later on? it may be set to 1. Now I want to check the current state. What would be the syntax to get the current value of ivrflag --- On Sun, 29/8/10, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] Esl rubymod get variable To: "FreeSWITCH Users Help" Date: Sunday, 29 August, 2010, 12:57 while e = conn.recvEvent?? ? ? ? ? ?name = e.getHeader("Event-Name")?? ? ? ?var = e.getHeader("variable_blah")?? ? ? ?puts e.serializeend On Sun, Aug 29, 2010 at 8:36 PM, Nigel Kent wrote: How can the value of a variable be retrieved in esl ruby _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/fc7e936d/attachment-0001.html From gmaruzz at celliax.org Mon Aug 30 03:35:45 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Aug 2010 12:35:45 +0200 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: Sameer, I'm not sure I can distribute Skype client, and I'm exposed as mod_skypopen developer. So, please ask someone else to give you the ***static*** build of 2.0.72 -giovanni On Mon, Aug 30, 2010 at 9:06 AM, Sameer Khan wrote: > Hi dear Giovani, > i need to use multiple instances of same skype username, i googled but i > could not find skype 2.0.72. on latest skype it doesnt work > can you please provide me if you have it with you... please upload it > somewhere and provide the link and also add link to wiki page so it will ?be > helpful for everyone. > ' > Thanks > > On Mon, Aug 23, 2010 at 9:26 PM, Giovanni Maruzzelli > wrote: >> >> As I yet told you, you have to find someone that has it, and that has >> it in the static build. >> >> But this is a problem only if you use multiple skype interfaces, I >> believe. >> >> Anyway, I'll contact you off list >> >> -giovanni >> >> On Sun, Aug 22, 2010 at 4:06 PM, Shamun toha md >> wrote: >> > Hi, Which skype to use? Isn't that 2.0.72 version ancient skype? IF >> > 2.0.72 >> > to be used, where can we download this for Fedora/CentOS? Any idea >> > guys!!! >> > >> > Which Skype Client to use on Linux >> > >> > Use the static build of the stable Skype client (2.0.72). >> > >> > Don't use the build for your distro, neither the 'dynamic build'. >> > >> > Don't use the beta Skype client or more recent "stable" (2.0.72 is the >> > one >> > you want to use) if you want to have multiple channels with the same >> > skypename. >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From sameer2k3t at gmail.com Mon Aug 30 06:03:54 2010 From: sameer2k3t at gmail.com (Sameer Khan) Date: Mon, 30 Aug 2010 18:03:54 +0500 Subject: [Freeswitch-users] hello every one Message-ID: Please, I'm looking for skype static build 2.0.72. Would you guys contact me off list? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/98d16020/attachment.html From mustafa.pk at gmail.com Mon Aug 30 06:29:52 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Mon, 30 Aug 2010 18:29:52 +0500 Subject: [Freeswitch-users] hello every one In-Reply-To: References: Message-ID: Hi, Please set a proper subject line when sending emails (specially on mailing lists) now regarding your question rather demand, this is not a proper forum to ask for an old skype build/version, however i hope someone might be able to help you here. Best of luck. -m On Mon, Aug 30, 2010 at 6:03 PM, Sameer Khan wrote: > Please, I'm looking for skype static build 2.0.72. > > Would you guys contact me off list? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/360b383e/attachment.html From dave.redmore at spigotsystems.com Mon Aug 30 06:43:09 2010 From: dave.redmore at spigotsystems.com (Dave Redmore) Date: Mon, 30 Aug 2010 08:43:09 -0500 (CDT) Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <22012409.141283175777923.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: <15376238.161283175789348.JavaMail.root@zimbra1.spigotsystems.com> Thanks for the thoughtful response. I think that it has been by some happy accident that I have not run into more NAT issues. I usually expect to have problems and then rarely do. For example, I have a FreePBX server sitting behind the pfSense as well (on a different subnet) which is my main office phone system, it is connected to my ITSP (Teliax) with absolutely no problems - no port forward or static port NATing needed - as soon as I cut over to the pfSense, I was up and running. I think you are right re: with the rport issue. At the end of the day, I think the combination of Freeswitch not detecting NAT and the HT-287 not supporting rport was what caused the problem. Will be doing some more investigating on all this - in particular I am now curious to look into how/why my RTP streams work. I have about a dozen customers running various combination's of FreePBX/ATAs/Etc and have never had audio issues - again, probably a happy accident, but now I am motivated to understand the dance that is happening between everything. Thanks again, Dave ----- Original Message ----- From: "David Ponzone" To: "FreeSWITCH Users Help" Sent: Sunday, August 29, 2010 5:06:36 PM GMT -06:00 US/Canada Central Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... Dave, I misread your mail the first time and did not see you sent traces. I think there are some interesting things in those. For the packet coming from your HT-287 without the static port NAT: Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 -> rport is missing For the packet coming from the one behind a DD-WRT: rport is missing too, but the source port in the Via matches the source port of the packet, so it works, the same way it works with your pfSense if you add the static port NAT. But why FS manages to guess it's behind NAT eludes me, but the NAT detecting algo in FS is clearly complex. For the packet coming from the HT-503: I think you made a mistake. You said at the beginning of your mail that it is behind DD-WRT, but before the trace, you say "one more packet coming from Comcast/SMC". Anyway, this one is interesting. rport is there, so HT-503 is rport-capable (but HT-287 is not). I would check if a configuration or a firmware upgrade could enable rport on the HT-287. Also, you wonder if modern routers have some automatic static NAT. Actually, no, but what they do quite often is to not change the source port of the packet if this port is available on the external interface. For instance, if your device sends a packet from port 5060, and this port is free on the external side of the router, it will preserve 5060. Then if a second device sends a packet from port 5060, as it is already used, it will use a random port. You end up having a pseudo static NAT behaviour for the first device on your network. I saw that on business routers from Draytek, Funkwerk and others. So perhaps your ipcop was doing that ? Wild guess: if you add a second phone with the same source port 5060 behind a DD-WRT router, I am pretty sure you will have issues with its registration. About your question "is FreeSWITCH not tagging the device as behind nat because it is on the same subnet as pfSense ?". That's quite possible. I think there are places in FS conf where you define what is local. I think it's the special keyword localnet.auto. But I really think the end of all isues is rport. If rport is not available on your device, you can still force rport on FS side: David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IP eva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 29/08/2010 ? 09:01, Dave Redmore a ?crit : Hello All, I ran into an issue today that has burned up most of my day troubleshooting. I have resolved the problem, but would really like to understand what caused it, or some of the internal Freeswitch plumbing that is at play so that I can learn something from all of this time I have invested. I have a Freeswitch server running that acts as a proxy to an account with an ITSP for doing T38 faxing. The Freeswitch server has a public IP address - there are four "users" who register simple FXS ATAs to my server and it then proxies to the ITSP using the "proxy_media" functionality. It has been working very well for the last 6 months or so. I have never had to deal with any NAT traversal issues - I just point the ATA to the IP to register and everything is great. Here is what the four users "looked" like - User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> Freeswitch Proxy User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> Freeswitch Proxy User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> Freeswitch Proxy User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> Freeswitch Proxy (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit on the same Comcast Gateway) As I said, this all worked perfectly without any need to "fiddle" with anything on any firewalls - worked right out of the box. So, today I changed out my IPCOP firewall for a pfsense firewall - and my HT-287 would no longer register. After much head-scratching, packet captures, etc. I found that I needed to set up a Static Port NAT for the port the HT-287 was using (5062) in order to get this to work. So, I see WHAT is happening, but I really want to know WHY it is happening. Here are the gory details: The sofia status of the profile looks like this - when the I have the Static Port NAT in place (details changed for security): _______________________________________________________________ Call-ID: 0e551b3c694a793c at 192.168.1.137 User: 8885554525 at 173.11.22.111 Contact: "user" < sip:8885554525 at 192.168.1.137;fs_nat=yes;fs_path=sip%3A8885554525%40173.22.22.55%3A5060 > Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 173.22.22.55 Port: 5060 Auth-User: 8885554525 Auth-Realm: 173.11.22.111 MWI-Account: 8885554525 at 173.11.22.111 Call-ID: 1716488819-5062-1 at 192.168.7.150 User: 8885554544 at 173.11.22.111 Contact: "user" < sip:8885554544 at 192.168.7.150:5062;user=phone;fs_nat=yes ; fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> Agent: Grandstream HT-502 V1.1B 1.0.1.63 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 98.255.0.11 Port: 5062 Auth-User: 8885554544 Auth-Realm: 173.11.22.111 MWI-Account: 8885554544 at 173.11.22.111 Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 User: 8885554549 at 173.11.22.111 Contact: "user" < sip:8885554549 at 10.8.11.149:5062 > Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 173.11.22.99 Port: 5062 Auth-User: 8885554549 Auth-Realm: 173.11.22.111 MWI-Account: 8885554549 at 173.11.22.111 Call-ID: 1035241259-5060-1 at 10.1.10.150 User: 8885554547 at 173.11.22.111 Contact: "user" < sip:8885554547 at 10.1.10.150:5060;user=phone;fs_nat=yes;fs _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> Agent: Grandstream HT-503 V1.1B 1.0.1.63 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 98.222.55.100 Port: 5060 Auth-User: 8885554547 Auth-Realm: 173.11.22.111 MWI-Account: 8885554547 at 173.11.22.111 ___________________________________________________________ The "User4" account is in red. The "Contact" field is substantially different and the "Status" indicates "Registered (UDP)", rather than "Registered (UDP-NAT)" as the others. When I do a packet capture on the external NIC interface (eth0) - I see the following when the HT-287 tries to register and the Static Port NAT is NOT in place: ___________________________________________________________________ Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Method: REGISTER Request-URI: sip:173.11.22.111:5090 Request-URI Host Part: 173.11.22.111 Request-URI Host Port: 5090 Message Header Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 Transport: UDP Sent-by Address: 10.8.11.149 Sent-by port: 5062 Branch: z9hG4bKda48f838c8689e41 From: < sip:8885554549 at 173.11.22.111:5090 >;tag=c8a0d452edc5ac4b SIP from address: sip:8885554549 at 173.11.22.111:5090 SIP tag: c8a0d452edc5ac4b To: < sip:8885554549 at 173.11.22.111:5090 > Contact: < sip:88855564549 at 10.8.11.149:5062 > Contact Binding: < sip:8885554549 at 10.8.11.149:5062 > Supported: replaces, timer Call-ID: aa77d777bae71be6 at 10.8.11.149 CSeq: 100 REGISTER Sequence Number: 100 Method: REGISTER Expires: 3600 User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 _______________________________________________________________ When Freeswitch replies back with a "401 Unauthorized" - asking for further Auth - it replies back to port 5062 - so the packet never comes back (pfsense is looking for a packet back on port 11521 in this case). If I put the Static Port NAT in place - all is well, because the "Source" port shows as "5062" - the rest of the packet looks pretty much the same. Now, here is a packet coming from one of the other Users - this one comes through a DD-WRT router - here we see that the Source Port is 5060 : _________________________________________________________________ Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Method: REGISTER Request-URI: sip:173.11.22.111:5090 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b Transport: UDP Sent-by Address: 192.168.1.137 Branch: z9hG4bK665bc67a1c64292b From: "fax" < sip:8885554525 at 173.11.22.111:5090 >;tag=8dc68b35111c4261 To: < sip:8156564525 at 173.15.28.101:5090 > Contact: < sip:8885554525 at 192.168.1.137 > Contact Binding: < sip:8885554525 at 192.168.1.137 > Call-ID: 0e551b3c694a793c at 192.168.1.137 CSeq: 503 REGISTER Sequence Number: 503 Method: REGISTER Expires: 3600 User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 ______________________________________________________________________ Here is one more packet coming from a Comcast/SMC Router - again, the source port is correct: ______________________________________________________________________ Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport Transport: UDP Sent-by Address: 10.1.10.150 Sent-by port: 5060 Branch: z9hG4bK58981045 RPort: rport From: < sip:8885554547 at 173.11.22.111:5090;user=phone >;tag=138706651 To: < sip:8885554547 at 173.11.22.111:5090;user=phone > Call-ID: 1035241259-5060-1 at 10.1.10.150 CSeq: 79875 REGISTER Sequence Number: 79875 Method: REGISTER Contact: < sip:8885554547 at 10.1.10.150:5060;user=phone >;reg-id=1;+sip.instance="< urn:uuid:00000000-0000-1000-8000-000B821F9A84 >" Contact Binding: < sip:8885554547 at 10.1.10.150:5060;user=phone >;reg-id=1;+sip.instance="< urn:uuid:00000000-0000-1000-8000-000B821F9A84 >" Max-Forwards: 70 User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 Supported: path Expires: 300 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 ___________________________________________________________ So, here are my questions: - Why is the Sofia Status so much different for the registration coming through the pfSense firewall. It looks like it doesn't get tagged as being NAT'd and the "Contact" info is much less. - Do most modern routers automatically Static Port NAT any SIP traffic? Both DD-WRT and SMC routers appear to be doing this - and not just on a simple Port bases (UDP 5060 only), because one of these examples is on 5062. Are these "SIP aware" firewalls that are doing this automatically, as the IPCOP did before? - Is the extra "Contact" data in the last packet example different because it is a different UA (HT-503 rather than an HT-287) - Is Freeswitch not flagging the registration from my office (User4) as being NAT'd because it is coming in on the same subnet as the interface Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and pfsense is at 173.11.22.99)? Sorry for this terribly long posting - I'm just very curious to understand what is going on here, now that I have collected all this information. Thanks, Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/6d5b5c9d/attachment-0001.html From dujinfang at gmail.com Mon Aug 30 06:52:43 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 30 Aug 2010 21:52:43 +0800 Subject: [Freeswitch-users] windows build error Message-ID: Hi, I'm new to windows build. I cloned code a few days ago and pulled to head. Can could not build sofia. c:\workspace\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(119) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory I tried to rebuild but the result was the same. I can start freeswitch though mod_sofia and mod file format (not exactly remember) cannot load. Any hint? I will re-clone code and try again tomorrow. Thanks. -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From david.varnes at gmail.com Mon Aug 30 07:06:14 2010 From: david.varnes at gmail.com (david varnes) Date: Tue, 31 Aug 2010 00:06:14 +1000 Subject: [Freeswitch-users] [announce] First release of Java ESL Client library Message-ID: Hi FreeSWITCHERs, I have finally got around to lining up all the bits I wanted in order to do a first release (version 0.9.2) of the Java ESL Client that has been in the freeswitch-contrib git repository for some time now. First thanks to the people who checked it out, compiled it and reported some bugs. (I believe I have fixed them all :-) I have created a couple of new wiki pages for the Java and ESL topics: [1] to cover the various options that I know about to use Java for connecting to the very useful and versatile Event Socket. If there are other options or info, you know what to do, it is a wiki. [2] to describe my contributed Java ESL Client, with download, compile, and basic usage documented. The details are there, a quick summary of this client library features: * Apache License (ASL) version 2 * Standalone ready to use Inbound client * Framework classes to very easily create an Outbound socket client * based on the excellent Netty [3] nio library * logging via slf4j * only runtime dependencies are slf4j-api and netty (MIT and Apache licensed) * single jar which is a valid OSGi bundle * built using maven, * jar, source and javadocs available from maven central (GPG signed) * source is committed as an eclipse project * good coverage of java docs Now it is available as released binary jars it is easier to experiment with. Roadmap: Tentative next steps are outlined in [2] and a TODO file in git. After wider usage and feedback on the API design I would aim to release a version 1.0 and then simply track any ongoing changes in the core FreeSWITCH ESL. I enjoy getting feedback if you find it useful or have issues, patches or enhancement ideas. I know a few projects around the place have been using it even in un-released state :-) Hope it finds use out there ... davidv [1] http://wiki.freeswitch.org/wiki/Java_ESL [2] http://wiki.freeswitch.org/wiki/Java_ESL_Client [3] http://www.jboss.org/netty -- david varnes e: david.varnes at gmail.com p: +61 404 925 633 From peter.olsson at visionutveckling.se Mon Aug 30 07:13:07 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 30 Aug 2010 16:13:07 +0200 Subject: [Freeswitch-users] windows build error In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05818D@cooper> Make sure to disable all kinds of CRLF/LF conversions. By default git on Windows seems to enable this. There is a setting called autocrlf, make sure it's set to false. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Seven Du [dujinfang at gmail.com] Skickat: den 30 augusti 2010 15:52 Till: freeswitch-users ?mne: [Freeswitch-users] windows build error Hi, I'm new to windows build. I cloned code a few days ago and pulled to head. Can could not build sofia. c:\workspace\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(119) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory I tried to rebuild but the result was the same. I can start freeswitch though mod_sofia and mod file format (not exactly remember) cannot load. Any hint? I will re-clone code and try again tomorrow. Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c7bb97c32934187118539! From moises.silva at gmail.com Mon Aug 30 07:16:32 2010 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 30 Aug 2010 10:16:32 -0400 Subject: [Freeswitch-users] freetdm config problem In-Reply-To: <20381.1283145116@ccs.covici.com> References: <20381.1283145116@ccs.covici.com> Message-ID: On Mon, Aug 30, 2010 at 1:11 AM, wrote: > Hi. I was trying to config freetdm, but it complained that the name was > not provided and would not load the module. > > Go to pastebin.freeswitch.org and paste the relevant parts of the error logged, then paste a link here. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/eb64a3af/attachment.html From david.varnes at gmail.com Mon Aug 30 07:25:53 2010 From: david.varnes at gmail.com (david varnes) Date: Tue, 31 Aug 2010 00:25:53 +1000 Subject: [Freeswitch-users] [contrib] a new java ESL inbound/outbound client In-Reply-To: <1281626673.3347.341.camel@damjan-laptop> References: <74a861001001050541v4e02f487xd21fe8fc13e8ed0a@mail.gmail.com> <1281622424.3347.334.camel@damjan-laptop> <1281626673.3347.341.camel@damjan-laptop> Message-ID: Damjan, Thanks for your patch and enthusiastic feedback :-) I have applied it and some other pending fixes/changes and cut a release as per earlier email. See http://wiki.freeswitch.org/wiki/Java_ESL_Client thanks again, davidv On 13 August 2010 01:24, Damjan Jovanovic wrote: > Below is the patch I had to use to get it to compile and run. > > By the way, your ESL library rules. I tried fseslib > (http://versafon.com/versafonweb/Software.jsp) but it's GPL(v3!) and doesn't > handle event bodies; even with a hack to it still doesn't provide essential > fields. The ESL library for Java built into Freeswitch uses native code and > seems to only handle outbound. Only your library provides everything, is > 100% Java, and is licensed nicely. > > Thank you so much > Damjan > [snip] -- david varnes e: david.varnes at gmail.com p: +61 404 925 633 From david.ponzone at ipeva.fr Mon Aug 30 07:38:30 2010 From: david.ponzone at ipeva.fr (David Ponzone) Date: Mon, 30 Aug 2010 16:38:30 +0200 Subject: [Freeswitch-users] NAT traversal questions - (long)... In-Reply-To: <15376238.161283175789348.JavaMail.root@zimbra1.spigotsystems.com> References: <15376238.161283175789348.JavaMail.root@zimbra1.spigotsystems.com> Message-ID: Because for RTP, FS does RTP auto-adjust. Whatever was offered in the SDP, it waits for an inbound stream on the port it announced to the client, and lears the real ip/port from that stream. That's the same mechanism most commercial SBCs use. David Ponzone Direction Technique email: david.ponzone at ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur. Le 30/08/2010 ? 15:43, Dave Redmore a ?crit : > Thanks for the thoughtful response. I think that it has been by > some happy accident that I have not run into more NAT issues. I > usually expect to have problems and then rarely do. For example, I > have a FreePBX server sitting behind the pfSense as well (on a > different subnet) which is my main office phone system, it is > connected to my ITSP (Teliax) with absolutely no problems - no port > forward or static port NATing needed - as soon as I cut over to the > pfSense, I was up and running. > > I think you are right re: with the rport issue. At the end of the > day, I think the combination of Freeswitch not detecting NAT and the > HT-287 not supporting rport was what caused the problem. > > Will be doing some more investigating on all this - in particular I > am now curious to look into how/why my RTP streams work. I have > about a dozen customers running various combination's of FreePBX/ > ATAs/Etc and have never had audio issues - again, probably a happy > accident, but now I am motivated to understand the dance that is > happening between everything. > > Thanks again, > > Dave > > ----- Original Message ----- > From: "David Ponzone" > To: "FreeSWITCH Users Help" > Sent: Sunday, August 29, 2010 5:06:36 PM GMT -06:00 US/Canada Central > Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... > > Dave, > > I misread your mail the first time and did not see you sent traces. > > I think there are some interesting things in those. > > For the packet coming from your HT-287 without the static port NAT: > Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 > -> rport is missing > > For the packet coming from the one behind a DD-WRT: > rport is missing too, but the source port in the Via matches the > source port of the packet, so it works, the same way it works with > your pfSense if you add the static port NAT. > But why FS manages to guess it's behind NAT eludes me, but the NAT > detecting algo in FS is clearly complex. > > For the packet coming from the HT-503: > I think you made a mistake. You said at the beginning of your mail > that it is behind DD-WRT, but before the trace, you say "one more > packet coming from Comcast/SMC". > Anyway, this one is interesting. > rport is there, so HT-503 is rport-capable (but HT-287 is not). > I would check if a configuration or a firmware upgrade could enable > rport on the HT-287. > > Also, you wonder if modern routers have some automatic static NAT. > Actually, no, but what they do quite often is to not change the > source port of the packet if this port is available on the external > interface. > For instance, if your device sends a packet from port 5060, and this > port is free on the external side of the router, it will preserve > 5060. > Then if a second device sends a packet from port 5060, as it is > already used, it will use a random port. > You end up having a pseudo static NAT behaviour for the first device > on your network. > I saw that on business routers from Draytek, Funkwerk and others. > So perhaps your ipcop was doing that ? > Wild guess: if you add a second phone with the same source port 5060 > behind a DD-WRT router, I am pretty sure you will have issues with > its registration. > > About your question "is FreeSWITCH not tagging the device as behind > nat because it is on the same subnet as pfSense ?". > That's quite possible. > I think there are places in FS conf where you define what is local. > I think it's the special keyword localnet.auto. > > But I really think the end of all isues is rport. > If rport is not available on your device, you can still force rport > on FS side: > > > David Ponzone Direction Technique > email: david.ponzone at ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > Ce message et toutes les pi?ces jointes sont confidentiels et > ?tablis ? l'intention exclusive de ses destinataires. Toute > utilisation ou diffusion non autoris?e est interdite. Tout message > ?lectronique est susceptible d'alt?ration. IPeva d?cline toute > responsabilit? au titre de ce message s'il a ?t? alt?r?, > d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce > message, merci de le d?truire imm?diatement et d'avertir > l'exp?diteur. > > > > > Le 29/08/2010 ? 09:01, Dave Redmore a ?crit : > > Hello All, > > I ran into an issue today that has burned up most of my day > troubleshooting. I have resolved the problem, but would really like > to understand what caused it, or some of the internal Freeswitch > plumbing that is at play so that I can learn something from all of > this time I have invested. > > I have a Freeswitch server running that acts as a proxy to an > account with an ITSP for doing T38 faxing. The Freeswitch server > has a public IP address - there are four "users" who register simple > FXS ATAs to my server and it then proxies to the ITSP using the > "proxy_media" functionality. It has been working very well for the > last 6 months or so. I have never had to deal with any NAT > traversal issues - I just point the ATA to the IP to register and > everything is great. > > Here is what the four users "looked" like - > > User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> > Freeswitch Proxy > User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> > Freeswitch Proxy > User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet - > > Freeswitch Proxy > User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway > -> Freeswitch Proxy > > (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy > sit on the same Comcast Gateway) > > As I said, this all worked perfectly without any need to "fiddle" > with anything on any firewalls - worked right out of the box. > > So, today I changed out my IPCOP firewall for a pfsense firewall - > and my HT-287 would no longer register. > > After much head-scratching, packet captures, etc. I found that I > needed to set up a Static Port NAT for the port the HT-287 was using > (5062) in order to get this to work. > > So, I see WHAT is happening, but I really want to know WHY it is > happening. > > Here are the gory details: > > The sofia status of the profile looks like this - when the I have > the Static Port NAT in place (details changed for security): > > _______________________________________________________________ > Call-ID: 0e551b3c694a793c at 192.168.1.137 > User: 8885554525 at 173.11.22.111 > Contact: "user" > > Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 173.22.22.55 > Port: 5060 > Auth-User: 8885554525 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554525 at 173.11.22.111 > > Call-ID: 1716488819-5062-1 at 192.168.7.150 > User: 8885554544 at 173.11.22.111 > Contact: "user" ; fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> > Agent: Grandstream HT-502 V1.1B 1.0.1.63 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 98.255.0.11 > Port: 5062 > Auth-User: 8885554544 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554544 at 173.11.22.111 > > Call-ID: 090ee80e1a0ec9ed at 10.8.11.149 > User: 8885554549 at 173.11.22.111 > Contact: "user" > Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 173.11.22.99 > Port: 5062 > Auth-User: 8885554549 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554549 at 173.11.22.111 > > Call-ID: 1035241259-5060-1 at 10.1.10.150 > User: 8885554547 at 173.11.22.111 > Contact: "user" _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> > Agent: Grandstream HT-503 V1.1B 1.0.1.63 > Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) > Host: 173-11-22-111-illinois.hfc.comcastbusiness.net > IP: 98.222.55.100 > Port: 5060 > Auth-User: 8885554547 > Auth-Realm: 173.11.22.111 > MWI-Account: 8885554547 at 173.11.22.111 > ___________________________________________________________ > > The "User4" account is in red. The "Contact" field is substantially > different and the "Status" indicates "Registered (UDP)", rather than > "Registered (UDP-NAT)" as the others. > > When I do a packet capture on the external NIC interface (eth0) - I > see the following when the HT-287 tries to register and the Static > Port NAT is NOT in place: > > ___________________________________________________________________ > Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Method: REGISTER > Request-URI: sip:173.11.22.111:5090 > Request-URI Host Part: 173.11.22.111 > Request-URI Host Port: 5090 > Message Header > Via: SIP/2.0/UDP > 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 > Transport: UDP > Sent-by Address: 10.8.11.149 > Sent-by port: 5062 > Branch: z9hG4bKda48f838c8689e41 > From: ;tag=c8a0d452edc5ac4b > SIP from address: sip:8885554549 at 173.11.22.111:5090 > SIP tag: c8a0d452edc5ac4b > To: > Contact: > Contact Binding: > Supported: replaces, timer > Call-ID: aa77d777bae71be6 at 10.8.11.149 > CSeq: 100 REGISTER > Sequence Number: 100 > Method: REGISTER > Expires: 3600 > User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > _______________________________________________________________ > > When Freeswitch replies back with a "401 Unauthorized" - asking for > further Auth - it replies back to port 5062 - so the packet never > comes back (pfsense is looking for a packet back on port 11521 in > this case). > > If I put the Static Port NAT in place - all is well, because the > "Source" port shows as "5062" - the rest of the packet looks pretty > much the same. > > Now, here is a packet coming from one of the other Users - this one > comes through a DD-WRT router - here we see that the Source Port is > 5060 : > > _________________________________________________________________ > Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Method: REGISTER > Request-URI: sip:173.11.22.111:5090 > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b > Transport: UDP > Sent-by Address: 192.168.1.137 > Branch: z9hG4bK665bc67a1c64292b > From: "fax" 8885554525 at 173.11.22.111:5090>;tag=8dc68b35111c4261 > To: > Contact: > Contact Binding: > Call-ID: 0e551b3c694a793c at 192.168.1.137 > CSeq: 503 REGISTER > Sequence Number: 503 > Method: REGISTER > Expires: 3600 > User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > ______________________________________________________________________ > > Here is one more packet coming from a Comcast/SMC Router - again, > the source port is correct: > > ______________________________________________________________________ > Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: > 173.11.22.111 (173.11.22.111) > User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) > Session Initiation Protocol > Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 > Message Header > Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport > Transport: UDP > Sent-by Address: 10.1.10.150 > Sent-by port: 5060 > Branch: z9hG4bK58981045 > RPort: rport > From: 8885554547 at 173.11.22.111:5090;user=phone>;tag=138706651 > To: > Call-ID: 1035241259-5060-1 at 10.1.10.150 > CSeq: 79875 REGISTER > Sequence Number: 79875 > Method: REGISTER > Contact: ;reg- > id=1;+sip.instance="" > Contact Binding: >;reg-id=1;+sip.instance=" >" > Max-Forwards: 70 > User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 > Supported: path > Expires: 300 > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, > INFO, REFER, UPDATE > Content-Length: 0 > ___________________________________________________________ > > So, here are my questions: > > - Why is the Sofia Status so much different for the registration > coming through the pfSense firewall. It looks like it doesn't get > tagged as being NAT'd and the "Contact" info is much less. > > - Do most modern routers automatically Static Port NAT any SIP > traffic? Both DD-WRT and SMC routers appear to be doing this - and > not just on a simple Port bases (UDP 5060 only), because one of > these examples is on 5062. Are these "SIP aware" firewalls that are > doing this automatically, as the IPCOP did before? > > - Is the extra "Contact" data in the last packet example different > because it is a different UA (HT-503 rather than an HT-287) > > - Is Freeswitch not flagging the registration from my office (User4) > as being NAT'd because it is coming in on the same subnet as the > interface Freeswitch received the packet on (Freeswitch is at > 173.11.22.111 and pfsense is at 173.11.22.99)? > > Sorry for this terribly long posting - I'm just very curious to > understand what is going on here, now that I have collected all this > information. > > Thanks, > > Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ FreeSWITCH-users > mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/3edeab1f/attachment-0001.html From stevendt at primrosebank.net Mon Aug 30 07:49:12 2010 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 30 Aug 2010 15:49:12 +0100 Subject: [Freeswitch-users] iPhone SIP Client for FreeSwitch ? Message-ID: <40DF06C869E44C0586585623F052F131@bp1.ad.bp.com> Hi, I have an iPhone 4 (not Jailbroken), and now that IPhone OS4 supports multi-tasking, I want to use it with FreeSwitch. I'd be interested to hear what experience others have with iPhone SIP apps connecting to FreeSwitch. There seems to be a few Apps out there which would work, some free, some not. I have one which seems to work (Media5 VOIP) and supports mutli-tasking, i.e., I can have it in the background and it alerts me what a SIP call comes in, so it might be all that I need. I would just like to hear if anyone thinks that there's something better out there- which is not too expensive obviously :-) regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/2e7f1f12/attachment.html From jpitcher at nuvio.com Mon Aug 30 08:13:18 2010 From: jpitcher at nuvio.com (Jonathan Pitcher) Date: Mon, 30 Aug 2010 10:13:18 -0500 Subject: [Freeswitch-users] Mod Fifo Message-ID: Hi all, I am attempting to setup a dynamic fifo group. I actually have the group working and can have agents log into the que and receive calls, but what I have having trouble with is Caller ID on the call to the agent from the que. I have been looking at: http://wiki.freeswitch.org/wiki/Mod_fifo And it says that to change the caller ID to the agent you can change it in 2 places, one in the script that logs the agent in and secondly in the call that gets sent to fifo in. I have choosen the second option because it allows customers to change the name of the que and have it immediately change instead of having to force users to log in and log out again to see the name change take place. When the caller dialing into the que calls the que, this is what I return to FS.
This is what fifo list shows : {fd=1000076.pprd.nuvio.net,}user/638 at 1000076.pprd.nuvio.net {fd=1000076.pprd.nuvio.net,}user/638 at 1000076.pprd.nuvio.net Now everything works like it should but I can't seem to get the origination_caller_id_name variable to be used in the incoming dialplan. What am I missing ? Thanks in advance, Jonathan Pitcher From tayeb.meftah at gmail.com Tue Aug 31 07:28:48 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 31 Aug 2010 16:28:48 +0200 Subject: [Freeswitch-users] iPhone SIP Client for FreeSwitch ? In-Reply-To: <40DF06C869E44C0586585623F052F131@bp1.ad.bp.com> References: <40DF06C869E44C0586585623F052F131@bp1.ad.bp.com> Message-ID: <4C7D11A0.2030504@gmail.com> media5fone is clever work very very well with fs and 100% interoperable +HD audio Le 30/08/2010 16:49, Dave Stevenson a ?crit : > Hi, > I have an iPhone 4 (not Jailbroken), and now that IPhone OS4 supports > multi-tasking, I want to use it with FreeSwitch. > I'd be interested to hear what experience others have with iPhone SIP > apps connecting to FreeSwitch. > There seems to be a few Apps out there which would work, some free, > some not. I have one which seems to work (Media5 VOIP) and supports > mutli-tasking, i.e., I can have it in the background and it alerts me > what a SIP call comes in, so it might be all that I need. I would just > like to hear if anyone thinks that there's something better out there- > which is not too expensive obviously :-) > regards > Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb Sales Manager Ove Telecom direct: +13477595883 INUM: +883510001289101 MSN: sqlsrvx86 at hotmail.com yahoo: vsdevx86 OVETEL USA: +1-212-401-0707 OVETEL UK: +44-20-34110486 OVETEL Ukraine: +380-44-3607309 OVETEL Jordan: +962-6-2508905 OVETEL South Africa: +27-11-4613345 http://www.ovetel.com follow us on Twitter for the latest price updates: http://twitter.com/ovetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/db958e84/attachment.html From tayeb.meftah at gmail.com Tue Aug 31 07:36:29 2010 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 31 Aug 2010 16:36:29 +0200 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: References: Message-ID: <4C7D136D.6040301@gmail.com> if is provided in gtalk yes but if only gmail no Le 29/08/2010 21:22, Malay Thakershi a ?crit : > Hello, > > As you know, Google introduced new "Call US/Canada" free from gMail. > > Is it possible to make outgoing calls from FreeSwitch to USA number > using gTalk/gMail? > > Thank you for help/guidance. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Meftah Tayeb Sales Manager Ove Telecom direct: +13477595883 INUM: +883510001289101 MSN: sqlsrvx86 at hotmail.com yahoo: vsdevx86 OVETEL USA: +1-212-401-0707 OVETEL UK: +44-20-34110486 OVETEL Ukraine: +380-44-3607309 OVETEL Jordan: +962-6-2508905 OVETEL South Africa: +27-11-4613345 http://www.ovetel.com follow us on Twitter for the latest price updates: http://twitter.com/ovetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/4125f833/attachment.html From michaelt at mdevt.com Mon Aug 30 00:08:03 2010 From: michaelt at mdevt.com (Michael Toop) Date: Mon, 30 Aug 2010 09:08:03 +0200 Subject: [Freeswitch-users] Selecting codec by sample rate In-Reply-To: References: Message-ID: Hi, I would also be willing to contribute to the bounty as well, let me know. We need something like this for some faulty phones we have that break on 60ms ptime. Cheers, Michael On Sat, Aug 21, 2010 at 1:27 AM, Kristian Kielhofner wrote: > Tony, > > Thanks, that's what I was trying to asses (trivial vs. non-trivial). > I'm spoiled because most of my requests seem to be trivial and get > fixed in an hour or less :). > > Well the bounty is still up and open. This is something I'm very > interested in and if you get any more details let me know. > > Thanks again! > > On Fri, Aug 20, 2010 at 3:48 PM, Anthony Minessale > wrote: > > The first part is straightforward but the part after the "likewise" > > with the same codec in many rates may be problematic. > > I'll have to think about it but it's not at all trivial of a change > > and will have to be carefully tested to avoid regressions. > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/83835c9a/attachment-0001.html From alarosa at openintellect.eu Mon Aug 30 00:49:16 2010 From: alarosa at openintellect.eu (Alberto La Rosa) Date: Mon, 30 Aug 2010 09:49:16 +0200 (CEST) Subject: [Freeswitch-users] Stream audio file/live to multiple SIP endpoints with IP multicast In-Reply-To: References: Message-ID: Hi Thomas, I was reading old messages on freeswitch-users mailing list and I found one your old request. As I am facing similar issue and I would like to know if you already found a way to handle SIP INVITE to endpoints in order to join the audio stream on IP multicast. I am going to investigate on sofia sip internals in order to understand how to generate SDP that specifies audio stream on IP multicast. Many thanks in adavance. Best Regards, Alberto On Tue, 9 Sep 2008, Sluschny, Thomas wrote: > Hi, > > i want to stream audio (from file and live) from FreeSwitch to multiple > SIP endpoints (these are proprietary). > Because of bandwith limitations we have to use IP multicast for audio > data. > > Now the question: how can i do this smartly in FreeSwitch? > > I have read some Wiki entries: > - mod_esf: seams to use multicast, but i couldn't find usage information > - mod_shout: streaming, but from internet ? MP3, but i want wav-files to > stream > - mod_local_stream: seams to play files in a loop, i want to play a > whole file once. > - mod_conference: seams not to support IP multicast > > Im searching for a simple way to send a SIP INVITE to all my endpoints > and > then stream audio (live from soundcard or from file) over IP multicast > to all these endpoints. > > Thanks in advance, > Thomas From adolfo at delorenzo.mobi Mon Aug 30 03:57:56 2010 From: adolfo at delorenzo.mobi (Adolfo Delorenzo) Date: Mon, 30 Aug 2010 07:57:56 -0300 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: I installed the skype version that was available on the official website and it works like a charm for me with skypopen. I use ubuntu server 10.04. hope this helps. rgds On 30 August 2010 07:35, Giovanni Maruzzelli wrote: > Sameer, > > I'm not sure I can distribute Skype client, and I'm exposed as > mod_skypopen developer. > > So, please ask someone else to give you the ***static*** build of 2.0.72 > > -giovanni > > On Mon, Aug 30, 2010 at 9:06 AM, Sameer Khan wrote: > > Hi dear Giovani, > > i need to use multiple instances of same skype username, i googled but i > > could not find skype 2.0.72. on latest skype it doesnt work > > can you please provide me if you have it with you... please upload it > > somewhere and provide the link and also add link to wiki page so it will > be > > helpful for everyone. > > ' > > Thanks > > > > On Mon, Aug 23, 2010 at 9:26 PM, Giovanni Maruzzelli < > gmaruzz at celliax.org> > > wrote: > >> > >> As I yet told you, you have to find someone that has it, and that has > >> it in the static build. > >> > >> But this is a problem only if you use multiple skype interfaces, I > >> believe. > >> > >> Anyway, I'll contact you off list > >> > >> -giovanni > >> > >> On Sun, Aug 22, 2010 at 4:06 PM, Shamun toha md > >> wrote: > >> > Hi, Which skype to use? Isn't that 2.0.72 version ancient skype? IF > >> > 2.0.72 > >> > to be used, where can we download this for Fedora/CentOS? Any idea > >> > guys!!! > >> > > >> > Which Skype Client to use on Linux > >> > > >> > Use the static build of the stable Skype client (2.0.72). > >> > > >> > Don't use the build for your distro, neither the 'dynamic build'. > >> > > >> > Don't use the beta Skype client or more recent "stable" (2.0.72 is the > >> > one > >> > you want to use) if you want to have multiple channels with the same > >> > skypename. > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> > >> > >> -- > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/da438a2a/attachment.html From covici at ccs.covici.com Mon Aug 30 07:40:22 2010 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Mon, 30 Aug 2010 10:40:22 -0400 Subject: [Freeswitch-users] freetdm config problem In-Reply-To: References: <20381.1283145116@ccs.covici.com> Message-ID: <29535.1283179222@ccs.covici.com> Moises Silva wrote: > On Mon, Aug 30, 2010 at 1:11 AM, wrote: > > > Hi. I was trying to config freetdm, but it complained that the name was > > not provided and would not load the module. > > > > > Go to pastebin.freeswitch.org and paste the relevant parts of the error > logged, then paste a link here. http://pastebin.freeswitch.org/13752 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From covici at ccs.covici.com Mon Aug 30 08:41:58 2010 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Mon, 30 Aug 2010 11:41:58 -0400 Subject: [Freeswitch-users] freetdm config problem In-Reply-To: References: <20381.1283145116@ccs.covici.com> Message-ID: <31184.1283182918@ccs.covici.com> Moises Silva wrote: > On Mon, Aug 30, 2010 at 1:11 AM, wrote: > > > Hi. I was trying to config freetdm, but it complained that the name was > > not provided and would not load the module. > > > > > Go to pastebin.freeswitch.org and paste the relevant parts of the error > logged, then paste a link here. Sorry if this is a dupe, message seems not to have gotten through. Link is: http://pastebin.freeswitch.org/13752 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From fdelawarde at wirelessmundi.com Mon Aug 30 08:49:23 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 30 Aug 2010 17:49:23 +0200 Subject: [Freeswitch-users] directory user without password Message-ID: <1283183363.28748.58.camel@luna.tc.commsmundi.com> Hello, I don't really understand the logic when no password is specified for a specific user. Is it normal that freeswitch still sends an authentication request (401 or similar) in that case? Shouldn't it directly accept and send "200 OK" on REGISTER and INVITE requests when a userid is detected to have no configured password? Thanks, Fran?ois. From rupa at rupa.com Mon Aug 30 08:53:09 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 30 Aug 2010 10:53:09 -0500 Subject: [Freeswitch-users] iPhone SIP Client for FreeSwitch ? In-Reply-To: <40DF06C869E44C0586585623F052F131@bp1.ad.bp.com> References: <40DF06C869E44C0586585623F052F131@bp1.ad.bp.com> Message-ID: I've been happy with acrobit's softphone. On Mon, Aug 30, 2010 at 9:49 AM, Dave Stevenson wrote: > Hi, > > I have an iPhone 4 (not Jailbroken), and now that IPhone OS4 supports > multi-tasking, I want to use it with FreeSwitch. > > I'd be interested to hear what experience others have with iPhone SIP apps > connecting to FreeSwitch. > > There seems to be a few Apps out there which would work, some free, some > not. I have one which seems to work (Media5 VOIP) and supports > mutli-tasking, i.e., I can have it in the background and it alerts me what a > SIP call comes in, so it might be all that I need. I would just like to hear > if anyone thinks that there's something better out there- which is not too > expensive obviously :-) > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/ea791456/attachment.html From brian at freeswitch.org Mon Aug 30 08:59:14 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Aug 2010 10:59:14 -0500 Subject: [Freeswitch-users] directory user without password In-Reply-To: <1283183363.28748.58.camel@luna.tc.commsmundi.com> References: <1283183363.28748.58.camel@luna.tc.commsmundi.com> Message-ID: <013D22DA-74F1-4373-ADC2-97A102EC9B56@freeswitch.org> On Aug 30, 2010, at 10:49 AM, Fran?ois Delawarde wrote: > Hello, > > I don't really understand the logic when no password is specified for a > specific user. > > Is it normal that freeswitch still sends an authentication request (401 > or similar) in that case? Shouldn't it directly accept and send "200 OK" > on REGISTER and INVITE requests when a userid is detected to have no > configured password? Its the normal behavior. > > Thanks, > Fran?ois. From fdelawarde at wirelessmundi.com Mon Aug 30 09:07:43 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 30 Aug 2010 18:07:43 +0200 Subject: [Freeswitch-users] directory user without password In-Reply-To: <013D22DA-74F1-4373-ADC2-97A102EC9B56@freeswitch.org> References: <1283183363.28748.58.camel@luna.tc.commsmundi.com> <013D22DA-74F1-4373-ADC2-97A102EC9B56@freeswitch.org> Message-ID: <1283184463.28748.61.camel@luna.tc.commsmundi.com> On Mon, 2010-08-30 at 10:59 -0500, Brian West wrote: > > Is it normal that freeswitch still sends an authentication request (401 > > or similar) in that case? Shouldn't it directly accept and send "200 OK" > > on REGISTER and INVITE requests when a userid is detected to have no > > configured password? > > Its the normal behavior. Thanks, any way to disable authentication on INVITE on a per-user basis (I know of auth-calls for the profile)? From fdelawarde at wirelessmundi.com Mon Aug 30 09:11:51 2010 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 30 Aug 2010 18:11:51 +0200 Subject: [Freeswitch-users] directory user without password In-Reply-To: <1283184463.28748.61.camel@luna.tc.commsmundi.com> References: <1283183363.28748.58.camel@luna.tc.commsmundi.com> <013D22DA-74F1-4373-ADC2-97A102EC9B56@freeswitch.org> <1283184463.28748.61.camel@luna.tc.commsmundi.com> Message-ID: <1283184711.28748.65.camel@luna.tc.commsmundi.com> On Mon, 2010-08-30 at 18:07 +0200, Fran?ois Delawarde wrote: > Thanks, any way to disable authentication on INVITE on a per-user basis > (I know of auth-calls for the profile)? Nevermind, I just saw that cool dp_tool: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_respond Only bad thing is that I will have to filter my CDRs as using this application from the dialplan (ROUTING) probably generates one. No big deal tho! Fran?ois. From sos at sokhapkin.dyndns.org Mon Aug 30 09:18:31 2010 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 30 Aug 2010 12:18:31 -0400 Subject: [Freeswitch-users] directory user without password In-Reply-To: <1283184711.28748.65.camel@luna.tc.commsmundi.com> References: <1283183363.28748.58.camel@luna.tc.commsmundi.com> <1283184463.28748.61.camel@luna.tc.commsmundi.com> <1283184711.28748.65.camel@luna.tc.commsmundi.com> Message-ID: <201008301218.31773.sos@sokhapkin.dyndns.org> On Monday 30 August 2010, Fran?ois Delawarde wrote: > On Mon, 2010-08-30 at 18:07 +0200, Fran?ois Delawarde wrote: > > Thanks, any way to disable authentication on INVITE on a per-user basis > > (I know of auth-calls for the profile)? > > Nevermind, I just saw that cool dp_tool: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_respond > > Only bad thing is that I will have to filter my CDRs as using this > application from the dialplan (ROUTING) probably generates one. No big > deal tho! > > Fran?ois. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gmaruzz at celliax.org Mon Aug 30 09:25:52 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Aug 2010 18:25:52 +0200 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: On Mon, Aug 30, 2010 at 12:57 PM, Adolfo Delorenzo wrote: > I installed the skype version that was available on the official website and > it works like a charm for me with skypopen. > > I use ubuntu server 10.04. Ciao Adolfo, are you using multiple concurrent skype clients in your setup? And if so, they're using the same skypename? And if so, they're correctly answering multiple concurrent inbound calls to the same skypeuser, and are correctly making multiple concurrent outbound calls? Sorry for the long list or questions, but skype 2.1.81 beta is known not to work on centos (not just with skypopen), and I saw that it works well on Ubuntu 10.04, but was giving problems to me using multiple concurrent instances of the same skypeuser (and I cannot retest it for a couple of weeks). -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From jock.mckechnie at gmail.com Mon Aug 30 11:49:28 2010 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Mon, 30 Aug 2010 13:49:28 -0500 Subject: [Freeswitch-users] Early Media troubles Message-ID: Good arvo, all; I'm battling with a problem in making Early Media play and getting nowhere. I've searched the wiki, forums and the web in general without anything that seems to help. This suggests to me that I'm either doing something stupendously stupid, or there's a very, very odd thing going on here. I've been tasked with something that should be fairly simple: Accept an inbound call Play a specific audio file as early media Pickup the call after a certain time interval Play another audio file into the off-hook channel Disconnect the call. Sounds simple. My extension looks like so: (Replace the 212 number with a real inbound DID which is being routed from Verizon (via SIP) to this FreeSWITCH machine) The call is received, a 183 is sent by FreeSWITCH (pre_answer), it waits the 10 seconds, picks up the call and plays the playback.PCMU file... but there's no early media. I've tcpdumped on the FS machine and verified it definitely is /not/ sending out RTP until the call 200 OKs and the Playback begins. As you can see, I've also attempted to replace the ringback= to a cadence instead of a WAV file, just to ensure it isn't unhappy with the WAV format, with the same results. Boy, I'd sure love it if someone could point out what potentially brain damaged thing I'm doing (assuming it's nothing arcane). My thanks to all; - Jock -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/3910d254/attachment.html From brian at freeswitch.org Mon Aug 30 11:57:29 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Aug 2010 13:57:29 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: Message-ID: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> I already emailed you the solution but i'll reply so everyone will know. Set ringback before you pre_answer. /b On Aug 30, 2010, at 1:49 PM, Jock McKechnie wrote: > Good arvo, all; > > I'm battling with a problem in making Early Media play and getting nowhere. I've searched the wiki, forums and the web in general without anything that seems to help. This suggests to me that I'm either doing something stupendously stupid, or there's a very, very odd thing going on here. > > I've been tasked with something that should be fairly simple: > Accept an inbound call > Play a specific audio file as early media > Pickup the call after a certain time interval > Play another audio file into the off-hook channel > Disconnect the call. > > Sounds simple. My extension looks like so: > > > > > > > > > > > > (Replace the 212 number with a real inbound DID which is being routed from Verizon (via SIP) to this FreeSWITCH machine) > > The call is received, a 183 is sent by FreeSWITCH (pre_answer), it waits the 10 seconds, picks up the call and plays the playback.PCMU file... but there's no early media. I've tcpdumped on the FS machine and verified it definitely is /not/ sending out RTP until the call 200 OKs and the Playback begins. > > As you can see, I've also attempted to replace the ringback= to a cadence instead of a WAV file, just to ensure it isn't unhappy with the WAV format, with the same results. > > Boy, I'd sure love it if someone could point out what potentially brain damaged thing I'm doing (assuming it's nothing arcane). > > My thanks to all; > - Jock From jock.mckechnie at gmail.com Mon Aug 30 12:10:52 2010 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Mon, 30 Aug 2010 14:10:52 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> Message-ID: Thank you, Brian. I have actually omitted an awful lot of "grasping at straws" steps, however I have tried this (and I did it again, just now, both with the .wav ringback and the cadence ringback, just to be certain I hadn't imagined it) and there is still no RTP coming out after the 183. The ringback after pre_answer was tried because of this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready (I also tried ring_ready instead of a pre_answer without any particular progress - although I don't understand, for the same reason as above, why ringback being set after ring_ready actually works... Not that it does for me, anyway ;) So, apparently, not quite _that_ easy. Perhaps something else I've missed? - JP On Mon, Aug 30, 2010 at 1:57 PM, Brian West wrote: > I already emailed you the solution but i'll reply so everyone will know. > > Set ringback before you pre_answer. > > /b > > On Aug 30, 2010, at 1:49 PM, Jock McKechnie wrote: > > > Good arvo, all; > > > > I'm battling with a problem in making Early Media play and getting > nowhere. I've searched the wiki, forums and the web in general without > anything that seems to help. This suggests to me that I'm either doing > something stupendously stupid, or there's a very, very odd thing going on > here. > > > > I've been tasked with something that should be fairly simple: > > Accept an inbound call > > Play a specific audio file as early media > > Pickup the call after a certain time interval > > Play another audio file into the off-hook channel > > Disconnect the call. > > > > Sounds simple. My extension looks like so: > > > > > > > > data="ringback=/usr/local/freeswitch/sounds/early-media.wav"/> > > > > > > > > data="/usr/local/freeswitch/sounds/playback"/> > > > > > > > > (Replace the 212 number with a real inbound DID which is being routed > from Verizon (via SIP) to this FreeSWITCH machine) > > > > The call is received, a 183 is sent by FreeSWITCH (pre_answer), it waits > the 10 seconds, picks up the call and plays the playback.PCMU file... but > there's no early media. I've tcpdumped on the FS machine and verified it > definitely is /not/ sending out RTP until the call 200 OKs and the Playback > begins. > > > > As you can see, I've also attempted to replace the ringback= to a cadence > instead of a WAV file, just to ensure it isn't unhappy with the WAV format, > with the same results. > > > > Boy, I'd sure love it if someone could point out what potentially brain > damaged thing I'm doing (assuming it's nothing arcane). > > > > My thanks to all; > > - Jock > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/7494f554/attachment.html From brian at freeswitch.org Mon Aug 30 12:15:05 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Aug 2010 14:15:05 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> Message-ID: <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> I can tell you without a doubt it works. Ring_Read == 180 ringing, pre_answer == 183 early media. I suspect your firewall or you have nat involved and you're not catching on to what is going on. /b On Aug 30, 2010, at 2:10 PM, Jock McKechnie wrote: > Thank you, Brian. > > I have actually omitted an awful lot of "grasping at straws" steps, however I have tried this (and I did it again, just now, both with the .wav ringback and the cadence ringback, just to be certain I hadn't imagined it) and there is still no RTP coming out after the 183. > > The ringback after pre_answer was tried because of this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready > (I also tried ring_ready instead of a pre_answer without any particular progress - although I don't understand, for the same reason as above, why ringback being set after ring_ready actually works... Not that it does for me, anyway ;) > > So, apparently, not quite _that_ easy. Perhaps something else I've missed? > > - JP From kris at kriskinc.com Mon Aug 30 12:28:37 2010 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 30 Aug 2010 15:28:37 -0400 Subject: [Freeswitch-users] Stream audio file/live to multiple SIP endpoints with IP multicast In-Reply-To: References: Message-ID: Alberto, Any SIP/RTP compliant device should be able to handle unicast SIP invites with a multicast RTP stream specified in the SDP (all you have to do is add /x where x = TTL) to the connection address. On Mon, Aug 30, 2010 at 3:49 AM, Alberto La Rosa wrote: > Hi Thomas, > > I was reading old messages on freeswitch-users mailing list and I found > one your old request. > > As I am facing similar issue and I would like to know if you already found > a way to handle SIP INVITE to endpoints in order to join the audio stream > on IP multicast. > > I am going to investigate on sofia sip internals in order to > understand how to generate SDP that specifies audio stream on > IP multicast. > > Many thanks in adavance. > > Best Regards, > Alberto -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From jock.mckechnie at gmail.com Mon Aug 30 12:35:33 2010 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Mon, 30 Aug 2010 14:35:33 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> Message-ID: And I agree that it must work, somewhere. But it does not. Work with me here: Firstly, I tcpdumped on the FreeSWITCH machine. If I'm not seeing the RTP on the FS machine, it's not going to be anything like a firewall or a NAT issue. Secondly, and far more telling (as the above could be related to me screwing up the tcpdump somehow): I get the Playback audio, so it is possible to get audio back down the line eventually and the RTP does reach the caller, so RTP is able to flow from FS to caller without interference (as I said in the initial mail, the only RTP sent from FS starts _after_ the 200 OK). It simply is that FS is not sending out early media. Like I said, either I'm doing something remarkably stupid, or there is something extremely strange going on here. Should I try to use ring_ready AND pre_answer in combination? I have not attempted that yet. If so, in what order? - JP On Mon, Aug 30, 2010 at 2:15 PM, Brian West wrote: > I can tell you without a doubt it works. Ring_Read == 180 ringing, > pre_answer == 183 early media. I suspect your firewall or you have nat > involved and you're not catching on to what is going on. > > /b > > On Aug 30, 2010, at 2:10 PM, Jock McKechnie wrote: > > > Thank you, Brian. > > > > I have actually omitted an awful lot of "grasping at straws" steps, > however I have tried this (and I did it again, just now, both with the .wav > ringback and the cadence ringback, just to be certain I hadn't imagined it) > and there is still no RTP coming out after the 183. > > > > The ringback after pre_answer was tried because of this: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready > > (I also tried ring_ready instead of a pre_answer without any particular > progress - although I don't understand, for the same reason as above, why > ringback being set after ring_ready actually works... Not that it does for > me, anyway ;) > > > > So, apparently, not quite _that_ easy. Perhaps something else I've > missed? > > > > - JP > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/ef457189/attachment.html From bobc at devassert.com Mon Aug 30 13:20:45 2010 From: bobc at devassert.com (Bob Coleman) Date: Tue, 31 Aug 2010 08:20:45 +1200 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> Message-ID: I havent tried changing the ringing to something else, but I have had success with this: With no pre-answer. I found as soon as it pre-answered, in my situation, that the ringing tone stopped. On Tue, Aug 31, 2010 at 7:35 AM, Jock McKechnie wrote: > And I agree that it must work, somewhere. But it does not. > Work with me here: Firstly, I tcpdumped on the FreeSWITCH machine. If I'm > not seeing the RTP on the FS machine, it's not going to be anything like a > firewall or a NAT issue. Secondly, and far more telling (as the above could > be related to me screwing up the tcpdump somehow): I get the Playback audio, > so it is possible to get audio back down the line eventually and the RTP > does reach the caller, so RTP is able to flow from FS to caller without > interference (as I said in the initial mail, the only RTP sent from FS > starts _after_ the 200 OK). It simply is that FS is not sending out early > media. > Like I said, either I'm doing something remarkably stupid, or there is > something extremely strange going on here. > Should I try to use ring_ready AND pre_answer in combination? I have not > attempted that yet. If so, in what order? > ?- JP From bobc at devassert.com Mon Aug 30 13:20:45 2010 From: bobc at devassert.com (Bob Coleman) Date: Tue, 31 Aug 2010 08:20:45 +1200 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> Message-ID: I havent tried changing the ringing to something else, but I have had success with this: With no pre-answer. I found as soon as it pre-answered, in my situation, that the ringing tone stopped. On Tue, Aug 31, 2010 at 7:35 AM, Jock McKechnie wrote: > And I agree that it must work, somewhere. But it does not. > Work with me here: Firstly, I tcpdumped on the FreeSWITCH machine. If I'm > not seeing the RTP on the FS machine, it's not going to be anything like a > firewall or a NAT issue. Secondly, and far more telling (as the above could > be related to me screwing up the tcpdump somehow): I get the Playback audio, > so it is possible to get audio back down the line eventually and the RTP > does reach the caller, so RTP is able to flow from FS to caller without > interference (as I said in the initial mail, the only RTP sent from FS > starts _after_ the 200 OK). It simply is that FS is not sending out early > media. > Like I said, either I'm doing something remarkably stupid, or there is > something extremely strange going on here. > Should I try to use ring_ready AND pre_answer in combination? I have not > attempted that yet. If so, in what order? > ?- JP From jock.mckechnie at gmail.com Mon Aug 30 13:45:40 2010 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Mon, 30 Aug 2010 15:45:40 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> Message-ID: Thank you for the suggestion, Bob. I slimmed down my config to this, plus the sleep/playback/hangup. I confirm seeing the 180 in the signalling, no RTP, it waits a bit, then the 200 OK and RTP floods through. I _know_ this should simply work, but... it doesn't. Anyone else got any thoughts? Cheers again; - Jock On Mon, Aug 30, 2010 at 3:20 PM, Bob Coleman wrote: > I havent tried changing the ringing to something else, but I have had > success with this: > > > > > With no pre-answer. I found as soon as it pre-answered, in my > situation, that the ringing tone stopped. > > On Tue, Aug 31, 2010 at 7:35 AM, Jock McKechnie > wrote: > > And I agree that it must work, somewhere. But it does not. > > Work with me here: Firstly, I tcpdumped on the FreeSWITCH machine. If I'm > > not seeing the RTP on the FS machine, it's not going to be anything like > a > > firewall or a NAT issue. Secondly, and far more telling (as the above > could > > be related to me screwing up the tcpdump somehow): I get the Playback > audio, > > so it is possible to get audio back down the line eventually and the RTP > > does reach the caller, so RTP is able to flow from FS to caller without > > interference (as I said in the initial mail, the only RTP sent from FS > > starts _after_ the 200 OK). It simply is that FS is not sending out early > > media. > > Like I said, either I'm doing something remarkably stupid, or there is > > something extremely strange going on here. > > Should I try to use ring_ready AND pre_answer in combination? I have not > > attempted that yet. If so, in what order? > > - JP > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/75d4c303/attachment.html From brian at freeswitch.org Mon Aug 30 13:51:54 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Aug 2010 15:51:54 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> Message-ID: 180 from us won't include media... its just a ringing indication. /b On Aug 30, 2010, at 3:45 PM, Jock McKechnie wrote: > Thank you for the suggestion, Bob. > > I slimmed down my config to this, plus the sleep/playback/hangup. I confirm seeing the 180 in the signalling, no RTP, it waits a bit, then the 200 OK and RTP floods through. > > I _know_ this should simply work, but... it doesn't. Anyone else got any thoughts? > > Cheers again; > > - Jock > From jock.mckechnie at gmail.com Mon Aug 30 14:13:43 2010 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Mon, 30 Aug 2010 16:13:43 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> Message-ID: On Mon, Aug 30, 2010 at 3:51 PM, Brian West wrote: > 180 from us won't include media... its just a ringing indication. > That's what I thought, I appreciate the confirmation on that. I've reset it back to: And using: ngrep -q '' 'port !ssh' Can confirm there's no RTP until the 200 OK. Demented. - JP On Aug 30, 2010, at 3:45 PM, Jock McKechnie wrote: > Thank you for the suggestion, Bob. > > I slimmed down my config to this, plus the sleep/playback/hangup. I confirm seeing the 180 in the signalling, no RTP, it waits a bit, then the 200 OK and RTP floods through. > > I _know_ this should simply work, but... it doesn't. Anyone else got any thoughts? > > Cheers again; > > - Jock > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/2bf8e67e/attachment-0001.html From anthony.minessale at gmail.com Mon Aug 30 14:15:14 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Aug 2010 16:15:14 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: Message-ID: ringback has nothing to do with playback, it's only relevant to bridge. trying to call playback before you answer will implicitly create a pre_answer Do this in your dialplan: playback file1 sleep 10000 answer playback file2 if you expect the playback to loop endlessly it would take more steps, the easiest of which would be to make a 10 second audio file and use: playback file1 answer playback file2 On Mon, Aug 30, 2010 at 1:49 PM, Jock McKechnie wrote: > Good arvo, all; > I'm battling with a problem in making Early Media play and getting nowhere. > I've searched the wiki, forums and the web in general without anything that > seems to help. This suggests to me that I'm either doing something > stupendously stupid, or there's a very, very odd thing going on here. > I've been tasked with something that should be fairly simple: > Accept an inbound call > Play a specific audio file as early media > Pickup the call after a certain time interval > Play another audio file into the off-hook channel > Disconnect the call. > Sounds simple. My extension looks like so: > ?? > ?? ? ? > ?? ? ? ? ? > ?? ? ? ? ? data="ringback=/usr/local/freeswitch/sounds/early-media.wav"/> > > ?? ? ? ? ? > ?? ? ? ? ? > ?? ? ? ? ? data="/usr/local/freeswitch/sounds/playback"/> > ?? ? ? ? ? > ?? ? ? > ?? > (Replace the 212 number with a real inbound DID which is being routed from > Verizon (via SIP) to this FreeSWITCH machine) > The call is received, a 183 is sent by FreeSWITCH (pre_answer), it waits the > 10 seconds, picks up the call and plays the playback.PCMU file... but > there's no early media. I've tcpdumped on the FS machine and verified it > definitely is /not/ sending out RTP until the call 200 OKs and the Playback > begins. > As you can see, I've also attempted to replace the ringback= to a cadence > instead of a WAV file, just to ensure it isn't unhappy with the WAV format, > with the same results. > Boy, I'd sure love it if someone could point out what potentially brain > damaged thing I'm doing (assuming it's nothing arcane). > My thanks to all; > ?- Jock > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From brian at freeswitch.org Mon Aug 30 14:23:04 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Aug 2010 16:23:04 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> Message-ID: <964D421A-B03D-4FCB-90BE-0F49D573050F@freeswitch.org> Well nothing is going to be sent because it has NOTHING to send... what exactly do you wish it to do? Do you want to send your own ringback inband? Or have the far end do it? So in your previous example the 180 was sent but no RTP because thats fine. You can also do inband early media but you'll have to generate it via the various methods we have. The defaults have examples of this. /b On Aug 30, 2010, at 4:13 PM, Jock McKechnie wrote: > Can confirm there's no RTP until the 200 OK. > > Demented. > > - JP From jock.mckechnie at gmail.com Mon Aug 30 14:59:47 2010 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Mon, 30 Aug 2010 16:59:47 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: <964D421A-B03D-4FCB-90BE-0F49D573050F@freeswitch.org> References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> <964D421A-B03D-4FCB-90BE-0F49D573050F@freeswitch.org> Message-ID: On Mon, Aug 30, 2010 at 4:23 PM, Brian West wrote: > Well nothing is going to be sent because it has NOTHING to send... what > exactly do you wish it to do? Do you want to send your own ringback inband? > Or have the far end do it? So in your previous example the 180 was sent > but no RTP because thats fine. You can also do inband early media but > you'll have to generate it via the various methods we have. The defaults > have examples of this. As in the initial eMail, I wish to play a specific audio file as early media, pick up the call, play another audio file post answer, and then disconnect. I had not tried the example given in the defaults, it's been a while since I'd looked through them. I ripped an example straight out of the default sample to try and get a working baseline I could modify, if I could get ANY early media it would be start. So far I have this: Quite a different set up, now I'm bridging the call and then playing back via this bridged loopback extension, but still no RTP on the 183. Is this closer to what you think should work out of the box? - JP > On Aug 30, 2010, at 4:13 PM, Jock McKechnie wrote: > > > Can confirm there's no RTP until the 200 OK. > > > > Demented. > > > > - JP > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/47730496/attachment.html From anthony.minessale at gmail.com Mon Aug 30 15:48:15 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Aug 2010 17:48:15 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> <964D421A-B03D-4FCB-90BE-0F49D573050F@freeswitch.org> Message-ID: did you try the ones I listed? I did not phrase them in xml because it's redundant but you should simply be able to make a simple extension that calls playback WITHOUT calling answer and the file will play in early media? Maybe it's too easy? This will play a file in early media, answer then play it again as in-call media. On Mon, Aug 30, 2010 at 4:59 PM, Jock McKechnie wrote: > > > On Mon, Aug 30, 2010 at 4:23 PM, Brian West wrote: >> >> Well nothing is going to be sent because it has NOTHING to send... what >> exactly do you wish it to do? ?Do you want to send your own ringback inband? >> ?Or have the far end do it? ?So in your previous example the 180 was sent >> but no RTP because thats fine. ?You can also do inband early media but >> you'll have to generate it via the various methods we have. ?The defaults >> have examples of this. > > As in the initial eMail, I wish to play a specific audio file as early > media, pick up the call, play another audio file post answer, and then > disconnect. I had not tried the example given in the defaults, it's been a > while since I'd looked through them. I ripped an example straight out of the > default sample to try and get a working baseline I could modify, if I could > get ANY early media it would be start. So far I have this: > ?? ? > ?? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? data="/usr/local/freeswitch/sounds/playback"/> > ?? ? ? ? > ?? ? ? > ?? ? > ?? ? > ?? > ?? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? > ?? ? > Quite a different set up, now I'm bridging the call and then playing back > via this bridged loopback extension, but still no RTP on the 183. > Is this closer to what you think should work out of the box? > ?- JP > > > >> >> On Aug 30, 2010, at 4:13 PM, Jock McKechnie wrote: >> >> > Can confirm there's no RTP until the 200 OK. >> > >> > Demented. >> > >> > ?- JP >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From ktngl at yahoo.co.uk Mon Aug 30 16:34:12 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Mon, 30 Aug 2010 23:34:12 +0000 (GMT) Subject: [Freeswitch-users] recvEvent in ruby Message-ID: <551602.81620.qm@web29204.mail.ird.yahoo.com> I have this code below in which I want to receive and extract dtmf events. It seems all previous events are queued and received together at this point as the while loop is run. I don't know why it does this I only need the current events to be received. Another issue happening at this point when I press any digit 0-9 on the phone it is not getting an incoming dtmf events while @con.connected ??? #debug??? ??? puts "start loop" ??? ??? #no digits ??? if dtmfbuffer.length == 0 ??? ??? e = @con.recvEventTimed(response_timeout) ??? # there are digits ??? elsif dtmfbuffer.length > 0 ??? ??? e = @con.recvEventTimed(interdigit_timeout) ??? end ??? if e ??? ??? name = e.getHeader("Event-Name") ??? ??? puts "#{name}" ??? ??? break if name == "SERVER_DISCONNECTED" ??? ??? if name == "DTMF" ??? ??? ??? digit = e.getHeader("DTMF-Digit") ??? ??? ??? duration = e.getHeader("DTMF-Duration")??? ??? ??? ??? ??? dtmfbuffer << digit ??? ??? ??? puts "*** Dtmf #{digit} dur:#{duration} from #{@cli}" ??? ??? end ??? end ??? if dtmfbuffer.length > 9 ??? ??? break ??? end ??? ??? #debug ??? ??? puts "end loop" end -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/1037b863/attachment.html From msc at freeswitch.org Mon Aug 30 17:13:36 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Aug 2010 17:13:36 -0700 Subject: [Freeswitch-users] recvEvent in ruby In-Reply-To: <551602.81620.qm@web29204.mail.ird.yahoo.com> References: <551602.81620.qm@web29204.mail.ird.yahoo.com> Message-ID: What integer values are in response_timeout and interdigit_timeout? Just curious, because if they are too large then it will indeed block while waiting for events. -MC On Mon, Aug 30, 2010 at 4:34 PM, Nigel Kent wrote: > I have this code below in which I want to receive and extract dtmf events. > It seems all previous events are queued and received together at this point > as the while loop is run. I don't know why it does this I only need the > current events to be received. > > Another issue happening at this point when I press any digit 0-9 on the > phone it is not getting an incoming dtmf events > > > > while @con.connected > #debug > puts "start loop" > > #no digits > if dtmfbuffer.length == 0 > e = @con.recvEventTimed(response_timeout) > # there are digits > elsif dtmfbuffer.length > 0 > e = @con.recvEventTimed(interdigit_timeout) > end > > > if e > name = e.getHeader("Event-Name") > puts "#{name}" > break if name == "SERVER_DISCONNECTED" > if name == "DTMF" > digit = e.getHeader("DTMF-Digit") > duration = e.getHeader("DTMF-Duration") > dtmfbuffer << digit > puts "*** Dtmf #{digit} dur:#{duration} from #{@cli}" > end > end > > if dtmfbuffer.length > 9 > break > end > #debug > puts "end loop" > end > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/a3cc1bd6/attachment-0001.html From ktngl at yahoo.co.uk Mon Aug 30 17:24:56 2010 From: ktngl at yahoo.co.uk (Nigel Kent) Date: Tue, 31 Aug 2010 00:24:56 +0000 (GMT) Subject: [Freeswitch-users] recvEvent in ruby In-Reply-To: Message-ID: <43673.20678.qm@web29214.mail.ird.yahoo.com> # time out and inter digit time out response_timeout = (20 * 1000) interdigit_timeout = (10 * 1000) --- On Tue, 31/8/10, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] recvEvent in ruby To: "FreeSWITCH Users Help" Date: Tuesday, 31 August, 2010, 0:13 What integer values are in response_timeout and interdigit_timeout? Just curious, because if they are too large then it will indeed block while waiting for events. -MC On Mon, Aug 30, 2010 at 4:34 PM, Nigel Kent wrote: I have this code below in which I want to receive and extract dtmf events. It seems all previous events are queued and received together at this point as the while loop is run. I don't know why it does this I only need the current events to be received. Another issue happening at this point when I press any digit 0-9 on the phone it is not getting an incoming dtmf events while @con.connected ??? #debug??? ??? puts "start loop" ??? ??? #no digits ??? if dtmfbuffer.length == 0 ??? ??? e = @con.recvEventTimed(response_timeout) ??? # there are digits ??? elsif dtmfbuffer.length > 0 ??? ??? e = @con.recvEventTimed(interdigit_timeout) ??? end ??? if e ??? ??? name = e.getHeader("Event-Name") ??? ??? puts "#{name}" ??? ??? break if name == "SERVER_DISCONNECTED" ??? ??? if name == "DTMF" ??? ??? ??? digit = e.getHeader("DTMF-Digit") ??? ??? ??? duration = e.getHeader("DTMF-Duration")??? ??? ??? ??? ??? dtmfbuffer << digit ??? ??? ??? puts "*** Dtmf #{digit} dur:#{duration} from #{@cli}" ??? ??? end ??? end ??? if dtmfbuffer.length > 9 ??? ??? break ??? end ??? ??? #debug ??? ??? puts "end loop" end _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/553c830d/attachment.html From msc at freeswitch.org Mon Aug 30 17:47:55 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Aug 2010 17:47:55 -0700 Subject: [Freeswitch-users] recvEvent in ruby In-Reply-To: <43673.20678.qm@web29214.mail.ird.yahoo.com> References: <43673.20678.qm@web29214.mail.ird.yahoo.com> Message-ID: On Mon, Aug 30, 2010 at 5:24 PM, Nigel Kent wrote: > > # time out and inter digit time out > response_timeout = (20 * 1000) > interdigit_timeout = (10 * 1000) > According to page 224 of the handy dandy bridge book :) the recvEventTimed method accepts milliseconds as its argument. So... it looks like you are blocking for 20 seconds and 10 seconds on these two values. I'm assuming that you don't want to block for that long? -MC > > > --- On *Tue, 31/8/10, Michael Collins * wrote: > > > From: Michael Collins > Subject: Re: [Freeswitch-users] recvEvent in ruby > To: "FreeSWITCH Users Help" > Date: Tuesday, 31 August, 2010, 0:13 > > > What integer values are in response_timeout and interdigit_timeout? Just > curious, because if they are too large then it will indeed block while > waiting for events. > -MC > > On Mon, Aug 30, 2010 at 4:34 PM, Nigel Kent > > wrote: > > I have this code below in which I want to receive and extract dtmf events. > It seems all previous events are queued and received together at this point > as the while loop is run. I don't know why it does this I only need the > current events to be received. > > Another issue happening at this point when I press any digit 0-9 on the > phone it is not getting an incoming dtmf events > > > > while @con.connected > #debug > puts "start loop" > > #no digits > if dtmfbuffer.length == 0 > e = @con.recvEventTimed(response_timeout) > # there are digits > elsif dtmfbuffer.length > 0 > e = @con.recvEventTimed(interdigit_timeout) > end > > > if e > name = e.getHeader("Event-Name") > puts "#{name}" > break if name == "SERVER_DISCONNECTED" > if name == "DTMF" > digit = e.getHeader("DTMF-Digit") > duration = e.getHeader("DTMF-Duration") > dtmfbuffer << digit > puts "*** Dtmf #{digit} dur:#{duration} from #{@cli}" > end > end > > if dtmfbuffer.length > 9 > break > end > #debug > puts "end loop" > end > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/2ad5f3e3/attachment.html From msc at freeswitch.org Mon Aug 30 17:50:33 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Aug 2010 17:50:33 -0700 Subject: [Freeswitch-users] WANTED: Cookbook recipes Message-ID: Hello all! We are investigating the possibility of writing a cookbook for FreeSWITCH. We would like to invite the community at large to submit ideas for recipes. I have started a wiki page here: http://wiki.freeswitch.org/wiki/Cookbook Please add your cookbook wishlist items here. If you have an actual recipe that you'd like to contribute then mention that fact as well. We are not looking for the actual recipes themselves but rather the ideas for the recipes. I've added a number of sections on the wiki but feel free to add another one if I have missed a category of recipes. Any questions please let me know! Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/a2c102c2/attachment.html From adolfo at delorenzo.mobi Mon Aug 30 17:59:24 2010 From: adolfo at delorenzo.mobi (Adolfo Delorenzo) Date: Mon, 30 Aug 2010 21:59:24 -0300 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: No I am not actually. It should work though. On 30 August 2010 13:25, Giovanni Maruzzelli wrote: > On Mon, Aug 30, 2010 at 12:57 PM, Adolfo Delorenzo > wrote: > > I installed the skype version that was available on the official website > and > > it works like a charm for me with skypopen. > > > > I use ubuntu server 10.04. > > Ciao Adolfo, > > are you using multiple concurrent skype clients in your setup? > > And if so, they're using the same skypename? > > And if so, they're correctly answering multiple concurrent inbound > calls to the same skypeuser, and are correctly making multiple > concurrent outbound calls? > > Sorry for the long list or questions, but skype 2.1.81 beta is known > not to work on centos (not just with skypopen), and I saw that it > works well on Ubuntu 10.04, but was giving problems to me using > multiple concurrent instances of the same skypeuser (and I cannot > retest it for a couple of weeks). > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100830/0264868c/attachment-0001.html From dujinfang at gmail.com Mon Aug 30 19:08:58 2010 From: dujinfang at gmail.com (Seven Du) Date: Tue, 31 Aug 2010 10:08:58 +0800 Subject: [Freeswitch-users] windows build error In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05818D@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05818D@cooper> Message-ID: Thanks, I re-Cloned with auto CRLF off and it worked. On Mon, Aug 30, 2010 at 10:13 PM, Peter Olsson wrote: > Make sure to disable all kinds of CRLF/LF conversions. By default git on Windows seems to enable this. There is a setting called autocrlf, make sure it's set to false. > > /Peter > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Seven Du [dujinfang at gmail.com] > Skickat: den 30 augusti 2010 15:52 > Till: freeswitch-users > ?mne: [Freeswitch-users] windows build error > > Hi, > > I'm new to windows build. I cloned code a few days ago and pulled to > head. Can could not build sofia. > > c:\workspace\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(119) : > fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': > No such file or directory > > > I tried to rebuild but the result was the same. > > I can start freeswitch though mod_sofia and mod file format (not > exactly remember) cannot load. > > Any hint? > > I will re-clone code and try again tomorrow. > > Thanks. > > -- > Blog: http://www.dujinfang.com > Proj: ?http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4c7bb97c32934187118539! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From moises.silva at gmail.com Mon Aug 30 21:02:42 2010 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 31 Aug 2010 00:02:42 -0400 Subject: [Freeswitch-users] freetdm config problem In-Reply-To: <31184.1283182918@ccs.covici.com> References: <20381.1283145116@ccs.covici.com> <31184.1283182918@ccs.covici.com> Message-ID: On Mon, Aug 30, 2010 at 11:41 AM, wrote: > Moises Silva wrote: > > > On Mon, Aug 30, 2010 at 1:11 AM, wrote: > > > > > Hi. I was trying to config freetdm, but it complained that the name > was > > > not provided and would not load the module. > > > > > > > > Go to pastebin.freeswitch.org and paste the relevant parts of the error > > logged, then paste a link here. > > Sorry if this is a dupe, message seems not to have gotten through. > > Link is: http://pastebin.freeswitch.org/13752 > I missed the obvious in your config. The span type you specified is invalid. [span zt myFxs] and or [span zt myFxo], there is a space between zt and the span name, zt is the special keyword to specify the span type, in this case, a zaptel/dahdi span. The string after the space is whatever span name you want to give to the span. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/b5e550c1/attachment.html From abubacker at bksystems.co.in Mon Aug 30 21:12:36 2010 From: abubacker at bksystems.co.in (abubacker) Date: Tue, 31 Aug 2010 09:42:36 +0530 Subject: [Freeswitch-users] Mod Fifo In-Reply-To: References: Message-ID: <4C7C8134.6040405@bksys.co.in> On Monday 30 August 2010 08:43 PM, Jonathan Pitcher wrote: > Hi all, > > I am attempting to setup a dynamic fifo group. I actually have the > group working and can have agents log into the que and receive calls, > but what I have having trouble with is Caller ID on the call to the > agent from the que. > > I have been looking at: http://wiki.freeswitch.org/wiki/Mod_fifo > > And it says that to change the caller ID to the agent you can change > it in 2 places, one in the script that logs the agent in and secondly > in the call that gets sent to fifo in. I have choosen the second > option because it allows customers to change the name of the que and > have it immediately change instead of having to force users to log in > and log out again to see the name change take place. > > When the caller dialing into the que calls the que, this is what I > return to FS. > > > >
> > > > data="domain_name=1000076.pprd.nuvio.net" /> > > > > > > > > > >
> > > This is what fifo list shows : > > > caller_count="0" waiting_count="0" importance="0" > outbound_per_cycle="0" outbound_priority="5" > outbound_strategy="ringall"> > > call-count="3" outbound-fail-count="0" taking-calls="1" status="" > outbound-call-total-count="3" outbound-fail-total-count="2" logged-on- > since="2010-08-30 08:02:50" manual-calls-out-count="0" manual-calls-in- > count="1" manual-calls-out-total-count="0" manual-calls-in-total- > count="1" ring-count="0" start-time="2010-08-30 10:08:42" stop- > time="2010-08-30 10:08:42" next-available="2010-08-30 > 10:08:48">{fd=1000076.pprd.nuvio.net,}user/638 at 1000076.pprd.nuvio.net member> > > > > > > caller_count="0" waiting_count="0" importance="0" > outbound_per_cycle="1" outbound_priority="5" > outbound_strategy="ringall"> > > > > > > waiting_count="0" importance="0" outbound_per_cycle="0" > outbound_priority="5" outbound_strategy="ringall"> > > call-count="3" outbound-fail-count="0" taking-calls="1" status="" > outbound-call-total-count="3" outbound-fail-total-count="2" logged-on- > since="2010-08-30 08:02:50" manual-calls-out-count="0" manual-calls-in- > count="1" manual-calls-out-total-count="0" manual-calls-in-total- > count="1" next-available="2010-08-30 > 10:08:48">{fd=1000076.pprd.nuvio.net,}user/638 at 1000076.pprd.nuvio.net member> > > > > > > > > > Now everything works like it should but I can't seem to get the > origination_caller_id_name variable to be used in the incoming dialplan. > > What am I missing ? > > Thanks in advance, > > Jonathan Pitcher > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Please try this $ fifo list_verbose -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer: http://www.bksystems.co.in/email-policy From chat2jesse at gmail.com Mon Aug 30 21:34:51 2010 From: chat2jesse at gmail.com (jesse) Date: Mon, 30 Aug 2010 21:34:51 -0700 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: References: Message-ID: what is the point use to gMail? The backend of Gmail dialing is actually Google Voice. FS can integrate with Google Voice via some java library to make outgoing calls for U.S and Canada.. jesse On Sun, Aug 29, 2010 at 12:22 PM, Malay Thakershi wrote: > Hello, > As you know, Google introduced new "Call US/Canada" free from gMail. > Is it possible to make outgoing calls from FreeSwitch to USA number using > gTalk/gMail? > Thank you for help/guidance. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From farhan.husain at csebuet.org Mon Aug 30 22:47:15 2010 From: farhan.husain at csebuet.org (Farhan Husain) Date: Tue, 31 Aug 2010 00:47:15 -0500 Subject: [Freeswitch-users] How to get switch_core_session for a call Message-ID: Hello, Is there a way to get the switch_core_session of a call? Is it possible by subscribing to any channel or any other event? Thanks, Farhan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/99542d80/attachment.html From mthakershi at gmail.com Mon Aug 30 23:58:30 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 31 Aug 2010 01:58:30 -0500 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: References: Message-ID: That is what I want to know. How can I integrate Google Voice (or whatever they use to provide free US/Canada calling) to make FS outbound calls? What is that library? On Mon, Aug 30, 2010 at 11:34 PM, jesse wrote: > what is the point use to gMail? The backend of Gmail dialing is > actually Google Voice. > > FS can integrate with Google Voice via some java library to make > outgoing calls for U.S and Canada.. > > jesse > > On Sun, Aug 29, 2010 at 12:22 PM, Malay Thakershi > wrote: > > Hello, > > As you know, Google introduced new "Call US/Canada" free from gMail. > > Is it possible to make outgoing calls from FreeSwitch to USA number using > > gTalk/gMail? > > Thank you for help/guidance. > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/e68bd44f/attachment.html From gmaruzz at celliax.org Tue Aug 31 02:36:00 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 31 Aug 2010 11:36:00 +0200 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: On Tue, Aug 31, 2010 at 2:59 AM, Adolfo Delorenzo wrote: > No I am not actually. It should work though. Adolfo, I had tested extensively skype 2.1.81 on Ubuntu 10.04 (and other platforms) and I am reasonably sure that multiple concurrent skype clients with same skypeusername do not work correctly with skypopen (particularly in answering more than one concurrent call) because the 2.1.81 behave in a different way than 2.0.72 at the protocol level (when more than one instance of same skypename). -giovanni > > > On 30 August 2010 13:25, Giovanni Maruzzelli wrote: >> >> On Mon, Aug 30, 2010 at 12:57 PM, Adolfo Delorenzo >> wrote: >> > I installed the skype version that was available on the official website >> > and >> > it works like a charm for me with skypopen. >> > >> > I use ubuntu server 10.04. >> >> Ciao Adolfo, >> >> are you using multiple concurrent skype clients in your setup? >> >> And if so, they're using the same skypename? >> >> And if so, they're correctly answering multiple concurrent inbound >> calls to the same skypeuser, and are correctly making multiple >> concurrent outbound calls? >> >> Sorry for the long list or questions, but skype 2.1.81 beta is known >> not to work on centos (not just with skypopen), and I saw that it >> works well on Ubuntu 10.04, but was giving problems to me using >> multiple concurrent instances of the same skypeuser (and I cannot >> retest it for a couple of weeks). >> >> -giovanni >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From covici at ccs.covici.com Tue Aug 31 02:57:25 2010 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 31 Aug 2010 05:57:25 -0400 Subject: [Freeswitch-users] freetdm config problem In-Reply-To: References: <20381.1283145116@ccs.covici.com> <31184.1283182918@ccs.covici.com> Message-ID: <28245.1283248645@ccs.covici.com> Moises Silva wrote: > On Mon, Aug 30, 2010 at 11:41 AM, wrote: > > > Moises Silva wrote: > > > > > On Mon, Aug 30, 2010 at 1:11 AM, wrote: > > > > > > > Hi. I was trying to config freetdm, but it complained that the name > > was > > > > not provided and would not load the module. > > > > > > > > > > > Go to pastebin.freeswitch.org and paste the relevant parts of the error > > > logged, then paste a link here. > > > > Sorry if this is a dupe, message seems not to have gotten through. > > > > Link is: http://pastebin.freeswitch.org/13752 > > > > I missed the obvious in your config. The span type you specified is invalid. > > [span zt myFxs] and or [span zt myFxo], there is a space between zt and the > span name, zt is the special keyword to specify the span type, in this case, > a zaptel/dahdi span. The string after the space is whatever span name you > want to give to the span. What is the name => item mean then? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From engineerzuhairraza at gmail.com Tue Aug 31 04:12:29 2010 From: engineerzuhairraza at gmail.com (Zuhair Raza) Date: Tue, 31 Aug 2010 16:12:29 +0500 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: hi guyz i have found skype 2.0.072 http://linux.softpedia.com/get/Communications/Chat/Skype-Portable-49021.shtml but it is useless for me as i dont use ubuntu, if any one can find this for centos. then please give it to me. thanks On Tue, Aug 31, 2010 at 2:36 PM, Giovanni Maruzzelli wrote: > On Tue, Aug 31, 2010 at 2:59 AM, Adolfo Delorenzo > wrote: > > No I am not actually. It should work though. > > Adolfo, > > I had tested extensively skype 2.1.81 on Ubuntu 10.04 (and other > platforms) and I am reasonably sure that multiple concurrent skype > clients with same skypeusername do not work correctly with skypopen > (particularly in answering more than one concurrent call) because the > 2.1.81 behave in a different way than 2.0.72 at the protocol level > (when more than one instance of same skypename). > > -giovanni > > > > > > > On 30 August 2010 13:25, Giovanni Maruzzelli > wrote: > >> > >> On Mon, Aug 30, 2010 at 12:57 PM, Adolfo Delorenzo > >> wrote: > >> > I installed the skype version that was available on the official > website > >> > and > >> > it works like a charm for me with skypopen. > >> > > >> > I use ubuntu server 10.04. > >> > >> Ciao Adolfo, > >> > >> are you using multiple concurrent skype clients in your setup? > >> > >> And if so, they're using the same skypename? > >> > >> And if so, they're correctly answering multiple concurrent inbound > >> calls to the same skypeuser, and are correctly making multiple > >> concurrent outbound calls? > >> > >> Sorry for the long list or questions, but skype 2.1.81 beta is known > >> not to work on centos (not just with skypopen), and I saw that it > >> works well on Ubuntu 10.04, but was giving problems to me using > >> multiple concurrent instances of the same skypeuser (and I cannot > >> retest it for a couple of weeks). > >> > >> -giovanni > >> > >> > >> -- > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/80ce46bb/attachment.html From babak.freeswitch at gmail.com Tue Aug 31 04:13:52 2010 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Tue, 31 Aug 2010 15:43:52 +0430 Subject: [Freeswitch-users] bridge Message-ID: Hi is it possible (in mod_managed or something similar) to create a new session and originate a call and if it was successful bridge it as the b leg of a call that moh is playing for it already(a waiting call)? thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/cf8addee/attachment.html From jim at k4gvo.com Tue Aug 31 04:21:48 2010 From: jim at k4gvo.com (Jim) Date: Tue, 31 Aug 2010 07:21:48 -0400 Subject: [Freeswitch-users] freetdm config problem In-Reply-To: <28245.1283248645@ccs.covici.com> References: <20381.1283145116@ccs.covici.com> <31184.1283182918@ccs.covici.com> <28245.1283248645@ccs.covici.com> Message-ID: <4C7CE5CC.7020907@k4gvo.com> On 08/31/2010 05:57 AM, covici at ccs.covici.com wrote: > Moises Silva wrote: > >> >> I missed the obvious in your config. The span type you specified is invalid. >> >> [span zt myFxs] and or [span zt myFxo], there is a space between zt and the >> span name, zt is the special keyword to specify the span type, in this case, >> a zaptel/dahdi span. The string after the space is whatever span name you >> want to give to the span. >> > What is the name => item mean then? > > In freetdm.conf if you have [span zt myFxs] Then in autoconfig/freetdm.conf.xml you need a section that (in part using your example) looks like: ... ... The myFxs provides the linkage to the section of the xml file from the freetdm.conf file. I hope that was the question, if not try again. Jim. From covici at ccs.covici.com Tue Aug 31 04:45:25 2010 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 31 Aug 2010 07:45:25 -0400 Subject: [Freeswitch-users] freetdm config problem In-Reply-To: References: <20381.1283145116@ccs.covici.com> <31184.1283182918@ccs.covici.com> Message-ID: <29817.1283255125@ccs.covici.com> Moises Silva wrote: > On Mon, Aug 30, 2010 at 11:41 AM, wrote: > > > Moises Silva wrote: > > > > > On Mon, Aug 30, 2010 at 1:11 AM, wrote: > > > > > > > Hi. I was trying to config freetdm, but it complained that the name > > was > > > > not provided and would not load the module. > > > > > > > > > > > Go to pastebin.freeswitch.org and paste the relevant parts of the error > > > logged, then paste a link here. > > > > Sorry if this is a dupe, message seems not to have gotten through. > > > > Link is: http://pastebin.freeswitch.org/13752 > > > > I missed the obvious in your config. The span type you specified is invalid. > > [span zt myFxs] and or [span zt myFxo], there is a space between zt and the > span name, zt is the special keyword to specify the span type, in this case, > a zaptel/dahdi span. The string after the space is whatever span name you > want to give to the span. OK, that makes it work -- still confused about the name => parameter. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From rupa at rupa.com Tue Aug 31 06:03:24 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 31 Aug 2010 08:03:24 -0500 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: References: Message-ID: The issue is initial call setup for PSTN access. The control for that is not exposed via gtalk, only through the gmail client. On Mon, Aug 30, 2010 at 11:34 PM, jesse wrote: > what is the point use to gMail? The backend of Gmail dialing is > actually Google Voice. > > FS can integrate with Google Voice via some java library to make > outgoing calls for U.S and Canada.. > > jesse > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/0a74433d/attachment.html From jock.mckechnie at gmail.com Tue Aug 31 06:58:33 2010 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Tue, 31 Aug 2010 08:58:33 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> <964D421A-B03D-4FCB-90BE-0F49D573050F@freeswitch.org> Message-ID: On Mon, Aug 30, 2010 at 5:48 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > did you try the ones I listed? > > I did not phrase them in xml because it's redundant but you should > simply be able to make a simple extension > that calls playback WITHOUT calling answer and the file will play in > early media? > > Maybe it's too easy? > > This will play a file in early media, answer then play it again as > in-call media. > > > > > > > > > > I did not, I misunderstood your initial eMail, clearly. I have rigged up the above and something quite odd happens - definitely much closer, but still odd. The CLI shows FreeSWITCH accept the call, there is almost a full ten second pause, and then I hear a 'click' on the line, there is another pause, and then I get the latter Playback (I've used two different audio files to ensure I can tell the difference). However, I've tcpdumped and according to Wireshark: INVITE 100 Trying (10.1 second pause) Early Media RTP 200 OK Latter Playback RTP BYE A review of the audio through WireShark verifies that FS is playing the first WAV file as, apparently, early media. So I appear to have two problems: Firstly, there's a ten second pause. I did not reinstate my sleep statement, so I don't understand why it's waiting so long after getting the call. And although FreeSWITCH is now definitely pumping out the early media, Verizon is, apparently, not handing it back. Is there any possibility something slightly non-standard is happening here, so VZ's equipment (cough, Sonus, cough) is unaware it should be passing RTP on to the calling party? I'm afraid my telephony knowledge is primarily VoIP related to Asterisk (wail, moan, etc, etc) and OpenSIPS. I was under the impression that a 183 Ringing was required in the flow to alert the far-end (calling party) that early media is coming, but I could, of course, be very wrong. Thank you very much Anthony; - Jock -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/9dc5a671/attachment-0001.html From adolfo at delorenzo.mobi Tue Aug 31 07:29:46 2010 From: adolfo at delorenzo.mobi (Adolfo Delorenzo) Date: Tue, 31 Aug 2010 11:29:46 -0300 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: Giovanni, There are some alternatives that I haven't tried yet but might work for you if you have the time to test them. 1) Instead of skypopen you can try sip2sis ( http://www.mhspot.com/sts/siptosis.html). I managed to use it with freeswitch once. 2) Download the version of skype you found on softpedia and convert the deb file to a rpm package with 'alien' Good luck! Adolfo On 31 August 2010 08:12, Zuhair Raza wrote: > hi guyz > > i have found skype 2.0.072 > > > http://linux.softpedia.com/get/Communications/Chat/Skype-Portable-49021.shtml > > but it is useless for me as i dont use ubuntu, if any one can find this for > centos. then please give it to me. > > thanks > > > On Tue, Aug 31, 2010 at 2:36 PM, Giovanni Maruzzelli wrote: > >> On Tue, Aug 31, 2010 at 2:59 AM, Adolfo Delorenzo >> wrote: >> > No I am not actually. It should work though. >> >> Adolfo, >> >> I had tested extensively skype 2.1.81 on Ubuntu 10.04 (and other >> platforms) and I am reasonably sure that multiple concurrent skype >> clients with same skypeusername do not work correctly with skypopen >> (particularly in answering more than one concurrent call) because the >> 2.1.81 behave in a different way than 2.0.72 at the protocol level >> (when more than one instance of same skypename). >> >> -giovanni >> >> > >> > >> > On 30 August 2010 13:25, Giovanni Maruzzelli >> wrote: >> >> >> >> On Mon, Aug 30, 2010 at 12:57 PM, Adolfo Delorenzo >> >> wrote: >> >> > I installed the skype version that was available on the official >> website >> >> > and >> >> > it works like a charm for me with skypopen. >> >> > >> >> > I use ubuntu server 10.04. >> >> >> >> Ciao Adolfo, >> >> >> >> are you using multiple concurrent skype clients in your setup? >> >> >> >> And if so, they're using the same skypename? >> >> >> >> And if so, they're correctly answering multiple concurrent inbound >> >> calls to the same skypeuser, and are correctly making multiple >> >> concurrent outbound calls? >> >> >> >> Sorry for the long list or questions, but skype 2.1.81 beta is known >> >> not to work on centos (not just with skypopen), and I saw that it >> >> works well on Ubuntu 10.04, but was giving problems to me using >> >> multiple concurrent instances of the same skypeuser (and I cannot >> >> retest it for a couple of weeks). >> >> >> >> -giovanni >> >> >> >> >> >> -- >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> Cell : +39-347-2665618 >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards, > Zuhair Raza > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/8ca1bee6/attachment.html From jason at cloudtree.net Tue Aug 31 07:32:19 2010 From: jason at cloudtree.net (Jason Jeffords) Date: Tue, 31 Aug 2010 10:32:19 -0400 Subject: [Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call? In-Reply-To: References: Message-ID: Hi Giovanni, Our tests are working using the latest git! Thank you so much for all of your help, Jason On Fri, Aug 27, 2010 at 1:56 PM, Giovanni Maruzzelli wrote: > Hi Jason, > > can you retest with latest git? > > Your problem was a bug in the management of the early media coming > from the leg B (eg: sofia, etc) to the A leg (eg skypopen). > > Now is fixed, I believe. > > Please let me know how it goes. > > -giovanni > > On Tue, Aug 24, 2010 at 1:34 AM, Jason Jeffords > wrote: > > Hi Giovanni, > > Our case 3 test was actually being bridged to a SIP endpoint (not in a > > conference, although > > this probably should not matter). > > We tested two type 3 cases: > > 1) Skype to FreeSWITCH Skype bridged to an extension of a registered SIP > > phone > > 2) Skype to FreeSWITCH Skype bridged to an outbound call through a PSTN > > gateway > > In both cases we are transiting FreeSWITCH, not terminating on it (could > > there be a > > race condition when doing signaling coordination to remote SIP endpoints, > > not FreeSWITCH > > itself?). > > When we terminate Skype calls on FreeSWITCH this works (case 2). It also > > works > > for outbound Skype calls (case1 - SIP to FreeSWITCH, FreeSWITCH Skype to > > Skype). > > Also, we are running the very latest (well, as of this morning) git > version, > > so that > > may introduce one more variable (if you are running an older version). > > Thanks for your help, > > Jason > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/afcf33f6/attachment.html From Nabble at slickdeals.endjunk.com Tue Aug 31 07:36:25 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Tue, 31 Aug 2010 07:36:25 -0700 (PDT) Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: References: Message-ID: <1283265385508-5482941.post@n2.nabble.com> jesse zhao wrote: > FS can integrate with Google Voice via some java library to make > outgoing calls for U.S and Canada.. > > jesse We certainly will appreciate if you can provide the link to show how to do this instead of having everyone re-invent the wheel. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/gTalk-new-phone-calls-feature-tp5475689p5482941.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gmaruzz at celliax.org Tue Aug 31 07:46:56 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 31 Aug 2010 16:46:56 +0200 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: On Tue, Aug 31, 2010 at 4:29 PM, Adolfo Delorenzo wrote: > 2) Download the version of skype you found on softpedia and convert the deb > file to a rpm package with 'alien' This will probably not work, because the package Zuhair found is in "runz" format, not in deb. -giovanni > > On 31 August 2010 08:12, Zuhair Raza wrote: >> >> hi guyz >> >> i have found skype 2.0.072 >> >> >> http://linux.softpedia.com/get/Communications/Chat/Skype-Portable-49021.shtml >> >> but it is useless for me as i dont use ubuntu, if any one can find this >> for centos. then please give it to me. >> >> thanks >> >> On Tue, Aug 31, 2010 at 2:36 PM, Giovanni Maruzzelli >> wrote: >>> >>> On Tue, Aug 31, 2010 at 2:59 AM, Adolfo Delorenzo >>> wrote: >>> > No I am not actually. It should work though. >>> >>> Adolfo, >>> >>> I had tested extensively skype 2.1.81 on Ubuntu 10.04 (and other >>> platforms) and I am reasonably sure that multiple concurrent skype >>> clients with same skypeusername do not work correctly with skypopen >>> (particularly in answering more than one concurrent call) because the >>> 2.1.81 behave in a different way than 2.0.72 at the protocol level >>> (when more than one instance of same skypename). >>> >>> -giovanni >>> >>> > >>> > >>> > On 30 August 2010 13:25, Giovanni Maruzzelli >>> > wrote: >>> >> >>> >> On Mon, Aug 30, 2010 at 12:57 PM, Adolfo Delorenzo >>> >> wrote: >>> >> > I installed the skype version that was available on the official >>> >> > website >>> >> > and >>> >> > it works like a charm for me with skypopen. >>> >> > >>> >> > I use ubuntu server 10.04. >>> >> >>> >> Ciao Adolfo, >>> >> >>> >> are you using multiple concurrent skype clients in your setup? >>> >> >>> >> And if so, they're using the same skypename? >>> >> >>> >> And if so, they're correctly answering multiple concurrent inbound >>> >> calls to the same skypeuser, and are correctly making multiple >>> >> concurrent outbound calls? >>> >> >>> >> Sorry for the long list or questions, but skype 2.1.81 beta is known >>> >> not to work on centos (not just with skypopen), and I saw that it >>> >> works well on Ubuntu 10.04, but was giving problems to me using >>> >> multiple concurrent instances of the same skypeuser (and I cannot >>> >> retest it for a couple of weeks). >>> >> >>> >> -giovanni >>> >> >>> >> >>> >> -- >>> >> Sincerely, >>> >> >>> >> Giovanni Maruzzelli >>> >> Cell : +39-347-2665618 >>> >> >>> >> _______________________________________________ >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Regards, >> Zuhair Raza >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From kond at nstel.ru Tue Aug 31 07:50:20 2010 From: kond at nstel.ru (Nikolay Kondratyev) Date: Tue, 31 Aug 2010 18:50:20 +0400 Subject: [Freeswitch-users] no ringback tone for transit h323 - sip call Message-ID: <20100831145020.8BA3B1308A@mail.nstel.ru> Hi all, I'm rather new to freeswitch, and i'm trying to setup FS (version 1.0.6) as sip-h323 gateway between sipX and avaya ip406. I have teh following test setup: sipX 172.23.9.2 beaver.sip.nstel.ru with extension 2853 (linksys spa 942 phone) Freeswitch 172.23.22.49 fs.lab.nstel.ru with extension 2854 (xlite) avaya ip406 with extension 5840 When i call 2853 -> 5840 i can hear ringback tone and voice is ok. When calling 5840 -> 2854, rbt and voice is ok. When calling 2854 -> 2853, rbt and voice is ok. When calling 5840 -> 2853, 2853 phone rings, there is no rbt, but voice is ok after answer. I have the following h323.conf file: [freeswitch at freeswitch autoload_configs]$ cat h323.conf.xml in the default.xml dialplan i have default "Local_Extension" (with the only addition of a single number 2854 to be matched). below Local_Extension i send 28xx numbers to sipx as following: > FS logs for 5840 -> 2854 (rbt and voice ok) and 5840 -> 2853 (no rbt, voice ok) are attached. In the log files i can see the moment when FS starts playing RBT into h323 channel in case 5840 -> 2854 call. And i see that FS does not do it in case of 5840 -> 2853 call. But i can not see why... Probably i misconfigured something... Can anybody please help with this problem? Thanks in advance, Nikolay. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/def08186/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: no-transit-rbt.zip Type: application/x-zip-compressed Size: 57376 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/def08186/attachment-0001.bin From shamun.toha at gmail.com Tue Aug 31 08:21:37 2010 From: shamun.toha at gmail.com (Shamun toha md) Date: Tue, 31 Aug 2010 17:21:37 +0200 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: I got once, but it was a virus for CentOS, lol :P Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/91a89406/attachment.html From esamuels777 at gmail.com Tue Aug 31 03:14:22 2010 From: esamuels777 at gmail.com (Errol Samuels) Date: Tue, 31 Aug 2010 11:14:22 +0100 Subject: [Freeswitch-users] Nextone (Genband) >> Freeswitch Ringback issue Message-ID: Hello All, I have a weird situation where a carrier is sending calls to our FS server but not hearing a Ringback tone, however once the call connects they are getting audio but when this carrier sends calls to our Yate server they are getting the Ringback without a problem. I have tried setting the Ring_Ready below but it didn't help the situation either. The traces from both the FS server and the Yate server show we are sending: 100 Trying 180 Ringing 183 Session Progress with SDP The carrier is claiming they are using the same configuration for both the Yate and FS server. Has anyone had similar a similar issue with Nextone (Genband)? Just to mention we are sending calls to this carrier from FS without issue. Thanks in advance, Errol -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/f1e5732b/attachment.html From mike at jerris.com Tue Aug 31 08:22:36 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Aug 2010 11:22:36 -0400 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: <1283265385508-5482941.post@n2.nabble.com> References: <1283265385508-5482941.post@n2.nabble.com> Message-ID: <34BC10D3-519C-45BE-82DB-E2DF0105313B@jerris.com> You know they are not going to let everyone move all their US termination over to googles bill, seems like everyone is all excited to run their call center and not have to pay for calls anymore. Its never going to happen. On Aug 31, 2010, at 10:36 AM, mazilo wrote: > > > jesse zhao wrote: >> FS can integrate with Google Voice via some java library to make >> outgoing calls for U.S and Canada.. >> >> jesse > We certainly will appreciate if you can provide the link to show how to do > this instead of having everyone re-invent the wheel. > From mike at jerris.com Tue Aug 31 08:28:29 2010 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Aug 2010 11:28:29 -0400 Subject: [Freeswitch-users] bridge In-Reply-To: References: Message-ID: uuid_bridge On Aug 31, 2010, at 7:13 AM, babak yakhchali wrote: > is it possible (in mod_managed or something similar) to create a new session and originate a call and if it was successful bridge it as the b leg of a call that moh is playing for it already(a waiting call)? From Nabble at slickdeals.endjunk.com Tue Aug 31 08:37:08 2010 From: Nabble at slickdeals.endjunk.com (mazilo) Date: Tue, 31 Aug 2010 08:37:08 -0700 (PDT) Subject: [Freeswitch-users] bridge In-Reply-To: References: Message-ID: <1283269028539-5483223.post@n2.nabble.com> Michael Jerris wrote: > > uuid_bridge Can you show us how to do this? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/bridge-tp5482167p5483223.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ovvenkatesan at gmail.com Tue Aug 31 08:57:48 2010 From: ovvenkatesan at gmail.com (ovvenkat) Date: Tue, 31 Aug 2010 21:27:48 +0530 Subject: [Freeswitch-users] not able to connect freeswitch with mysql Message-ID: Hi to all, I have following this wiki page to configure mysql and freeswitch. http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc When I run the command " *isql -v maxpowersoft_odbc* ", I can able to connect to the MySql database. When , I run the following java script file, *test.js* use("ODBC"); var db = new ODBC("maxpowersoft_odbc", "dbuser", "dbpass"); db.connect(); I am getting error like , 2010-08-31 21:24:06.583606 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified Any one help me to relove my issue plz. thanks in advance. Regards, Venkat. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/578186ea/attachment.html From adolfo at delorenzo.mobi Tue Aug 31 09:10:01 2010 From: adolfo at delorenzo.mobi (Adolfo Delorenzo) Date: Tue, 31 Aug 2010 13:10:01 -0300 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: I found version 2.0.0.68-2 deb and rpm. Please let me know if this works and I'll upload it on my webserver. BRgds, AD On 31 August 2010 12:21, Shamun toha md wrote: > I got once, but it was a virus for CentOS, lol :P > > Thanks > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/55b01d3e/attachment.html From gmaruzz at celliax.org Tue Aug 31 09:23:03 2010 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 31 Aug 2010 18:23:03 +0200 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: Adolfo, you're very very nice, thank you for your offer, but... The version that is proven to work well and that is recommended for production is the 2.0.72 ***static*** build, that is in .tar.gz format and is not packaged like rpm or deb. 2.0.72 is the *stable* version of Skype (2.1.81 is a beta) and in the "static build" is linked statically with QT, so it don't requires external qt libraries with all the nightmares and instability of the oh so many versions out there. -giovanni On Tue, Aug 31, 2010 at 6:10 PM, Adolfo Delorenzo wrote: > I found version 2.0.0.68-2 deb and rpm. Please let me know if this works and > I'll upload it on my webserver. > > BRgds, > > AD > > On 31 August 2010 12:21, Shamun toha md wrote: >> >> I got once, but it was a virus for CentOS, lol :P >> >> Thanks >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From adolfo at delorenzo.mobi Tue Aug 31 09:24:11 2010 From: adolfo at delorenzo.mobi (Adolfo Delorenzo) Date: Tue, 31 Aug 2010 13:24:11 -0300 Subject: [Freeswitch-users] incoming calls - newbie question Message-ID: Hello, I am not receiving incoming calls via enum (my ITAD is 1325) and if someone calls for example myextension at mydomain. Where in the documentation can I learn how to enable incoming calls? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/477818da/attachment.html From mthakershi at gmail.com Tue Aug 31 10:19:12 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 31 Aug 2010 12:19:12 -0500 Subject: [Freeswitch-users] mod_managed API interface question Message-ID: Hello, Please let me know if this is feasible / possible. I need to make outgoing calls based on a scheduler using FS (SIP). I am planning to write a windows service that runs at particular time interval (say 10 minutes), checks who needs to be called, fires commands to originate these calls. I have a single FS command line running (where FS console is running). I am planning to fire FS commands such as: ---------------------- >managed apidemo a1 123 API CALL [managed(apidemo a1 123)] output: ApiDemo executed with args 'a1 123' and event type API. >managedrun apidemo a1 123 API CALL [managedrun(apidemo a1 123)] output: +OK 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo on a background thread #(3), with args 'a1 123'. 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ExecuteBackground in apidemo completed. ---------------------- So I will pass some arguments to the mod_managed script and it will fire an API to originate call. My main question is: How can I communicate windows service to fire FreeSwitch commands? Thank you for any help or guidance. From mthakershi at gmail.com Tue Aug 31 10:19:42 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 31 Aug 2010 12:19:42 -0500 Subject: [Freeswitch-users] mod_managed API interface question Message-ID: Hello, Please let me know if this is feasible / possible. I need to make outgoing calls based on a scheduler using FS (SIP). I am planning to write a windows service that runs at particular time interval (say 10 minutes), checks who needs to be called, fires commands to originate these calls. I have a single FS command line running (where FS console is running). I am planning to fire FS commands such as: ---------------------- >managed apidemo a1 123 API CALL [managed(apidemo a1 123)] output: ApiDemo executed with args 'a1 123' and event type API. >managedrun apidemo a1 123 API CALL [managedrun(apidemo a1 123)] output: +OK 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo on a background thread #(3), with args 'a1 123'. 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ExecuteBackground in apidemo completed. ---------------------- So I will pass some arguments to the mod_managed script and it will fire an API to originate call. My main question is: How can I communicate windows service to fire FreeSwitch commands? Thank you for any help or guidance. From mthakershi at gmail.com Tue Aug 31 10:19:12 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 31 Aug 2010 12:19:12 -0500 Subject: [Freeswitch-users] mod_managed API interface question Message-ID: Hello, Please let me know if this is feasible / possible. I need to make outgoing calls based on a scheduler using FS (SIP). I am planning to write a windows service that runs at particular time interval (say 10 minutes), checks who needs to be called, fires commands to originate these calls. I have a single FS command line running (where FS console is running). I am planning to fire FS commands such as: ---------------------- >managed apidemo a1 123 API CALL [managed(apidemo a1 123)] output: ApiDemo executed with args 'a1 123' and event type API. >managedrun apidemo a1 123 API CALL [managedrun(apidemo a1 123)] output: +OK 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo on a background thread #(3), with args 'a1 123'. 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ExecuteBackground in apidemo completed. ---------------------- So I will pass some arguments to the mod_managed script and it will fire an API to originate call. My main question is: How can I communicate windows service to fire FreeSwitch commands? Thank you for any help or guidance. From mnhassan at usa.net Tue Aug 31 10:23:02 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Tue, 31 Aug 2010 23:23:02 +0600 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: <34BC10D3-519C-45BE-82DB-E2DF0105313B@jerris.com> References: <1283265385508-5482941.post@n2.nabble.com> <34BC10D3-519C-45BE-82DB-E2DF0105313B@jerris.com> Message-ID: And, it seems this feature is only available for Google Voice customers. Regards HASSAN On 2010-08-31, Michael Jerris wrote: > You know they are not going to let everyone move all their US termination > over to googles bill, seems like everyone is all excited to run their call > center and not have to pay for calls anymore. Its never going to happen. > > > On Aug 31, 2010, at 10:36 AM, mazilo wrote: > >> >> >> jesse zhao wrote: >>> FS can integrate with Google Voice via some java library to make >>> outgoing calls for U.S and Canada.. >>> >>> jesse >> We certainly will appreciate if you can provide the link to show how to do >> this instead of having everyone re-invent the wheel. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From mnhassan at usa.net Tue Aug 31 10:26:54 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Tue, 31 Aug 2010 23:26:54 +0600 Subject: [Freeswitch-users] mod_managed API interface question In-Reply-To: References: Message-ID: IMHO, the best way to do this is using the ESL. The wiki has some examples on how this can be done. Regards HASSAN On 2010-08-31, Malay Thakershi wrote: > Hello, > > Please let me know if this is feasible / possible. I need to make > outgoing calls based on a scheduler using FS (SIP). > > I am planning to write a windows service that runs at particular time > interval (say 10 minutes), checks who needs to be called, fires > commands to originate these calls. > > I have a single FS command line running (where FS console is running). > I am planning to fire FS commands such as: > ---------------------- >>managed apidemo a1 123 > API CALL [managed(apidemo a1 123)] output: > ApiDemo executed with args 'a1 123' and event type API. > > >managedrun apidemo a1 123 > API CALL [managedrun(apidemo a1 123)] output: > +OK > 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo > on a background thread #(3), with args 'a1 123'. > 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() > ExecuteBackground in apidemo completed. > ---------------------- > > So I will pass some arguments to the mod_managed script and it will > fire an API to originate call. > > My main question is: > How can I communicate windows service to fire FreeSwitch commands? > > Thank you for any help or guidance. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From mthakershi at gmail.com Tue Aug 31 10:32:07 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 31 Aug 2010 12:32:07 -0500 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: References: <1283265385508-5482941.post@n2.nabble.com> <34BC10D3-519C-45BE-82DB-E2DF0105313B@jerris.com> Message-ID: Does Google Voice (paid service) provide SIP terminate/originate like others (say Vitelity)? If so, what is their cost? On Tue, Aug 31, 2010 at 12:23 PM, Nyamul Hassan wrote: > And, it seems this feature is only available for Google Voice customers. > > Regards > HASSAN > > > > On 2010-08-31, Michael Jerris wrote: >> You know they are not going to let everyone move all their US termination >> over to googles bill, seems like everyone is all excited to run their call >> center and not have to pay for calls anymore. ?Its never going to happen. >> >> >> On Aug 31, 2010, at 10:36 AM, mazilo wrote: >> >>> >>> >>> jesse zhao wrote: >>>> FS can integrate with Google Voice via some java library to make >>>> outgoing calls for U.S and Canada.. >>>> >>>> jesse >>> We certainly will appreciate if you can provide the link to show how to do >>> this instead of having everyone re-invent the wheel. >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > Sent from my mobile device > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mnhassan at usa.net Tue Aug 31 10:33:26 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Tue, 31 Aug 2010 23:33:26 +0600 Subject: [Freeswitch-users] windows build error In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05818D@cooper> Message-ID: If not in the Wiki, can you please update this bit? Regards HASSAN On 2010-08-31, Seven Du wrote: > Thanks, I re-Cloned with auto CRLF off and it worked. > > On Mon, Aug 30, 2010 at 10:13 PM, Peter Olsson > wrote: >> Make sure to disable all kinds of CRLF/LF conversions. By default git on >> Windows seems to enable this. There is a setting called autocrlf, make >> sure it's set to false. >> >> /Peter >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> [freeswitch-users-bounces at lists.freeswitch.org] för Seven Du >> [dujinfang at gmail.com] >> Skickat: den 30 augusti 2010 15:52 >> Till: freeswitch-users >> ?mne: [Freeswitch-users] windows build error >> >> Hi, >> >> I'm new to windows build. I cloned code a few days ago and pulled to >> head. Can could not build sofia. >> >> c:\workspace\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(119) : >> fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': >> No such file or directory >> >> >> I tried to rebuild but the result was the same. >> >> I can start freeswitch though mod_sofia and mod file format (not >> exactly remember) cannot load. >> >> Any hint? >> >> I will re-clone code and try again tomorrow. >> >> Thanks. >> >> -- >> Blog: http://www.dujinfang.com >> Proj: ?http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:4c7bb97c32934187118539! >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Blog: http://www.dujinfang.com > Proj:? http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From mthakershi at gmail.com Tue Aug 31 10:39:15 2010 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 31 Aug 2010 12:39:15 -0500 Subject: [Freeswitch-users] mod_managed API interface question In-Reply-To: References: Message-ID: I am using Windows server. Can this be installed on Windows server 2008? On Tue, Aug 31, 2010 at 12:26 PM, Nyamul Hassan wrote: > IMHO, the best way to do this is using the ESL. The wiki has some > examples on how this can be done. > > Regards > HASSAN > > > On 2010-08-31, Malay Thakershi wrote: >> Hello, >> >> Please let me know if this is feasible / possible. I need to make >> outgoing calls based on a scheduler using FS (SIP). >> >> I am planning to write a windows service that runs at particular time >> interval (say 10 minutes), checks who needs to be called, fires >> commands to originate these calls. >> >> I have a single FS command line running (where FS console is running). >> I am planning to fire FS commands such as: >> ---------------------- >>>managed apidemo a1 123 >> ?API CALL [managed(apidemo a1 123)] output: >> ?ApiDemo executed with args 'a1 123' and event type API. >> >> ?>managedrun apidemo a1 123 >> ?API CALL [managedrun(apidemo a1 123)] output: >> ?+OK >> ?2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo >> on a background thread #(3), with args 'a1 123'. >> ?2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() >> ExecuteBackground in apidemo completed. >> ---------------------- >> >> So I will pass some arguments to the mod_managed script and it will >> fire an API to originate call. >> >> My main question is: >> How can I communicate windows service to fire FreeSwitch commands? >> >> Thank you for any help or guidance. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > Sent from my mobile device > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mnhassan at usa.net Tue Aug 31 10:51:18 2010 From: mnhassan at usa.net (Nyamul Hassan) Date: Tue, 31 Aug 2010 23:51:18 +0600 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: References: <1283265385508-5482941.post@n2.nabble.com> <34BC10D3-519C-45BE-82DB-E2DF0105313B@jerris.com> Message-ID: Google Voice is only available for customers in US and Canada. Regards HASSAN On 2010-08-31, Malay Thakershi wrote: > Does Google Voice (paid service) provide SIP terminate/originate like > others (say Vitelity)? If so, what is their cost? > > On Tue, Aug 31, 2010 at 12:23 PM, Nyamul Hassan wrote: >> And, it seems this feature is only available for Google Voice customers. >> >> Regards >> HASSAN >> >> >> >> On 2010-08-31, Michael Jerris wrote: >>> You know they are not going to let everyone move all their US termination >>> over to googles bill, seems like everyone is all excited to run their >>> call >>> center and not have to pay for calls anymore. ?Its never going to happen. >>> >>> >>> On Aug 31, 2010, at 10:36 AM, mazilo wrote: >>> >>>> >>>> >>>> jesse zhao wrote: >>>>> FS can integrate with Google Voice via some java library to make >>>>> outgoing calls for U.S and Canada.. >>>>> >>>>> jesse >>>> We certainly will appreciate if you can provide the link to show how to >>>> do >>>> this instead of having everyone re-invent the wheel. >>>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> -- >> Sent from my mobile device >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From peter.olsson at visionutveckling.se Tue Aug 31 10:56:41 2010 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 31 Aug 2010 19:56:41 +0200 Subject: [Freeswitch-users] windows build error In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05818D@cooper> , Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC0581A1@cooper> It should be there already - but I guess we might need to make that part a bit more clear. There's been quite a few questions about this after conversion to git. I will have a look on it. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Nyamul Hassan [mnhassan at usa.net] Skickat: den 31 augusti 2010 19:33 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] windows build error If not in the Wiki, can you please update this bit? Regards HASSAN On 2010-08-31, Seven Du wrote: > Thanks, I re-Cloned with auto CRLF off and it worked. > > On Mon, Aug 30, 2010 at 10:13 PM, Peter Olsson > wrote: >> Make sure to disable all kinds of CRLF/LF conversions. By default git on >> Windows seems to enable this. There is a setting called autocrlf, make >> sure it's set to false. >> >> /Peter >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >> [freeswitch-users-bounces at lists.freeswitch.org] för Seven Du >> [dujinfang at gmail.com] >> Skickat: den 30 augusti 2010 15:52 >> Till: freeswitch-users >> ?mne: [Freeswitch-users] windows build error >> >> Hi, >> >> I'm new to windows build. I cloned code a few days ago and pulled to >> head. Can could not build sofia. >> >> c:\workspace\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(119) : >> fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': >> No such file or directory >> >> >> I tried to rebuild but the result was the same. >> >> I can start freeswitch though mod_sofia and mod file format (not >> exactly remember) cannot load. >> >> Any hint? >> >> I will re-clone code and try again tomorrow. >> >> Thanks. >> >> -- >> Blog: http://www.dujinfang.com >> Proj: http://www.freeswitch.org.cn >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4c7d3e5532931622379309! From mustafa.pk at gmail.com Tue Aug 31 11:35:08 2010 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Tue, 31 Aug 2010 23:35:08 +0500 Subject: [Freeswitch-users] Skype 2.0.72 or 2.1.0.81 In-Reply-To: References: Message-ID: Hi everyone, drop me a line if you are looking for skype 2.0.72 (dynamically linked version) best regards. On Tue, Aug 31, 2010 at 9:23 PM, Giovanni Maruzzelli wrote: > Adolfo, > > you're very very nice, thank you for your offer, but... > > The version that is proven to work well and that is recommended for > production is the 2.0.72 ***static*** build, that is in .tar.gz format > and is not packaged like rpm or deb. > > 2.0.72 is the *stable* version of Skype (2.1.81 is a beta) and in the > "static build" is linked statically with QT, so it don't requires > external qt libraries with all the nightmares and instability of the > oh so many versions out there. > > -giovanni > > On Tue, Aug 31, 2010 at 6:10 PM, Adolfo Delorenzo > wrote: > > I found version 2.0.0.68-2 deb and rpm. Please let me know if this works > and > > I'll upload it on my webserver. > > > > BRgds, > > > > AD > > > > On 31 August 2010 12:21, Shamun toha md wrote: > >> > >> I got once, but it was a virus for CentOS, lol :P > >> > >> Thanks > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/adb51568/attachment.html From rupa at rupa.com Tue Aug 31 11:37:44 2010 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 31 Aug 2010 13:37:44 -0500 Subject: [Freeswitch-users] gTalk new phone calls feature In-Reply-To: References: <1283265385508-5482941.post@n2.nabble.com> <34BC10D3-519C-45BE-82DB-E2DF0105313B@jerris.com> Message-ID: No, they do not have a SIP interface we can use. They originate/terminate on the PSTN. (unless you have a gizmo5 account which are no longer offered). One would hope they offer a SIP interface again someday... On Tue, Aug 31, 2010 at 12:32 PM, Malay Thakershi wrote: > Does Google Voice (paid service) provide SIP terminate/originate like > others (say Vitelity)? If so, what is their cost? > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/18823b65/attachment.html From adminjew at gmail.com Tue Aug 31 12:20:01 2010 From: adminjew at gmail.com (Yitzchok) Date: Tue, 31 Aug 2010 15:20:01 -0400 Subject: [Freeswitch-users] mod_managed API interface question In-Reply-To: References: Message-ID: There is a .net implementation for the FreeSWITCH esl in the contrib git repository. You can use that in a .Net windows service if you want. Yitzchok On Tue, Aug 31, 2010 at 1:39 PM, Malay Thakershi wrote: > I am using Windows server. Can this be installed on Windows server 2008? > > On Tue, Aug 31, 2010 at 12:26 PM, Nyamul Hassan wrote: > > IMHO, the best way to do this is using the ESL. The wiki has some > > examples on how this can be done. > > > > Regards > > HASSAN > > > > > > On 2010-08-31, Malay Thakershi wrote: > >> Hello, > >> > >> Please let me know if this is feasible / possible. I need to make > >> outgoing calls based on a scheduler using FS (SIP). > >> > >> I am planning to write a windows service that runs at particular time > >> interval (say 10 minutes), checks who needs to be called, fires > >> commands to originate these calls. > >> > >> I have a single FS command line running (where FS console is running). > >> I am planning to fire FS commands such as: > >> ---------------------- > >>>managed apidemo a1 123 > >> API CALL [managed(apidemo a1 123)] output: > >> ApiDemo executed with args 'a1 123' and event type API. > >> > >> >managedrun apidemo a1 123 > >> API CALL [managedrun(apidemo a1 123)] output: > >> +OK > >> 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo > >> on a background thread #(3), with args 'a1 123'. > >> 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() > >> ExecuteBackground in apidemo completed. > >> ---------------------- > >> > >> So I will pass some arguments to the mod_managed script and it will > >> fire an API to originate call. > >> > >> My main question is: > >> How can I communicate windows service to fire FreeSwitch commands? > >> > >> Thank you for any help or guidance. > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > -- > > Sent from my mobile device > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/3421cbdf/attachment.html From jeff at jefflenk.com Tue Aug 31 12:37:59 2010 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 31 Aug 2010 12:37:59 -0700 (PDT) Subject: [Freeswitch-users] mod_managed API interface question In-Reply-To: References: Message-ID: <1283283479123-5484160.post@n2.nabble.com> you can also use the esl wrapper in /libs/esl/managed which may prove easier to support -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-managed-API-interface-question-tp5483674p5484160.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lloydie.t at gmail.com Tue Aug 31 12:37:47 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Tue, 31 Aug 2010 20:37:47 +0100 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world Message-ID: Went to my cousins party on saturday (Leeds UK) and bumped into a friend of hers, who just so happens know a bit about FS and has a couple of production boxes running FS for his wholesale carrier business. He recommended if I were running FS on ubuntu to recompile it, so that dynamic ticks were switch off and set the timer frequency to 1000 Hz. I have done that, but the CPU idle seems to have increased 5% to around 15% and spikes to 25% when I make a single call. Should that be expected on a lowly box with an old celeron CPU and 500mb RAM? model name : Celeron (Coppermine) stepping : 10 cpu MHz : 901.571 cache size : 128 KB MemTotal: 508316 kB MemFree: 276248 kB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/c5bc2a82/attachment.html From anthony.minessale at gmail.com Tue Aug 31 13:12:05 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Aug 2010 15:12:05 -0500 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world In-Reply-To: References: Message-ID: Try a newer or older version of the kernel as well. On Tue, Aug 31, 2010 at 2:37 PM, lloyd thomas wrote: > Went to my cousins party on saturday (Leeds UK) and bumped into a friend of > hers, who just so happens know a bit about FS and has a couple of production > boxes running FS for his wholesale carrier business. > He recommended if I were running FS on ubuntu to recompile it, so that > dynamic ticks were switch off and set the timer frequency to 1000 Hz. > I have done that, but the CPU idle seems to have increased 5% to around 15% > and spikes to 25% when I make a single call. > > Should that be expected on a lowly box with an old celeron CPU and 500mb > RAM? > > > model name????? : Celeron (Coppermine) > stepping??????? : 10 > cpu MHz???????? : 901.571 > cache size????? : 128 KB > > MemTotal:???????? 508316 kB > MemFree:????????? 276248 kB > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From brian at freeswitch.org Tue Aug 31 13:27:57 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Aug 2010 15:27:57 -0500 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world In-Reply-To: References: Message-ID: <4B72146B-D099-4AAD-B0FF-F1B66403595C@freeswitch.org> I would highly recommend faster boxes...we have wall plugs that move quicker these days. :) /b On Aug 31, 2010, at 2:37 PM, lloyd thomas wrote: > Went to my cousins party on saturday (Leeds UK) and bumped into a friend of hers, who just so happens know a bit about FS and has a couple of production boxes running FS for his wholesale carrier business. > He recommended if I were running FS on ubuntu to recompile it, so that dynamic ticks were switch off and set the timer frequency to 1000 Hz. > I have done that, but the CPU idle seems to have increased 5% to around 15% and spikes to 25% when I make a single call. > > Should that be expected on a lowly box with an old celeron CPU and 500mb RAM? > > > model name : Celeron (Coppermine) > stepping : 10 > cpu MHz : 901.571 > cache size : 128 KB > > MemTotal: 508316 kB > MemFree: 276248 kB From quentusrex at gmail.com Tue Aug 31 13:37:04 2010 From: quentusrex at gmail.com (William King) Date: Tue, 31 Aug 2010 13:37:04 -0700 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world In-Reply-To: <4B72146B-D099-4AAD-B0FF-F1B66403595C@freeswitch.org> References: <4B72146B-D099-4AAD-B0FF-F1B66403595C@freeswitch.org> Message-ID: <4C7D67F0.9010401@gmail.com> Which release of Ubuntu are you running? I have FS running on a very old machine(slower than yours) and I have seen an issue like this before. It was related to the release and the kernel. -William On 08/31/2010 01:27 PM, Brian West wrote: > I would highly recommend faster boxes...we have wall plugs that move quicker these days. :) > > /b > > On Aug 31, 2010, at 2:37 PM, lloyd thomas wrote: > >> Went to my cousins party on saturday (Leeds UK) and bumped into a friend of hers, who just so happens know a bit about FS and has a couple of production boxes running FS for his wholesale carrier business. >> He recommended if I were running FS on ubuntu to recompile it, so that dynamic ticks were switch off and set the timer frequency to 1000 Hz. >> I have done that, but the CPU idle seems to have increased 5% to around 15% and spikes to 25% when I make a single call. >> >> Should that be expected on a lowly box with an old celeron CPU and 500mb RAM? >> >> >> model name : Celeron (Coppermine) >> stepping : 10 >> cpu MHz : 901.571 >> cache size : 128 KB >> >> MemTotal: 508316 kB >> MemFree: 276248 kB > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From 12ukwn at gmail.com Tue Aug 31 13:58:00 2010 From: 12ukwn at gmail.com (Jean-Yves F. Barbier) Date: Tue, 31 Aug 2010 22:58:00 +0200 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world In-Reply-To: <4C7D67F0.9010401@gmail.com> References: <4B72146B-D099-4AAD-B0FF-F1B66403595C@freeswitch.org> <4C7D67F0.9010401@gmail.com> Message-ID: <20100831225800.3ab082f9@anubis.defcon1> On Tue, 31 Aug 2010 13:37:04 -0700, William King wrote: Yep, I confirm, 2.6.35 line seems to have timing PBs (and others), 2.6.33.4 is more stable (ubuntu often thinks newer is better...) > Which release of Ubuntu are you running? I have FS running on a very old > machine(slower than yours) and I have seen an issue like this before. It > was related to the release and the kernel. -- I try to keep an open mind, but not so open that my brains fall out. -- Judge Harold T. Stone From quentusrex at gmail.com Tue Aug 31 14:05:00 2010 From: quentusrex at gmail.com (William King) Date: Tue, 31 Aug 2010 14:05:00 -0700 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world In-Reply-To: <20100831225800.3ab082f9@anubis.defcon1> References: <4B72146B-D099-4AAD-B0FF-F1B66403595C@freeswitch.org> <4C7D67F0.9010401@gmail.com> <20100831225800.3ab082f9@anubis.defcon1> Message-ID: <4C7D6E7C.8080702@gmail.com> I am running this kernel in production and seems to be very stable: 2.6.32-24-server x86_64 GNU/Linux Using release 10.04.1 LTS -William King On 08/31/2010 01:58 PM, Jean-Yves F. Barbier wrote: > On Tue, 31 Aug 2010 13:37:04 -0700, William King > wrote: > > Yep, I confirm, 2.6.35 line seems to have timing PBs (and others), > 2.6.33.4 is more stable (ubuntu often thinks newer is better...) > >> Which release of Ubuntu are you running? I have FS running on a very old >> machine(slower than yours) and I have seen an issue like this before. It >> was related to the release and the kernel. > From lloydie.t at gmail.com Tue Aug 31 14:32:59 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Tue, 31 Aug 2010 22:32:59 +0100 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world In-Reply-To: <4C7D6E7C.8080702@gmail.com> References: <4B72146B-D099-4AAD-B0FF-F1B66403595C@freeswitch.org> <4C7D67F0.9010401@gmail.com> <20100831225800.3ab082f9@anubis.defcon1> <4C7D6E7C.8080702@gmail.com> Message-ID: I am running 2.6.28.10-freeswitch On 31 August 2010 22:05, William King wrote: > I am running this kernel in production and seems to be very stable: > 2.6.32-24-server x86_64 GNU/Linux > > Using release 10.04.1 LTS > > -William King > > On 08/31/2010 01:58 PM, Jean-Yves F. Barbier wrote: > > On Tue, 31 Aug 2010 13:37:04 -0700, William King > > wrote: > > > > Yep, I confirm, 2.6.35 line seems to have timing PBs (and others), > > 2.6.33.4 is more stable (ubuntu often thinks newer is better...) > > > >> Which release of Ubuntu are you running? I have FS running on a very old > >> machine(slower than yours) and I have seen an issue like this before. It > >> was related to the release and the kernel. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/86279737/attachment.html From lloydie.t at gmail.com Tue Aug 31 14:33:20 2010 From: lloydie.t at gmail.com (lloyd thomas) Date: Tue, 31 Aug 2010 22:33:20 +0100 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world In-Reply-To: References: <4B72146B-D099-4AAD-B0FF-F1B66403595C@freeswitch.org> <4C7D67F0.9010401@gmail.com> <20100831225800.3ab082f9@anubis.defcon1> <4C7D6E7C.8080702@gmail.com> Message-ID: 32 bit On 31 August 2010 22:32, lloyd thomas wrote: > I am running 2.6.28.10-freeswitch > > > On 31 August 2010 22:05, William King wrote: > >> I am running this kernel in production and seems to be very stable: >> 2.6.32-24-server x86_64 GNU/Linux >> >> Using release 10.04.1 LTS >> >> -William King >> >> On 08/31/2010 01:58 PM, Jean-Yves F. Barbier wrote: >> > On Tue, 31 Aug 2010 13:37:04 -0700, William King >> > wrote: >> > >> > Yep, I confirm, 2.6.35 line seems to have timing PBs (and others), >> > 2.6.33.4 is more stable (ubuntu often thinks newer is better...) >> > >> >> Which release of Ubuntu are you running? I have FS running on a very >> old >> >> machine(slower than yours) and I have seen an issue like this before. >> It >> >> was related to the release and the kernel. >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/0f6bacab/attachment-0001.html From jock.mckechnie at gmail.com Tue Aug 31 14:47:12 2010 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Tue, 31 Aug 2010 16:47:12 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> <964D421A-B03D-4FCB-90BE-0F49D573050F@freeswitch.org> Message-ID: I decided to go over all of the things I had tried, and discovered that the below gives me "early media", at least, it provides RTP but it is playing back silence, so it would appear, as Brian suggested above, somehow I'm not convincing FS _what_ it should be playing. I used a 'ringback', as I'm bridging, which I believe is the correct setting to use. I continue to simplify the config in hopes of just getting some kind of early media, which is why I'm using a cadence description for the ringback, instead of playing a WAV file, which is the final goal. With the above, I get apparently a normal "conversation" (INVITE, 100, 183, early media RTP, 200 OK, playback RTP), but the early media is "blank". I'm getting closer, but still not quite there. Can anyone point out what I'm still missing? My thanks; - Jock On Tue, Aug 31, 2010 at 8:58 AM, Jock McKechnie wrote: > > > On Mon, Aug 30, 2010 at 5:48 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> did you try the ones I listed? >> >> I did not phrase them in xml because it's redundant but you should >> simply be able to make a simple extension >> that calls playback WITHOUT calling answer and the file will play in >> early media? >> >> Maybe it's too easy? >> >> This will play a file in early media, answer then play it again as >> in-call media. >> >> >> >> >> >> >> >> >> >> > I did not, I misunderstood your initial eMail, clearly. > > I have rigged up the above and something quite odd happens - definitely > much closer, but still odd. The CLI shows FreeSWITCH accept the call, there > is almost a full ten second pause, and then I hear a 'click' on the line, > there is another pause, and then I get the latter Playback (I've used two > different audio files to ensure I can tell the difference). > > However, I've tcpdumped and according to Wireshark: > INVITE > 100 Trying > (10.1 second pause) > Early Media RTP > 200 OK > Latter Playback RTP > BYE > > A review of the audio through WireShark verifies that FS is playing the > first WAV file as, apparently, early media. > > So I appear to have two problems: Firstly, there's a ten second pause. I > did not reinstate my sleep statement, so I don't understand why it's waiting > so long after getting the call. And although FreeSWITCH is now definitely > pumping out the early media, Verizon is, apparently, not handing it back. Is > there any possibility something slightly non-standard is happening here, so > VZ's equipment (cough, Sonus, cough) is unaware it should be passing RTP on > to the calling party? I'm afraid my telephony knowledge is primarily VoIP > related to Asterisk (wail, moan, etc, etc) and OpenSIPS. I was under the > impression that a 183 Ringing was required in the flow to alert the far-end > (calling party) that early media is coming, but I could, of course, be very > wrong. > > Thank you very much Anthony; > > - Jock > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/ac2d4e81/attachment.html From brian at freeswitch.org Tue Aug 31 14:55:01 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Aug 2010 16:55:01 -0500 Subject: [Freeswitch-users] Early Media troubles In-Reply-To: References: <6392BDA3-442D-4F94-A67C-CB4878FDBA55@freeswitch.org> <8BF5D8CF-CA43-4152-8E1C-C253EBFDAD96@freeswitch.org> <964D421A-B03D-4FCB-90BE-0F49D573050F@freeswitch.org> Message-ID: <664D50F3-40B1-47C2-8022-418CCE419FAE@freeswitch.org> What device are you working with.. because handling early media is optional so your device might be ignoring it. /b On Aug 31, 2010, at 4:47 PM, Jock McKechnie wrote: > I'm getting closer, but still not quite there. Can anyone point out what I'm still missing? > > My thanks; > > - Jock From christian.loeschenkohl at xpirio.com Tue Aug 31 15:28:54 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Wed, 01 Sep 2010 00:28:54 +0200 Subject: [Freeswitch-users] setting switch_r_sdp with outbound event socket Message-ID: <4C7D8226.4080504@xpirio.com> hello i try to moditfy the sip sdp to get some user agent working, but setting switch_r_sdp doesn't work the way i try it on the loopback interface i do see the command of setting the variable like this sendmsg call-command: execute execute-app-name: set execute-app-arg: (btw this example sdp is ok) but on the console i get the error [ERR] switch_ivr.c:469 Invalid Command! and the b-leg doesn't get the modified sdp. what could i get wrong here? br -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From brian at freeswitch.org Tue Aug 31 15:34:26 2010 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Aug 2010 17:34:26 -0500 Subject: [Freeswitch-users] setting switch_r_sdp with outbound event socket In-Reply-To: <4C7D8226.4080504@xpirio.com> References: <4C7D8226.4080504@xpirio.com> Message-ID: <51496716-6E26-494D-8F41-B964BDB6675F@freeswitch.org> Why are you trying to put it in the CDATA? You know you only have to do that if you're trying to do that in the XML dialplan. /b On Aug 31, 2010, at 5:28 PM, Christian L?schenkohl wrote: > hello > > i try to moditfy the sip sdp to get some user agent working, > but setting switch_r_sdp doesn't work the way i try it > > on the loopback interface i do see the command of setting the variable like this > > sendmsg > call-command: execute > execute-app-name: set > execute-app-arg: o=root 713636917 713636917 IN IP4 x.x.x.x > s=call > c=IN IP4 x.x.x.x > t=0 0 > m=audio 17300 RTP/AVP 0 9 99 3 18 4 101 > a=rtpmap:0 pcmu/8000 > a=rtpmap:9 g722/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=fmtp:18 annexb=no > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > ]]> > > (btw this example sdp is ok) > > but on the console i get the error [ERR] switch_ivr.c:469 Invalid Command! > and the b-leg doesn't get the modified sdp. > > what could i get wrong here? > > br From christian.loeschenkohl at xpirio.com Tue Aug 31 15:55:35 2010 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Wed, 01 Sep 2010 00:55:35 +0200 Subject: [Freeswitch-users] setting switch_r_sdp with outbound event socket In-Reply-To: <51496716-6E26-494D-8F41-B964BDB6675F@freeswitch.org> References: <4C7D8226.4080504@xpirio.com> <51496716-6E26-494D-8F41-B964BDB6675F@freeswitch.org> Message-ID: <4C7D8867.50109@xpirio.com> thank you for your reply i read it here http://wiki.freeswitch.org/wiki/Codec_negotiation#Rewriting_SDP and it seemed logic to me (which isn't true as i seems). of course i tried it without CDATA - then the command on the loopback looks like this sendmsg call-command: execute execute-app-name: set execute-app-arg: switch_r_sdp=v=0 o=root 1051875459 1051875459 IN IP4 x.x.x.x s=call c=IN IP4 x.x.x.x t=0 0 m=audio 19156 RTP/AVP 0 9 99 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 and i also do get [ERR] switch_ivr.c:469 Invalid Command! these lines print out the error (switch_ivr.c:469) if (zstr(cmd)) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Invalid Command!\n"); return SWITCH_STATUS_FALSE; } br On 2010-09-01 00:34, Brian West wrote: > Why are you trying to put it in the CDATA? You know you only have to do that if you're trying to do that in the XML dialplan. > > /b > > On Aug 31, 2010, at 5:28 PM, Christian L?schenkohl wrote: > >> hello >> >> i try to moditfy the sip sdp to get some user agent working, >> but setting switch_r_sdp doesn't work the way i try it >> >> on the loopback interface i do see the command of setting the variable like this >> >> sendmsg >> call-command: execute >> execute-app-name: set >> execute-app-arg:> o=root 713636917 713636917 IN IP4 x.x.x.x >> s=call >> c=IN IP4 x.x.x.x >> t=0 0 >> m=audio 17300 RTP/AVP 0 9 99 3 18 4 101 >> a=rtpmap:0 pcmu/8000 >> a=rtpmap:9 g722/8000 >> a=rtpmap:3 gsm/8000 >> a=rtpmap:18 g729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:4 g723/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> ]]> >> >> (btw this example sdp is ok) >> >> but on the console i get the error [ERR] switch_ivr.c:469 Invalid Command! >> and the b-leg doesn't get the modified sdp. >> >> what could i get wrong here? >> >> br > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl at xpirio.com From msc at freeswitch.org Tue Aug 31 18:00:04 2010 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Aug 2010 18:00:04 -0700 Subject: [Freeswitch-users] Nextone (Genband) >> Freeswitch Ringback issue In-Reply-To: References: Message-ID: Get a pcap and see if RTP is flowing when the 183 is sent. If so then you know the carrier is blowing smoke. If not the you know that your ring_ready app isn't working for whatever reason. -MC On Tue, Aug 31, 2010 at 3:14 AM, Errol Samuels wrote: > Hello All, > > I have a weird situation where a carrier is sending calls to our FS server > but not hearing a Ringback tone, however once the call connects they are > getting audio but when this carrier sends calls to our Yate server they are > getting the Ringback without a problem. > > I have tried setting the Ring_Ready below but it didn't help the situation > either. > > > > > > > > The traces from both the FS server and the Yate server show we are sending: > > 100 Trying > 180 Ringing > 183 Session Progress with SDP > > The carrier is claiming they are using the same configuration for both the > Yate and FS server. Has anyone had similar a similar issue with Nextone > (Genband)? > > Just to mention we are sending calls to this carrier from FS without issue. > > Thanks in advance, > > Errol > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/fc410e02/attachment-0001.html From anthony.minessale at gmail.com Tue Aug 31 18:31:07 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Aug 2010 20:31:07 -0500 Subject: [Freeswitch-users] setting switch_r_sdp with outbound event socket In-Reply-To: <4C7D8867.50109@xpirio.com> References: <4C7D8226.4080504@xpirio.com> <51496716-6E26-494D-8F41-B964BDB6675F@freeswitch.org> <4C7D8867.50109@xpirio.com> Message-ID: You have to urlencode it too and that does not guarentee it will work.... On Aug 31, 2010 6:01 PM, "Christian L?schenkohl" < christian.loeschenkohl at xpirio.com> wrote: thank you for your reply i read it here http://wiki.freeswitch.org/wiki/Codec_negotiation#Rewriting_SDP and it seemed logic to me (which isn't true as i seems). of course i tried it without CDATA - then the command on the loopback looks like this sendmsg call-command: execute execute-app-name: set execute-app-arg: switch_r_sdp=v=0 o=root 1051875459 1051875459 IN IP4 x.x.x.x s=call c=IN IP4 x.x.x.x t=0 0 m=audio 19156 RTP/AVP 0 9 99 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb... and i also do get [ERR] switch_ivr.c:469 Invalid Command! these lines print out the error (switch_ivr.c:469) if (zstr(cmd)) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Invalid Command!\n"); return SWITCH_STATUS_FALSE; } br On 2010-09-01 00:34, Brian West wrote: > Why are you trying to put it in the CDATA? You know you... -- Ing. Christian L?schenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunik... FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mail... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/b738ce44/attachment.html From adminjew at gmail.com Tue Aug 31 21:44:42 2010 From: adminjew at gmail.com (Yitzchok) Date: Wed, 1 Sep 2010 00:44:42 -0400 Subject: [Freeswitch-users] mod_managed on linux centos Message-ID: Hi, I finally got mod_manage to load fine in FreeSWITCH on centos but now when I copied a .net dll that I built and runs on windows to the /mod/managed folder it loads fine on freeswitch startup but when try calling it (it's a AppPlugin) from the dialplan using I get these errors. 2010-09-01 00:21:23.303236 [ERR] switch_cpp.cpp:1177 Exception in Run(NameOfClass): System.NullReferenceException: Object reference not set to an instance of an object Server stack trace: at System.Runtime.Serialization.SerializationCallbacks.GetMethodsByAttribute (System.Type type, System.Type attr) [0x00000] in :0 at System.Runtime.Serialization.SerializationCallbacks..ctor (System.Type type) [0x00000] in :0 at System.Runtime.Serialization.SerializationCallbacks.GetSerializationCallbacks (System.Type t) [0x00000] in :0 at System.Runtime.Serialization.SerializationObjectManager.RegisterObject (System.Object obj) [0x00000] in :0 at System.Runtime.Serialization.Formatters.Binary.ObjectWriter.GetObjectData (System.Object obj, System.Runtime.Serialization.Formatters.Binary.TypeMetadata& metadata, System.Object& data) [0x00000] in :0 at System.Runtime.Serialization.Formatters.Binary.ObjectWriter.WriteObject (System.IO.BinaryWriter writer, Int64 id, System.Object obj) [0x00000] in :0 at System.Runtime.Serialization.Formatters.Binary.ObjectWriter.WriteObjectInstance (System.IO.BinaryWriter writer, System.Object obj, Boolean isValueObject) [0x00000] in :0 at System.Runtime.Serialization.Formatters.Binary.ObjectWriter.WriteQueuedObjects (System.IO.BinaryWriter writer) [0x00000] in :0 at System.Runtime.Serialization.Formatters.Binary.ObjectWriter.WriteObjectGraph (System.IO.BinaryWriter writer, System.Object obj, System.Runtime.Remoting.Messaging.Header[] headers) [0x00000] in :0 at System.Runtime.Serialization.Formatters.Binary.BinaryFormatter.Serialize (System.IO.Stream serializationStream, System.Object graph, System.Runtime.Remoting.Messaging.Header[] headers) [0x00000] in :0 at System.Runtime.Serialization.Formatters.Binary.BinaryFormatter.Serialize (System.IO.Stream serializationStream, System.Object graph) [0x00000] in :0 at System.Runtime.Remoting.RemotingServices.SerializeExceptionData (System.Exception ex) [0x00000] in :0 Exception rethrown at [0]: at (wrapper xdomain-invoke) FreeSWITCH.AppPluginExecutor:Execute (string,intptr) at (wrapper remoting-invoke-with-check) FreeSWITCH.AppPluginExecutor:Execute (string,intptr) at FreeSWITCH.Loader.Run (System.String command, IntPtr sessionHandle) [0x00000] in :0 2010-09-01 00:21:23.303236 [ERR] mod_managed.cpp:422 Application run failed for NameOfClass (unknown module or exception). Anyone have any idea what is the problem? Thanks. Yitzchok -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100901/b4cf1cb0/attachment.html From quentusrex at gmail.com Tue Aug 31 14:39:25 2010 From: quentusrex at gmail.com (William King) Date: Tue, 31 Aug 2010 14:39:25 -0700 Subject: [Freeswitch-users] Ubuntu Hi-Res timer - small world In-Reply-To: References: <4B72146B-D099-4AAD-B0FF-F1B66403595C@freeswitch.org> <4C7D67F0.9010401@gmail.com> <20100831225800.3ab082f9@anubis.defcon1> <4C7D6E7C.8080702@gmail.com> Message-ID: <4C7D768D.7000901@gmail.com> Which release? On 08/31/2010 02:33 PM, lloyd thomas wrote: > 32 bit > > On 31 August 2010 22:32, lloyd thomas > wrote: > > I am running 2.6.28.10-freeswitch > > > On 31 August 2010 22:05, William King > wrote: > > I am running this kernel in production and seems to be very stable: > 2.6.32-24-server x86_64 GNU/Linux > > Using release 10.04.1 LTS > > -William King > > On 08/31/2010 01:58 PM, Jean-Yves F. Barbier wrote: > > On Tue, 31 Aug 2010 13:37:04 -0700, William King > > > > wrote: > > > > Yep, I confirm, 2.6.35 line seems to have timing PBs (and others), > > 2.6.33.4 is more stable (ubuntu often thinks newer is better...) > > > >> Which release of Ubuntu are you running? I have FS running on > a very old > >> machine(slower than yours) and I have seen an issue like this > before. It > >> was related to the release and the kernel. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From xduvox at gmail.com Tue Aug 31 21:01:34 2010 From: xduvox at gmail.com (Octavio Duarte) Date: Tue, 31 Aug 2010 23:01:34 -0500 Subject: [Freeswitch-users] how to register a gateway to freeswitch Message-ID: Hello everybody I have FreeSwitch running and i have done some call between extensions, i bought the book on amazon its a great book but its focus to make a pbx , i could not find an answer to my problem, i have a Quintum and 2 land lines connected to it. My quintum does not have a fixed IP, i wanna use my Quintum as a gateway to sent call to PSTN, but i cant beacause i dont know the ip address. i wanna know if is there a way to register the gateway to freeswitch? or maybe i can register the quintum as a endpoin but how can i sent the dialed number()destination_number) to that endpoint or user? so i either want to use mod distributor (simple round-robin) and Limit on it, beacuse all examples i have seen is for using with gateways! I have read mod sofia and almost the whole book but i can find any example for doing this! Thanks in advance for everyone in this project that is amazing! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/6f27cb6f/attachment.html