[asterisk-users] chan_h323 and menuselect dependencies problem
Hello List, I've been trying to compile Asterisk with H.323 support and, after correctly installing PTLib and H323plus (OpenH323), the Asterisk configure script still doesn't detect the dependencies as installed. I know they are correctly installed because after going into [asterisk-source-directory]/channels/h323 and issuing a 'make opt', it correctly builds everything: - root@slackbox:# make opt make DEBUG= default_target make[1]: Entering directory `/tmp/asterisk-test/slackbuilds/asterisk-SlackBuild/asterisk-1.4.39/channels/h323' [CC] ast_h323.cxx [CC] compat_h323.cxx [CC] cisco-h225.cxx [CC] caps_h323.cxx ar crv libchanh323.a ./ast_h323.o ./compat_h323.o ./cisco-h225.o ./caps_h323.o a - ./ast_h323.o a - ./compat_h323.o a - ./cisco-h225.o a - ./caps_h323.o make[1]: Leaving directory `/tmp/asterisk-test/slackbuilds/asterisk-SlackBuild/asterisk-1.4.39/channels/h323' Nevertheless, the menuselect application doesn't let me select chan_h323. Its important to note that if I manually edit menuselect.makedeps and menuselect.makeopts in order to manually set chan_h323 support, it does build chan_h323.o without problems (and install it, after make install), but, trying to do it via command line does not work: From Asterisk source dir: # make menuselect.makeopts # menuselect/menuselect --enable chan_h323 menuselect.makeopts a. Could this be some problem in the configure script? (where it look for dependencies?) b. What can I do in order to force Asterisk to compile chan_h323 in a less 'dirty' way than manually editing previously mentioned files? (I have verified that in this case, it will not yield any errors) Additional Info: Asterisk Verison: 1.4.39 Bash version : GNU bash, version 4.1.7(2)-release (i486-slackware-linux-gnu) OS: Slackware 13.1.0 PTLib : 2.8.3 H323Plus: 1.22.0 -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323 and ToS
Hi all, I'm using asterisk 1.4.26.2. I need to set TOS on H.323 channel. Does chan_h323.conf support tos (or tos_audio) statement, as well as sip.conf and iax.conf ? Thanks, Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323 requirements
Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 requirements
This can help (script for Debian): apt-get install flex bison #dirty hack to prevent error from missing file cd /usr/include/linux touch compiler.h #PWLIB cd /usr/src wget http://kent.dl.sourceforge.net/sourceforge/openh323/pwlib-v1_10_0-src-tar.gz tar zxvf pwlib-v1_10_0-src-tar.gz cd pwlib_v1_10_0/ ./configure make make install make opt PWLIBDIR=/usr/src/pwlib_v1_10_0 export PWLIBDIR #OpenH323 cd /usr/src wget http://ovh.dl.sourceforge.net/sourceforge/openh323/openh323-v1_18_0-src-tar.gz tar zxvf openh323-v1_18_0-src-tar.gz cd openh323_v1_18_0/ ./configure make make opt make install OPENH323DIR=/usr/src/openh323_v1_18_0/ export OPENH323DIR cd /usr/src/asterisk/channels/h323/ make make opt cd /usr/src/asterisk ./configure make make install echo /usr/local/lib /etc/ld.so.conf ldconfig #or similar way #cp /usr/local/lib/* /usr/lib Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, February 21, 2008 10:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_h323 requirements Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 requirements
Hello, To compile chan_h323 as is distributed you need to download OpenH323 v1.18.0 and PwLib v1.10.0 from: http://www.voxgratia.org Some months ago I had made a patch to compile the 1.4.x version and the trunk version (which evolved to 1.6.x) with H323+. Sadly, the patch was not included in the 1.6.x version which is being released soon. So, for the time being you need to use the above versions from Voxgratia. Best regards, Vlasis Hatzistavrou. Bruce McAlister wrote: Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 requirements
Hi, Thanks for the information, I will keep this for reference. Thanks Bruce Mindaugas Kezys wrote: This can help (script for Debian): apt-get install flex bison #dirty hack to prevent error from missing file cd /usr/include/linux touch compiler.h #PWLIB cd /usr/src wget http://kent.dl.sourceforge.net/sourceforge/openh323/pwlib-v1_10_0-src-tar.gz tar zxvf pwlib-v1_10_0-src-tar.gz cd pwlib_v1_10_0/ ./configure make make install make opt PWLIBDIR=/usr/src/pwlib_v1_10_0 export PWLIBDIR #OpenH323 cd /usr/src wget http://ovh.dl.sourceforge.net/sourceforge/openh323/openh323-v1_18_0-src-tar.gz tar zxvf openh323-v1_18_0-src-tar.gz cd openh323_v1_18_0/ ./configure make make opt make install OPENH323DIR=/usr/src/openh323_v1_18_0/ export OPENH323DIR cd /usr/src/asterisk/channels/h323/ make make opt cd /usr/src/asterisk ./configure make make install echo /usr/local/lib /etc/ld.so.conf ldconfig #or similar way #cp /usr/local/lib/* /usr/lib Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, February 21, 2008 10:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_h323 requirements Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 requirements
Hi, Thank you for the details of which versions to get. I will be building these two versions on Solaris to test chan_h323. Did your patch for building with OpenH323+ make it into the 1.4 edition of Asterisk? Thanks Bruce Vlasis Hatzistavrou (KTI) wrote: Hello, To compile chan_h323 as is distributed you need to download OpenH323 v1.18.0 and PwLib v1.10.0 from: http://www.voxgratia.org Some months ago I had made a patch to compile the 1.4.x version and the trunk version (which evolved to 1.6.x) with H323+. Sadly, the patch was not included in the 1.6.x version which is being released soon. So, for the time being you need to use the above versions from Voxgratia. Best regards, Vlasis Hatzistavrou. Bruce McAlister wrote: Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 requirements
Hello Bruce, Bruce McAlister wrote: Did your patch for building with OpenH323+ make it into the 1.4 edition of Asterisk? No, it didn't as it was considered a new feature and by Digium's policy new features can only be added in the trunk versions. The strange thing is that I added it in trunk version, too, but it didn't make it in the upcoming 1.6 version either. Best regards, Vlasis Hatzistavrou ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_h323 isn`t dropping calls comming with wrong codecs
I` using chan_h323 on my asterisk-1.4 to receive incomings calls. I need to set just two codecs to receive this call (g723 and g729), but I`m using disallow=all allow=g729 allow=g723.1 In h323.conf, but when I received a call using codec g711 for example, the call is answered, but doesn`t have audio. I made a test today using just disallow=all In h323.conf, but the call was answered too!! the log of this test: [Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2112 setup_incoming_call: Setting up incoming call for ip$189.0.24.69:4020/28391 -- Setting up Call -- CLICall token: [ip$189.0.24.69:4020/28391] -- CLICalling party name: [200] -- CLICalling party number: [200] -- CLICalled party name: [30144588] -- CLICalled party number: [30144588] -- CLICalling party IP: [189.0.24.69] [Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:1611 find_user: Could not find user by name 200 or address 189.0.24.69 [Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2177 setup_incoming_call: Sending [EMAIL PROTECTED] to context [ss7] extension 30144588 [Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2478 set_local_capabilities: Setting capabilities for connection ip$189.0.24.69:4020/28391 Setting capabilities to 0x0 (nothing) Capabilities in preference order is () DTMF mode is 1 Allowed Codecs for ip$189.0.24.69:4020/28391 (ip$201.7.99.242:1720): Zap/33-1 answered H323/ip$189.0.24.69:4020/28391 [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:666 oh323_answer: Answering on H323/ip$189.0.24.69:4020/28391 Answering call ip$189.0.24.69:4020/28391 Receiving RFC2833 on payload 101 Peer capability is G.711-uLaw-64k 1 Found peer capability G.711-uLaw-64k 1, Asterisk code is 4, frame size (in ms) is 20 Peer capability is G.711-ALaw-64k 2 Found peer capability G.711-ALaw-64k 2, Asterisk code is 8, frame size (in ms) is 20 Peer capabilities = 0xc (ulaw|alaw), ordered list is (ulaw|alaw) [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2448 set_peer_capabilities: Got remote capabilities from connection ip$189.0.24.69:4020/28391 [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2462 set_peer_capabilities: prefs[0]=ulaw:20 [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2462 set_peer_capabilities: prefs[1]=alaw:20 =-= In OnConnectionEstablished for call 28391 -- Connection Established with 200 [189.0.24.69] [Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2055 connection_made: Call ip$189.0.24.69:4020/28391 answered Some one knows why isn`t asterisk droping the call? Andre Luiz Martins ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323 build failure - `IPTOS_MINCOST' undeclared
Hi All, I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris 10. When I compile asterisk, the build fails at chan_h323 with: -- chan_h323.c: In function `reload_config': chan_h323.c:2863: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_h323.c:2863: error: (Each undeclared identifier is reported only once chan_h323.c:2863: error: for each function it appears in.) gmake[1]: *** [chan_h323.o] Error 1 gmake: *** [channels] Error 2 -- I have downloaded PWLIB v1.10.0 and OpenH323 v1.18.0 and they are both built and installed properly. Has anyone come across this issue, or do I have to log a bug report at Digiums bug tracker? Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323 and asterisk 1.2
If I let modules.conf autoload chan_h323.so then when I try to stop asterisk, it *does* stop (files in /var/run/asterisk/ are removed and connection via -vr from another console is not possible) but the asterisk process stays alive and stalled. In other words, a 'ps -ae | grep asterisk' show that the process is there after running 'stop now'. I either need to press CTRL-C from the *CLI or 'killall asterisk' from system console. *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). [infinite wait... user presses CTRL-C] Killed # However, if I specify not to load h323 then the asterisk process is cleanly terminated. # cat modules.conf | grep -i chan_h323 noload = chan_h323.so I'm using: PWlib 1.10.10 openh323 1.18.0 Asterisk 1.2.21.1 native h323 Is native h323 buggy in Asterisk 1.2.21.1? I tried ooh323 in Asterisk 1.2.21.1 and it doesn't seem to hang. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 compilation
Dear Kiven; Actually it is default and not degault. Also, I was doing the compilation remotely via the Putty. Another thing, I did another senario and got another thing, as below: I copied /usr/local/lib to /usr/lib and then I restarted asterisk, but when I come back to run it, then it was giving error that Segmentation error or Segmentation fail, actually I did not remeber it exactly, but was something related to segmentation. Then, I moved to the site where Asterisk existed and I decided to recompile h323 and then asterisk, when I run the make and make opt at the server it self and under the directory: /usr/src/asterisk-1.4/channels/h323, it was take the commands without error but does not give any text output (messages), I do not know if that good indication or not, then I compiled asterisk again, and it worked fine. Till now, I do not know why it was giving me default and does not know how to know if chan_h323 is really working or not, how can I test? Is it by establishing h323 trunk? Regards Bilal cd /usr/src/asterisk-1.4/channels/h323 When I type make, it gives me: make: Nothing to be done for 'degault' This is *exactly* what showed up on your session? The word 'degault' does not appear in the Makefile at all, so if that is the message that you got then your source tree is corrupted. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323 compilation
Hi All; I am trying now to compile h323 to be able to use it, I did the pwlib and openh323 successfully and I exported the PWLIBDIR=/usr/src/pwlib_v1_10_0 and the OPENH323DIR=/usr/src/openh323_v1_18_0, then I was need to compile h323 as following: cd /usr/src/asterisk-1.4/channels/h323 When I type make, it gives me: make: Nothing to be done for 'degault' And when I type make opt, it takes it but I do not see any sentences as output, just it takes it without and messages. Is it going fine? Why that is happening? How can I know that I really did an h323 installation for chan_h323 and I can use it? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 compilation
bilal ghayyad wrote: cd /usr/src/asterisk-1.4/channels/h323 When I type make, it gives me: make: Nothing to be done for 'degault' This is *exactly* what showed up on your session? The word 'degault' does not appear in the Makefile at all, so if that is the message that you got then your source tree is corrupted. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323.so Asterisk Beta compilation
I have had some interesting compiling results with the latest beta release of Asterisk. With reference to this channel After running the make opt in the H323 directory, and the make install in the Asterisk directory, there is still no chan_h323.so file Created.. Are there any other args or commands that need to be set to get this to work? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323.so Asterisk Beta compilation
On Mon, Oct 23, 2006 at 04:09:43PM +0200, Patrick wrote: I have had some interesting compiling results with the latest beta release of Asterisk.. With reference to this channel. After running the make opt in the H323 directory, and the make install in the Asterisk directory, there is still no chan_h323.so file Created.. Are there any other args or commands that need to be set to get this to work? For the module to be built you need autoconf to detect your version of openh323 (and pwlib), and to have that module selected. When you run 'menuselect', and enter the channels section, do you see the module chan_h323: selected, unselected, or XXX-ed out? -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] chan_h323.so Asterisk Beta compilation
When I have a look and the menuselect.makeopts file.. MENUSELECT_CHANNELS= chan_gtalk chan_h323 ... is there, and also during the ./configure, all the various pwlib and openh323 version checks seem valid.. but still not sure where you enable the channel to be built... According to the H323 README, it just says make opt, then go to the asterisk directory, and make install, but that still has no effect because again the actual chan_h323.so file is not built... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 23 October 2006 04:29 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] chan_h323.so Asterisk Beta compilation On Mon, Oct 23, 2006 at 04:09:43PM +0200, Patrick wrote: I have had some interesting compiling results with the latest beta release of Asterisk.. With reference to this channel. After running the make opt in the H323 directory, and the make install in the Asterisk directory, there is still no chan_h323.so file Created.. Are there any other args or commands that need to be set to get this to work? For the module to be built you need autoconf to detect your version of openh323 (and pwlib), and to have that module selected. When you run 'menuselect', and enter the channels section, do you see the module chan_h323: selected, unselected, or XXX-ed out? -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 problem
Ganbold Tsagaankhuu wrote: Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: -- X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN boldsoft*CLI show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by [EMAIL PROTECTED] on a i686 running Linux I can make H323 call without any problem from X-Pro and from X-lite dead-air both end. My default h323.conf === [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=g723 allow=alaw allow=ulaw allow=gsm gatekeeper = a.b.c.d AllowGKRouted = yes noFastStart = no noH245Tunneling = no noSilenceSuppression = no Modified h323.conf == [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=g723 allow=alaw allow=ulaw allow=gsm gatekeeper = a.b.c.d AllowGKRouted = yes noFastStart = yes noH245Tunneling = yes noSilenceSuppression = no I can to hear one-way audio from X-lite side, but no audio from PSTN side I did upgrade Asterisk to Asterisk-1.2.5 and chan_h323 doesn't work default and even modified config. Any suggestion? Which H323 channel module is better (chan_h323, oh323, ooh323)? I downloaded ooh323c 0.8.1 and don't know how to create asterisk module using source? Regards, Ganbold ___ Hi, I had this same problem, play aroud with noFastStart = yes or noFastStart = no noH245Tunneling = yes or noH245Tunneling = no yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 problem
I tried many different combination of nofaststart, noh245tunneling and no success. Balgaa - Original Message - From: yusuf [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 24, 2006 4:27 PM Subject: Re: [Asterisk-Users] chan_h323 problem Ganbold Tsagaankhuu wrote: Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: -- X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN boldsoft*CLI show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by [EMAIL PROTECTED] on a i686 running Linux I can make H323 call without any problem from X-Pro and from X-lite dead-air both end. My default h323.conf === [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=g723 allow=alaw allow=ulaw allow=gsm gatekeeper = a.b.c.d AllowGKRouted = yes noFastStart = no noH245Tunneling = no noSilenceSuppression = no Modified h323.conf == [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=g723 allow=alaw allow=ulaw allow=gsm gatekeeper = a.b.c.d AllowGKRouted = yes noFastStart = yes noH245Tunneling = yes noSilenceSuppression = no I can to hear one-way audio from X-lite side, but no audio from PSTN side I did upgrade Asterisk to Asterisk-1.2.5 and chan_h323 doesn't work default and even modified config. Any suggestion? Which H323 channel module is better (chan_h323, oh323, ooh323)? I downloaded ooh323c 0.8.1 and don't know how to create asterisk module using source? Regards, Ganbold ___ Hi, I had this same problem, play aroud with noFastStart = yes or noFastStart = no noH245Tunneling = yes or noH245Tunneling = no yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 problem
Hello, I installed Asterisk fromCVSon Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecstoo. My network connection diagram: -- X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN boldsoft*CLI show versionAsterisk CVS-v1-0-03/24/06-15:27:08 built by [EMAIL PROTECTED] on a i686 running Linux I can make H323 call without any problem from X-Pro and from X-lite dead-air both end. My default h323.conf === [general]port = 1720bindaddr = 0.0.0.0 disallow=allallow=g729allow=g723allow=alawallow=ulawallow=gsm gatekeeper = a.b.c.dAllowGKRouted = yesnoFastStart = nonoH245Tunneling = nonoSilenceSuppression = no Modifiedh323.conf == [general]port = 1720bindaddr = 0.0.0.0 disallow=allallow=g729allow=g723allow=alawallow=ulawallow=gsm gatekeeper = a.b.c.dAllowGKRouted = yesnoFastStart = yesnoH245Tunneling = yesnoSilenceSuppression = no I can to hear one-way audio from X-lite side, but no audio from PSTN side I did upgrade Asterisk to Asterisk-1.2.5 and chan_h323 doesn't work defaultand even modified config. Any suggestion? Which H323 channel module is better (chan_h323, oh323, ooh323)? I downloaded ooh323c 0.8.1 and don't know how to create asterisk module using source? Regards, Balgaa ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 problem
Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: -- X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN boldsoft*CLI show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by [EMAIL PROTECTED] on a i686 running Linux I can make H323 call without any problem from X-Pro and from X-lite dead-air both end. My default h323.conf === [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=g723 allow=alaw allow=ulaw allow=gsm gatekeeper = a.b.c.d AllowGKRouted = yes noFastStart = no noH245Tunneling = no noSilenceSuppression = no Modified h323.conf == [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=g723 allow=alaw allow=ulaw allow=gsm gatekeeper = a.b.c.d AllowGKRouted = yes noFastStart = yes noH245Tunneling = yes noSilenceSuppression = no I can to hear one-way audio from X-lite side, but no audio from PSTN side I did upgrade Asterisk to Asterisk-1.2.5 and chan_h323 doesn't work default and even modified config. Any suggestion? Which H323 channel module is better (chan_h323, oh323, ooh323)? I downloaded ooh323c 0.8.1 and don't know how to create asterisk module using source? Regards, Ganbold ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 passes no audio?
I'm attempting to get chan_h323 working on Asterisk CVS-STABLE.
I've compiled it ok using the Janus release of pwlib/openh323, by
editing the makefile as per the comments.
Call setup and cleardown seems to work fine, but no audio is being
passed in either direction.
Doing an h.323 trace 9, I noticed the following sequence at the end
of the call setup:
h323.cxx(1685) H225Handling PDU: Connect callRef=13295
h323.cxx(1925) H225Set protocol version to 4
h323.cxx(2152) H225Set remote party name: root [192.168.0.20]
h323.cxx(2160) H225Set remote application name: The NuFone Network's
H.323 Channel Driver for Asterisk1.0.0 (OpenH323 v1.13.5)9/61
h323ep.cxx(1992) H225Received connect PDU.
h323.cxx(3205) H245Started control channel
h323neg.cxx(540) H245TerminalCapabilitySet already sent.
h323.cxx(3962) H323InternalEstablishedConnectionCheck:
connectionState=HasExecutedSignalConnect fastStartState=FastStartDisabled
h323.cxx(4019) H245Default OnSelectLogicalChannels, FastStartDisabled
h323caps.cxx(1814) H323FindCapability: G.711-ALaw-64k{sw} 1
h323caps.cxx(1818) H323Found capability: G.711-ALaw-64k{sw} 1
h323.cxx(4066) H323Selecting G.711-ALaw-64k{sw} 1
h323neg.cxx(743) H245Opening channel: T-101
rtp.cxx(1605) RTP_UDP Session 1 created: 192.168.0.1:5000-5001
ssrc=2079793508
rtp.cxx(1398) RTP Adding session RTP_UDP
channels.cxx(1048) H323RTP Transmitter created using session 1
channels.cxx(893) H323RTP OnSendingPDU
h323rtp.cxx(193) RTP OnSendingPDU
codecs.cxx(1271) Codec G711 ALaw encoder created for at 64k, 240 samples
h323ep.cxx(2143) Codec Could not open sound channel /dev/dsp for
recording:
channels.cxx(1096) LogChan Transmit thread aborted (open fail) for
G.711-ALaw-64k{sw} 1
h323.cxx(4072) H323OnSelectLogicalChannels, OpenLogicalChannel failed:
G.711-ALaw-64k{sw} 1
Why does it want to open /dev/dsp? I'm not trying to play or record using
a local sound device, I just want to pass digital audio between Asterisk
and the network.
The second machine that I am testing this with, trying to get them to talk
to each other, doesn't have any sound hardware, and the third message from
last reads: Could not open sound channel for recording:
I assume it is these errors that are preventing audio from being passed.
Can anyone suggest how I might overcome this to make chan_h323 work?
Thanks
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] chan_h323 vs chan_oh323 chan_ooh323
Please post ur installation script for chan_h323 - Original Message - From: Atif Rasheed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, June 20, 2005 7:21 AM Subject: [Asterisk-Users] chan_h323 vs chan_oh323 chan_ooh323 hello there, can somebody please comment which one of these channel drivers will give best output doing g729|g723 pass-thru. only pass-thru is needed no transcoding. please share your experience. if somebody has some figures (simultanous calls using a certain channel driver) it will be apericiated. I have installed chan_h323 (by McNamara) and its working fine with me. I just want to know if I run this driver on a Dual-Xeon machine. can it handle 500 or 500 simultanous calls in pass-thru mode. Regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 vs chan_oh323 chan_ooh323
hello there, can somebody please comment which one of these channel drivers will give best output doing g729|g723 pass-thru. only pass-thru is needed no transcoding. please share your experience. if somebody has some figures (simultanous calls using a certain channel driver) it will be apericiated. I have installed chan_h323 (by McNamara) and its working fine with me. I just want to know if I run this driver on a Dual-Xeon machine. can it handle 500 or 500 simultanous calls in pass-thru mode. Regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 context
Hi all, All incoming H.323 calls on chan_h323 were forwarded to default context but not detroit. It seems context=detroit is not effective. Any helps??? [det-gw] type=h323 prefix=1248,1313 context=detroit Thanks. IM ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323
Hi I've installed successfully: - PWlib v1.6.7 library -Openh323 v1.13.5 library -asterisk-oh323 v0.6.5 and so the modules chan_oh323 is installed successfully Now I try to install chan_h323 First question: is this necessary? I edit the Makefile in the directory /usr/src/asterisk-1.0.7/channels/h323 to point to the right includes directories I do makeand I've the following error: make:***[ast_h323.o] Error 1 Have you some suggestions? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323
[EMAIL PROTECTED] a écrit : Hi I've installed successfully: - PWlib v1.6.7 library -Openh323 v1.13.5 library -asterisk-oh323 v0.6.5 and so the modules chan_oh323 is installed successfully Now I try to install chan_h323 First question: is this necessary? No, it's or oh323 or h323. I suggest you to stay with oh323 -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 codecs
Hi Chetan, Per my understanding to the chan_h323 operation, if no codecs are loaded, Asterisk will perform a pass-through function, which means that signaling is passed via Asterisk, but RTP passes between the endpoints. This is NOT proxy function, as with proxy function means that RTP passes via the Asterisk box, which means you require the codecs modules loaded. Now, I hope that JerJer will correct me if I'm wrong, but I believe that if you load a particular codec module, that codec will be proxied while if a codec module is not loaded, that module will be pass-through only. Regards, Nir S - Original Message - From: Chetan Sarva [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 11:08 PM Subject: [Asterisk-Users] chan_h323 codecs Hi, Can anyone confirm that if I want to do h323 proxying that I do not need codecs installed? For example if the codec being used is g723.1, I don't need the codec installed locally because there is no compression or decompression being done on my server; the incoming traffic is simply being sent out on another h323 channel (h323 in-h323 out). Is this correct? Thanks, Chetan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 codecs
Hi, Can anyone confirm that if I want to do h323 proxying that I do not need codecs installed? For example if the codec being used is g723.1, I don't need the codec installed locally because there is no compression or decompression being done on my server; the incoming traffic is simply being sent out on another h323 channel (h323 in-h323 out). Is this correct? Thanks, Chetan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 on AMD64
Ok, I have some more info. The code from openh323.org will not compile on x86_64 but the latest from the OpenH323 project on sourceforge will compile just fine on x86_64. Asterisk 1.0 will not compile with this new openh323 code but it looks like the latest cvs-head does. In channels/chan_h323 I had to make a few tweaks to get it to compile since chan_h323 assumes x86. First, I had to change the makefile from: CFLAGS += -march=$(shell uname -m) To: CFLAGS += -march=athlon64 to get rid of this error: ast_h323.cpp:1: error: bad value (x86_64) for -march= switch ast_h323.cpp:1: error: bad value (x86_64) for -mtune= switch Then I had to make some symlinks: ln -s /usr/local/src/openh323/lib/libh323_linux_x86_64_r.so /usr/local/src/openh323/lib/libh323_linux_x86_r.so ln -s /usr/local/src/pwlib/lib/libpt_linux_x86_64_r.so /usr/local/src/pwlib/lib/libpt_linux_x86_r.so Well, that's got me compiling cleanly at least. However asterisk still has problems. When I try to make an H323 call to the box the rtp stream never gets connected. tcpdump shows the clients packets hitting the asterisk box but asterisk never transmits anything. Another odd thing is that stop now can no longer be used to shut down asterisk after an attempt is made at making an h323 call but it works just fine as long as you don't try making an h323 call. I suspect a problem in chan_h323 locking things up. If I do a show channels after having tried to make a call I get: bit64*CLI show channels Channel (ContextExtensionPri ) State Appl. Data SIP/pstn1-e7c5 (default 1 ) Up Bridged Call H323/ip$[myip]:30005/28852 1 active channel(s) Nov 24 02:51:14 WARNING[10527]: channel.c:494 ast_channel_walk_locked: Avoided deadlock for 'H323/ip$[myip]:30005/28852', 10 retries! Anyone know of any major signalling or socket differences between x86 and x86_64? -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpyUAcBGQapA.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 on AMD64
Tracy R Reed a écrit : [...] bit64*CLI show channels Channel (ContextExtensionPri ) State Appl. Data SIP/pstn1-e7c5 (default 1 ) Up Bridged Call H323/ip$[myip]:30005/28852 1 active channel(s) Nov 24 02:51:14 WARNING[10527]: channel.c:494 ast_channel_walk_locked: Avoided deadlock for 'H323/ip$[myip]:30005/28852', 10 retries! I repeat: I have the same problem, same message on my Dual Celeron. So it's *not* a AMD64 problem. I just have to call from * to GnuGK (or reverse), it's ringing, I hangup, no audio. I wait 30 sec and my H323 EP (GM) hangup with abnormal end of call. Debug mode shows nothing. 90 s after, I see Sending RTP 'US' 0.0.0.0:7136 (4 times) Device 'H323/ip$192.168.10.250:30004/11671' changed to state '2' in * debug logs. Then, around 4 mn after Avoiding deadlock for 'H323/ip$192.168.10.250:30004/11671' (10 times) in the mean time, on console, I start to have the same message that you. And this each 5mn, will never stop till I restart * -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 on AMD64
Tracy R Reed a écrit : Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic with tcpdump I see the * box receiving my UDP packets but never transmitting anything. SIP works just fine and is talking to my TNT. I am trying to receive incoming H323 to terminate to the call on the TNT. Same problem here. My * box is connected to GnuGK. CVS Head 11/02/04, kernel 2.4.26-SMP From openh323 version.h: #define MAJOR_VERSION 1 #define MINOR_VERSION 15 From pwlib version.h: #define MAJOR_VERSION 1 #define MINOR_VERSION 8 Same versions. Before, I was running pwlib 1.5 and openh323 1.12 and had no problem. -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 on AMD64
On Tue, Nov 23, 2004 at 12:42:07PM +0100, administrator tootai spake thusly: with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic with tcpdump I see the * box receiving my Same problem here. My * box is connected to GnuGK. CVS Head 11/02/04, kernel 2.4.26-SMP Same problem as in you ran tcpdump or something and saw the odd behavior of receiving but not sending any packets? VERY interesting. Were you on an x86-64 bit box or regular x86? I was thinking this odd behavior was some odd interation with x86-64. Same versions. Before, I was running pwlib 1.5 and openh323 1.12 and had no problem. Perhaps I should switch to that version and see how it goes then. I was afraid my problem was related to the platform. I am really hoping H323 stabilizes on Asterisk. It's a shame that it is such a pain in the neck add-on when it is still really the backbone of VOIP. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpjLbRYmggEL.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 on AMD64
Tracy R Reed wrote: On Tue, Nov 23, 2004 at 12:42:07PM +0100, administrator tootai spake thusly: with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic with tcpdump I see the * box receiving my Same problem here. My * box is connected to GnuGK. CVS Head 11/02/04, kernel 2.4.26-SMP Same problem as in you ran tcpdump or something and saw the odd behavior of receiving but not sending any packets? VERY interesting. Were you on an x86-64 bit box or regular x86? I was thinking this odd behavior was some odd interation with x86-64. Same versions. Before, I was running pwlib 1.5 and openh323 1.12 and had no problem. Perhaps I should switch to that version and see how it goes then. I was afraid my problem was related to the platform. I am really hoping H323 stabilizes on Asterisk. It's a shame that it is such a pain in the neck add-on when it is still really the backbone of VOIP. Have you tried asterisk-oh323? Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 on AMD64
Michael Manousos a écrit : Tracy R Reed wrote: [...] Same problem as in you ran tcpdump or something and saw the odd behavior of receiving but not sending any packets? VERY interesting. Were you on an x86-64 bit box or regular x86? I was thinking this odd behavior was some odd interation with x86-64. x86 dual Celeron 400 I didn't check with tcpdump [...] Perhaps I should switch to that version and see how it goes then. I was afraid my problem was related to the platform. Yes, but they aren't working with CVS (from Nov sure), stable (1.0.2) should be ok [...] Have you tried asterisk-oh323? No. I perhaps should ;-) -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 on AMD64
Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic with tcpdump I see the * box receiving my UDP packets but never transmitting anything. SIP works just fine and is talking to my TNT. I am trying to receive incoming H323 to terminate to the call on the TNT. Linux mybox 2.6.9-1.667 #1 Tue Nov 2 14:50:10 EST 2004 x86_64 x86_64 x86_64 GNU/Linux Asterisk CVS-HEAD-11/17/04-21:47:41 built by [EMAIL PROTECTED] on a x86_64 running Linux From openh323 version.h: #define MAJOR_VERSION 1 #define MINOR_VERSION 15 From pwlib version.h: #define MAJOR_VERSION 1 #define MINOR_VERSION 8 -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgppgxLK3mEnN.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323: forcing 20ms packetisation
Hi All Is there a better mailing list where I should ask these questions ? Thanks Mike O'Connor wrote: Hi all I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant. Our supplier of call termination requires h323 the following: * The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99. I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling. But the 20ms packetisation has me beat. I have made a test call to the provider which did not work becase I was sending 30ms voice packets. SO my question does any one know now to force the correct voice packet size ? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323: forcing 20ms packetisation
Is this mike oconnor as in the Australian mick oconnor -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike O'Connor Sent: Wednesday, 20 October 2004 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] chan_h323: forcing 20ms packetisation Hi All Is there a better mailing list where I should ask these questions ? Thanks Mike O'Connor wrote: Hi all I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant. Our supplier of call termination requires h323 the following: * The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99. I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling. But the 20ms packetisation has me beat. I have made a test call to the provider which did not work becase I was sending 30ms voice packets. SO my question does any one know now to force the correct voice packet size ? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323: forcing 20ms packetisation
HI Mike, You wouldn't be trying to connect to Comindico in Australia by any chance? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike O'Connor Sent: Monday, 18 October 2004 02:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_h323: forcing 20ms packetisation Hi all I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant. Our supplier of call termination requires h323 the following: * The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99. I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling. But the 20ms packetisation has me beat. I have made a test call to the provider which did not work becase I was sending 30ms voice packets. SO my question does any one know now to force the correct voice packet size ? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323: forcing 20ms packetisation
Hi all I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant. Our supplier of call termination requires h323 the following: * The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99. I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling. But the 20ms packetisation has me beat. I have made a test call to the provider which did not work becase I was sending 30ms voice packets. SO my question does any one know now to force the correct voice packet size ? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323: remote ip address - context
Hi,
I'm looking for a mean in chan_h323 to jump to a specific
context dependent on the remote ip address.
E.g. an argument, let's tell it ignore_h323_name, in h323.conf
users like this:
[BillyBob]
ignore_h323_name=yes
type=user
host=1.2.3.4
context=path1
in a way, every incoming call from ip 1.2.3.4
will fit this user, not only when the H323-name is BillyBob.
Or a variable like chan_oh323's one: ${OH323_RADDR}
which can be used in the gotoIf statement in extensions.conf.
Or any other mean, which allows me to lead a call into a certain
context just dependent on the remote ip address.
Thanks for any hints!
Roger.
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[Asterisk-Users] chan_h323 doesn't pass audio before call is answered
Hi, I have the following topology: PSTN/H323 gateway-GNUGK-chan_h323/chan_sip-SIP EP Mostly everything works fine except chan_h323 is not passing audio from PSTN before the call is answered and as a result users can't hear PSTN announcements (like the number is not in service) that's played on unanswered call. All they hear is just continuous ringtone as though remote phone is ringing but noone answering. I confirmed that gateway (Lucent MAX TNT) and GNUGK is not a problem because I can hear those announcements on another h323 endpoint registered directly with GNUGK. Also it did work with previous versions of asterisk (up to 0.9.1) with some 3d-party patch downloaded at some point from bugtracker. Now the patch cannot be correctly applied to asterisk-1.0-RC2 and my attempts to manually apply it failed. Therefore I would like to request some help on solving this issue. Does it work for anybody? May be with different PSTN gateway? Do you have any suggestions on fixing it? In addition as we are considering to get some Cisco PSTN/SIP gateway, I would like to ask if it's also the issue with SIP driver? Thanks a lot, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CHAN_H323 bridge SIP no audio
I tried a lot of times to get it worked, but I cant obtain audio using SIP-chan_h323 or chan_h323-SIP I tried disbling FastStart without good results... What's the problem? I need to do BRIDGE between SIP and H.323!! help!! Sebastian.-
RE: [Asterisk-Users] chan_h323 no audio both ways
Sorry, Tom, I missed this message when it came through. It seems this problem is a continuing issue among the asterisk folk. Tell me, what versions of IOS have you tested with, do you have any of the h323 options enable/disabled in the 5300? -g On Fri, 2004-06-18 at 21:09, T. Chan wrote: Hi Glen, I have had the same problem for quite awhile, since around February, all cvs codes that I have tried, and with h323, I have been getting no audio. I am forced to stay with mid-Jan version of the cvs because of this. I tried using ulaw, g729, but same results, I have in a few occasions dropped a few lines here to ask for advice, but no response, may be we could try to exchange some ideas. Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, June 14, 2004 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_h323 no audio both ways I've compiled chan_h323 with the latest cvs code, but my calls don't pass audio. The call connects just fine, as there are no errors reported on either side, nor in a traffic examination with ethereal. I've tried the following: voip phone - asterisk - asterisk - voip phone voip phone - asterisk - asterisk zap - asterisk - asterisk zap - asterisk - cisco cisco - asterisk I'm using ulaw on all connections. Any clues, ideas, or directions would be appreciated. Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323 no audio both ways
Hi Glen, I have had the same problem for quite awhile, since around February, all cvs codes that I have tried, and with h323, I have been getting no audio. I am forced to stay with mid-Jan version of the cvs because of this. I tried using ulaw, g729, but same results, I have in a few occasions dropped a few lines here to ask for advice, but no response, may be we could try to exchange some ideas. Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, June 14, 2004 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_h323 no audio both ways I've compiled chan_h323 with the latest cvs code, but my calls don't pass audio. The call connects just fine, as there are no errors reported on either side, nor in a traffic examination with ethereal. I've tried the following: voip phone - asterisk - asterisk - voip phone voip phone - asterisk - asterisk zap - asterisk - asterisk zap - asterisk - cisco cisco - asterisk I'm using ulaw on all connections. Any clues, ideas, or directions would be appreciated. Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 no audio both ways
I've compiled chan_h323 with the latest cvs code, but my calls don't pass audio. The call connects just fine, as there are no errors reported on either side, nor in a traffic examination with ethereal. I've tried the following: voip phone - asterisk - asterisk - voip phone voip phone - asterisk - asterisk zap - asterisk - asterisk zap - asterisk - cisco cisco - asterisk I'm using ulaw on all connections. Any clues, ideas, or directions would be appreciated. Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 chan_oh323
Hello, Has anyone gotten chan_h323 and chan_oh323 to run on the same system at the same time? Provided you change the listening ports of course. I can get both of them to start, but whenever I try to make a call using chan_h323 I get a segmentation fault. This doesn't happen if I disable chan_oh323. With them both running I can make a call using chan_oh323 with no errors. Very strange. Thanks -Matt
[Asterisk-Users] Chan_h323 docs
Title: Chan_h323 docs Jeremy, In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution. Could you post those docs in your download directory? I'm trying to understand the nuances of your driver, gnugk, and extensions. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] Chan_h323 gnugk
Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h323.conf like: [office] type=h323 prefix=9 context=outbound I get a message saying: WARNING[1074403072]: File chan_h323.c, Line 215 (build_alias): Keyword h323 does not make sense in type=h323 Why is that? 3) When making a call from an h323 client such as ohPhone registered with gnugk, I can make the call, but it uses the context from the [general] section, rather than the context in [office]. Is this supposed to be? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 readme file
Hi Brian, I will first use the listed versions as you suggested. It is great if the readme file is corrected. Instead of You first need the latest release versions of PWLib and Open H.323 from ..., it should read You first need the LISTED release versions of PWLib and Open H.323 from Then poor newbies do not need to get RTFM'ed :). Also please take a look at this posting; http://lists.digium.com/pipermail/asterisk-cvs/2003-November/000498.html So there were changes in mid Novemeber time frame. Most notably the referance to G729 has been taken away. A big requirement to run Open h323 v1.11.7 was that the it was the last version with G729. So chan_h323 must have been modified that it can now use G729 (if license is bought from Digium) without depending on open h323. Please correct me if I am wrong. Cheers SW Date: Mon, 8 Dec 2003 22:59:05 -0600 (CST) From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Digium. Com [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_h323 readme file Reply-To: [EMAIL PROTECTED] If you have problems use the latest ones. Otherwise use whats listed because thats what JJ has tested and is known to work. bkw On Mon, 8 Dec 2003, SW wrote: Hello I am getting ready to install chan_h323. Just updated my * with the latest code from CVS (12/08/03). I was reading the Readme file and confused. Quoted from the README NOTICE: Whatever you do, DO NOT USE distrubution specific installs of Open H.323 and PWLib. In fact you should check to make sure your distro didn't install them for you without your knowledge. Check everything out of CVS. If you dont know how to deal with cvs, learn. Also, if you are not using the listed versions of Open H.323 or PWlib you are on your own, sorry. To compile this code: You first need the latest release versions of PWLib and Open H.323 from http://www.openh323.org/. Make sure you follow the build instructions EXCPLICTLY. Unquote First para says that I should only use the listed versions of PWlib and open h.323 (v1.4.11 and v1.11.7) Then it says that I should use the latest release versions of PWLib and Open H.323 (v1.5.2 and v1.12.2.) I've read Wi-Ki and many other places that Chan_H323 use open h323 v1.11.7, but wondering whether JJ has made some changes lately :) What should be the correct way ??? Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 readme file
Hello I am getting ready to install chan_h323. Just updated my * with the latest code from CVS (12/08/03). I was reading the Readme file and confused. Quoted from the README NOTICE: Whatever you do, DO NOT USE distrubution specific installs of Open H.323 and PWLib. In fact you should check to make sure your distro didn't install them for you without your knowledge. Check everything out of CVS. If you dont know how to deal with cvs, learn. Also, if you are not using the listed versions of Open H.323 or PWlib you are on your own, sorry. To compile this code: You first need the latest release versions of PWLib and Open H.323 from http://www.openh323.org/. Make sure you follow the build instructions EXCPLICTLY. Unquote First para says that I should only use the listed versions of PWlib and open h.323 (v1.4.11 and v1.11.7) Then it says that I should use the latest release versions of PWLib and Open H.323 (v1.5.2 and v1.12.2.) I've read Wi-Ki and many other places that Chan_H323 use open h323 v1.11.7, but wondering whether JJ has made some changes lately :) What should be the correct way ??? Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 readme file
If you have problems use the latest ones. Otherwise use whats listed because thats what JJ has tested and is known to work. bkw On Mon, 8 Dec 2003, SW wrote: Hello I am getting ready to install chan_h323. Just updated my * with the latest code from CVS (12/08/03). I was reading the Readme file and confused. Quoted from the README NOTICE: Whatever you do, DO NOT USE distrubution specific installs of Open H.323 and PWLib. In fact you should check to make sure your distro didn't install them for you without your knowledge. Check everything out of CVS. If you dont know how to deal with cvs, learn. Also, if you are not using the listed versions of Open H.323 or PWlib you are on your own, sorry. To compile this code: You first need the latest release versions of PWLib and Open H.323 from http://www.openh323.org/. Make sure you follow the build instructions EXCPLICTLY. Unquote First para says that I should only use the listed versions of PWlib and open h.323 (v1.4.11 and v1.11.7) Then it says that I should use the latest release versions of PWLib and Open H.323 (v1.5.2 and v1.12.2.) I've read Wi-Ki and many other places that Chan_H323 use open h323 v1.11.7, but wondering whether JJ has made some changes lately :) What should be the correct way ??? Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-Digium Partner
If anybody still have any G729 handshake problem with Asterisk and other non-Digium partner, I *really* recommend to use this patch: http://bugs.digium.com/bug_view_page.php?bug_id=421 6 monhts passed and finally my problem seems to be solved. Thanks Adam! Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 - Segmentation fault (core dumped)
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790
#1 0x41f8879c in create_connection (call_reference=1349809548) at
chan_h323.c:928
#2 0x41f8f34b in
MyH323Connection::CreateRealTimeLogicalChannel(H323Capability const,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) (this=0x8178758, [EMAIL PROTECTED], dir=IsTransmitter,
sessionID=1)
at ast_h323.cpp:626
#3 0x49470170 in H323RealTimeCapability::CreateChannel(H323Connection,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) const () from /root/openh323/lib/libh323_linux_x86_r.so.1
#4 0x4946071d in H245NegLogicalChannel::OpenWhileLocked(H323Capability
const, unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#5 0x494604e6 in H245NegLogicalChannel::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#6 0x49462423 in H245NegLogicalChannels::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#7 0x4944d311 in H323Connection::OpenLogicalChannel(H323Capability const,
unsigned, H323Channel::Directions) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#8 0x4944cf98 in H323Connection::SelectDefaultLogicalChannel(unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#9 0x4944c9d2 in H323Connection::OnSelectLogicalChannels() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#10 0x4944c8b1 in H323Connection::InternalEstablishedConnectionCheck() ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#11 0x4944a6d1 in H323Connection::HandleControlData(PPER_Stream) () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#12 0x4944a28c in H323Connection::HandleControlChannel() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#13 0x494992ee in H245TransportThread::Main() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#14 0x48d33177 in PThread::PX_ThreadStart(void*) () from
/root/pwlib/lib/libpt_linux_x86_r.so.1
#15 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0
And this is the console log:
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [Gustavo]
-- Calling party number: [1152880056]
-- Called party name: [0111553037260]
-- Called party number: [0111553037260]
e164: [0111553037263]
-- Executing Dial(H323/ip$10.60.144.14:1240/4096,
Zap/1/0111553037260) in new stack
-- Called 1/0111553037260
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
=*= In CreateRealTimeLogicalChannel for call 4096
-- externalIpAddress: 172.16.254.107
-- externalPort: 13488
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-ALaw-64k{sw}
-- channelsOpen = 2
-- remoteIpAddress: 0.0.0.0
-- remotePort: 0
-- ExternalIpAddress: 172.16.254.107
-- ExternalPort: 13488
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Call with ended abnormally
== H.323 Connection deleted.
channelsOpen = 1
-- Closing logical channel...
channelsOpen = 0
Segmentation fault (core dumped)
[EMAIL PROTECTED] asterisk]#
What is wrong?
Thanks in advance,
Gus
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Re: [Asterisk-Users] chan_h323 - Segmentation fault (core dumped)
Are you using the recommended pwlib and openh323 tarballs?
bkw
On Mon, 13 Oct 2003, CW_ASN wrote:
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790
#1 0x41f8879c in create_connection (call_reference=1349809548) at
chan_h323.c:928
#2 0x41f8f34b in
MyH323Connection::CreateRealTimeLogicalChannel(H323Capability const,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) (this=0x8178758, [EMAIL PROTECTED], dir=IsTransmitter,
sessionID=1)
at ast_h323.cpp:626
#3 0x49470170 in H323RealTimeCapability::CreateChannel(H323Connection,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) const () from /root/openh323/lib/libh323_linux_x86_r.so.1
#4 0x4946071d in H245NegLogicalChannel::OpenWhileLocked(H323Capability
const, unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#5 0x494604e6 in H245NegLogicalChannel::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#6 0x49462423 in H245NegLogicalChannels::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#7 0x4944d311 in H323Connection::OpenLogicalChannel(H323Capability const,
unsigned, H323Channel::Directions) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#8 0x4944cf98 in H323Connection::SelectDefaultLogicalChannel(unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#9 0x4944c9d2 in H323Connection::OnSelectLogicalChannels() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#10 0x4944c8b1 in H323Connection::InternalEstablishedConnectionCheck() ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#11 0x4944a6d1 in H323Connection::HandleControlData(PPER_Stream) () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#12 0x4944a28c in H323Connection::HandleControlChannel() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#13 0x494992ee in H245TransportThread::Main() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#14 0x48d33177 in PThread::PX_ThreadStart(void*) () from
/root/pwlib/lib/libpt_linux_x86_r.so.1
#15 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0
And this is the console log:
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [Gustavo]
-- Calling party number: [1152880056]
-- Called party name: [0111553037260]
-- Called party number: [0111553037260]
e164: [0111553037263]
-- Executing Dial(H323/ip$10.60.144.14:1240/4096,
Zap/1/0111553037260) in new stack
-- Called 1/0111553037260
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
=*= In CreateRealTimeLogicalChannel for call 4096
-- externalIpAddress: 172.16.254.107
-- externalPort: 13488
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-ALaw-64k{sw}
-- channelsOpen = 2
-- remoteIpAddress: 0.0.0.0
-- remotePort: 0
-- ExternalIpAddress: 172.16.254.107
-- ExternalPort: 13488
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Call with ended abnormally
== H.323 Connection deleted.
channelsOpen = 1
-- Closing logical channel...
channelsOpen = 0
Segmentation fault (core dumped)
[EMAIL PROTECTED] asterisk]#
What is wrong?
Thanks in advance,
Gus
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Re: [Asterisk-Users] chan_h323 - Segmentation fault (core dumped)
Yes, I donwload tgz's from nufone (http://www.nufone.net/downloads/). All
sources was compiled as Jeremy recommeds, and I didn't have troubles with
that. Oh, I'm using RH9.
This is my h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=alaw
allow=gsm
dtmfmode=rfc2833
gatekeeper = DISABLE
[Gustavo]
type=user
host=10.60.144.14
context=default
incominglimit=31
Regards,
Gus
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 13, 2003 4:46 PM
Subject: Re: [Asterisk-Users] chan_h323 - Segmentation fault (core dumped)
Are you using the recommended pwlib and openh323 tarballs?
bkw
On Mon, 13 Oct 2003, CW_ASN wrote:
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790
#1 0x41f8879c in create_connection (call_reference=1349809548) at
chan_h323.c:928
#2 0x41f8f34b in
MyH323Connection::CreateRealTimeLogicalChannel(H323Capability const,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) (this=0x8178758, [EMAIL PROTECTED], dir=IsTransmitter,
sessionID=1)
at ast_h323.cpp:626
#3 0x49470170 in H323RealTimeCapability::CreateChannel(H323Connection,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) const () from /root/openh323/lib/libh323_linux_x86_r.so.1
#4 0x4946071d in H245NegLogicalChannel::OpenWhileLocked(H323Capability
const, unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#5 0x494604e6 in H245NegLogicalChannel::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#6 0x49462423 in H245NegLogicalChannels::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#7 0x4944d311 in H323Connection::OpenLogicalChannel(H323Capability
const,
unsigned, H323Channel::Directions) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#8 0x4944cf98 in H323Connection::SelectDefaultLogicalChannel(unsigned)
()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#9 0x4944c9d2 in H323Connection::OnSelectLogicalChannels() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#10 0x4944c8b1 in H323Connection::InternalEstablishedConnectionCheck()
()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#11 0x4944a6d1 in H323Connection::HandleControlData(PPER_Stream) ()
from
/root/openh323/lib/libh323_linux_x86_r.so.1
#12 0x4944a28c in H323Connection::HandleControlChannel() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#13 0x494992ee in H245TransportThread::Main() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#14 0x48d33177 in PThread::PX_ThreadStart(void*) () from
/root/pwlib/lib/libpt_linux_x86_r.so.1
#15 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0
And this is the console log:
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [Gustavo]
-- Calling party number: [1152880056]
-- Called party name: [0111553037260]
-- Called party number: [0111553037260]
e164: [0111553037263]
-- Executing Dial(H323/ip$10.60.144.14:1240/4096,
Zap/1/0111553037260) in new stack
-- Called 1/0111553037260
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
=*= In CreateRealTimeLogicalChannel for call 4096
-- externalIpAddress: 172.16.254.107
-- externalPort: 13488
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-ALaw-64k{sw}
-- channelsOpen = 2
-- remoteIpAddress: 0.0.0.0
-- remotePort: 0
-- ExternalIpAddress: 172.16.254.107
-- ExternalPort: 13488
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Call with ended abnormally
== H.323 Connection deleted.
channelsOpen = 1
-- Closing logical channel...
channelsOpen = 0
Segmentation fault (core dumped)
[EMAIL PROTECTED] asterisk]#
What is wrong?
Thanks in advance,
Gus
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[Asterisk-Users] chan_h323 Ringing Congestion causes * segfault
We have an odd problem, where inbound H323 (chan_h323) calls will sometimes cause a Ringing Congestion that appears to keep the channels open and never release it until we kill and restart asterisk. These Ringing Congestions start to pile up, which eventually crashes Asterisk. H323 Gateway - Asterisk (chan_h323) - Tor2/PRI - PSTN Has anyone ran into this problem or know how to resolve it? The H323 device making the calls doesn't seem to have a problem calling other H323 gateways or gatekeepers, this problem only appears in Asterisk. Again this problem is intermittent and occurs once a day. I have included a paste of the Ringing Congestions below as well as the GDB dump. Thanks --- H323/ip$61.33.231.34:24585/5 (h323 17704703893 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24581/2 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24596/15 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24592/11 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24591/10 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24589/8 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24581/1 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24647/67 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24644/64 (h323 14349230857 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24643/63 (h323 14349230857 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24641/61 (h323 19788482664 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24640/60 (h323 19788482994 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24634/54 (h323 18586380364 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24608/28 (h323 12062233600 4 ) Ringing Congestion(Empty) - (gdb) bt #0 connection_made (call_reference=1106240992) at chan_h323.c:1188 #1 0x41ef7973 in MyH323EndPoint::OnConnectionEstablished(H323Connection, PString const) ( this=0x814c1a8, [EMAIL PROTECTED], [EMAIL PROTECTED]) at ast_h323.cpp:294 #2 0x482985f5 in H323Connection::OnEstablished() () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #3 0x482a215e in H323Connection::InternalEstablishedConnectionCheck() () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x48297d28 in H323Connection::HandleSignalPDU(H323SignalPDU) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x48297902 in H323Connection::HandleSignallingChannel() () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x482a8795 in H225CallThread::Main() () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x47b750a7 in PThread::PX_ThreadStart(void*) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #8 0x40031332 in start_thread () from /lib/tls/libpthread.so.0 (gdb) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 config
Hello, Camparing chan_h323 config with chan_oh323 config, In the codec section chan_oh323 allow me to specify frame value. Is there a equivalent in chan_h323? Or if not, what is the default frame value if I use G.729(digium). Foong
RE: [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-Digium Partner
Since then I couldn't test it, but now I installed EtheReal last version with h323 support. Did some calls and perceived that the call is being cut after the Master/Slave negotiation. Asterisk is sending an EndSession as you can see in the file attached. If the list doesn't allow attachments, the same file can be found at http://isamarmaia.org/packets.pak BTW, I didn't find the patch you mentioned. Could you gimme its URL? Thanks a lot, Isamar Maia On Wed, 27 Aug 2003 [EMAIL PROTECTED] wrote: Hi The endpoint seems to be running Radvision h323 stack, and I know chan_h323 works with Radvision, there could be a couple of reasons!! 1) You dont have G729A in the capabilities of remote endpoint 2) The packetization interval is way off The best way would be to run ethereal or dump323 and see what is being negotiated. Also try to use fastConnect on both sides and force same packetization, (you can use my patch posted a couple of days ago to force packetization interval in G729 in chan_h323) Isamar Said I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2 HSMU RAD: cmHookSend(masterSlaveDeterminationAck) 11- HSMU 0 Remote capabilities list: 0- HSMU 0 [1] g729AnnexA: Audio Receive 0- HSMU 0 Try matching local element: 0- HSMU 0 [1] g7231: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [2] g729: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [3] g711Ulaw64k: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [4] t38fax: Data Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [5] g729: Audio Receive and Transmit 0- HSMU 3 HSMU 2: Capabilities: NO MATCH FOUND! 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE - error release 1- HSMU 2 HSMU 2: abort timer in state H245 WAIT COMPLETE 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE == RELEASING 10- RADH 2 HSMU RAD: cmHookSend(endSessionCommand) 3- RAD 2 HSMU 2: cmEvCallControlStateChanged(cmControlStateTransportDisconnected, - ) Anybody has any idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users packets.pak Description: Binary data
[Asterisk-Users] chan_h323 as a gatekeeper?
hi IIRC, Jeremy once said that chan_h323 could be used as a gatekeeper but perhaps lacking a few features as compared to gnugk. Is this possible? I have some dlink DPH-100H phoes here for testing, but they require a gatekeeper, and if I can do it, I'd love to keep gnugk out of this. thanks roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 core dump on reload, works fine atstartup
This happens only on relaod. You can disable reload routine in chan_h323.c ... Martin On 1 Sep 2003, Michael wrote: I'm running the CVS from last week and from day one (over 4 months now) I've had this problem where asterisk core dumps when using chan_h323. It appears to be a problem with pwlib and the console, but I'm not sure how to read the below output from gdb. I can start Asterisk just fine and chan_h323 works great when sending and receiving calls. I only have this core dump problem when sending a reload to Asterisk via the CLI or asterisk -rx reload. Environment paths: LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 Core dump info: (gdb) bt #0 0x4831e20c in PSocket::os_select(int, fd_set*, fd_set*, fd_set*, PIntArray const, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #1 0x48315a2a in PSocket::Select(PSocket::SelectList, PSocket::SelectList, PSocket::SelectList, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #2 0x483151a7 in PSocket::Select(PSocket::SelectList, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x48aec440 in H323TransportUDP::DiscoverGatekeeper(H323Gatekeeper, H323RasPDU, H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x48b023b9 in H323Gatekeeper::StartDiscovery(H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x48b020e3 in H323Gatekeeper::DiscoverByAddress(H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x48aa10ae in H323EndPoint::SetGatekeeper(PString const, H323Transport*) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x41ef5d11 in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x41efe680 65.39.220.195, secret=0x41efe700 ) at ast_h323.cpp:949 #8 0x41eeed81 in reload () at chan_h323.c:1595 #9 0x08055362 in ast_module_reload () at loader.c:159 #10 0x0806b63a in handle_reload (fd=10, argc=1, argv=0x4effe6dc) at cli.c:105 #11 0x0806b42a in ast_cli_command (fd=10, s=0x0) at cli.c:1006 #12 0x08080aa0 in netconsole (vconsole=0x80b6348) at asterisk.c:192 #13 0x4002f2b6 in start_thread () from /lib/tls/libpthread.so.0 Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 core dump on reload, works fine atstartup
On Tue, 2003-09-02 at 09:24, Martin Pycko wrote: This happens only on relaod. You can disable reload routine in chan_h323.c ... Thanks. I'll give it a try. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 core dump on reload, works fine atstartup
It only core's when u use the gatekeeper component, due to the way pwlib deals with memory allocation. This is going to take quite a lot of trying various different incantations to fix, unfortunately I cannot justify dedicating that kind time, at this point. Sorry, Jeremy McNamara Martin Pycko wrote: This happens only on relaod. You can disable reload routine in chan_h323.c ... Martin On 1 Sep 2003, Michael wrote: I'm running the CVS from last week and from day one (over 4 months now) I've had this problem where asterisk core dumps when using chan_h323. It appears to be a problem with pwlib and the console, but I'm not sure how to read the below output from gdb. I can start Asterisk just fine and chan_h323 works great when sending and receiving calls. I only have this core dump problem when sending a reload to Asterisk via the CLI or asterisk -rx reload. Environment paths: LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 Core dump info: (gdb) bt #0 0x4831e20c in PSocket::os_select(int, fd_set*, fd_set*, fd_set*, PIntArray const, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #1 0x48315a2a in PSocket::Select(PSocket::SelectList, PSocket::SelectList, PSocket::SelectList, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #2 0x483151a7 in PSocket::Select(PSocket::SelectList, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x48aec440 in H323TransportUDP::DiscoverGatekeeper(H323Gatekeeper, H323RasPDU, H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x48b023b9 in H323Gatekeeper::StartDiscovery(H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x48b020e3 in H323Gatekeeper::DiscoverByAddress(H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x48aa10ae in H323EndPoint::SetGatekeeper(PString const, H323Transport*) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x41ef5d11 in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x41efe680 65.39.220.195, secret=0x41efe700 ) at ast_h323.cpp:949 #8 0x41eeed81 in reload () at chan_h323.c:1595 #9 0x08055362 in ast_module_reload () at loader.c:159 #10 0x0806b63a in handle_reload (fd=10, argc=1, argv=0x4effe6dc) at cli.c:105 #11 0x0806b42a in ast_cli_command (fd=10, s=0x0) at cli.c:1006 #12 0x08080aa0 in netconsole (vconsole=0x80b6348) at asterisk.c:192 #13 0x4002f2b6 in start_thread () from /lib/tls/libpthread.so.0 Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 core dump on reload, works fine at startup
I'm running the CVS from last week and from day one (over 4 months now) I've had this problem where asterisk core dumps when using chan_h323. It appears to be a problem with pwlib and the console, but I'm not sure how to read the below output from gdb. I can start Asterisk just fine and chan_h323 works great when sending and receiving calls. I only have this core dump problem when sending a reload to Asterisk via the CLI or asterisk -rx reload. Environment paths: LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 Core dump info: (gdb) bt #0 0x4831e20c in PSocket::os_select(int, fd_set*, fd_set*, fd_set*, PIntArray const, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #1 0x48315a2a in PSocket::Select(PSocket::SelectList, PSocket::SelectList, PSocket::SelectList, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #2 0x483151a7 in PSocket::Select(PSocket::SelectList, PTimeInterval const) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x48aec440 in H323TransportUDP::DiscoverGatekeeper(H323Gatekeeper, H323RasPDU, H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x48b023b9 in H323Gatekeeper::StartDiscovery(H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x48b020e3 in H323Gatekeeper::DiscoverByAddress(H323TransportAddress const) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x48aa10ae in H323EndPoint::SetGatekeeper(PString const, H323Transport*) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x41ef5d11 in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x41efe680 65.39.220.195, secret=0x41efe700 ) at ast_h323.cpp:949 #8 0x41eeed81 in reload () at chan_h323.c:1595 #9 0x08055362 in ast_module_reload () at loader.c:159 #10 0x0806b63a in handle_reload (fd=10, argc=1, argv=0x4effe6dc) at cli.c:105 #11 0x0806b42a in ast_cli_command (fd=10, s=0x0) at cli.c:1006 #12 0x08080aa0 in netconsole (vconsole=0x80b6348) at asterisk.c:192 #13 0x4002f2b6 in start_thread () from /lib/tls/libpthread.so.0 Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 core dump on reload, works fine atstartup
On Mon, 2003-09-01 at 11:19, Brian West wrote: Are you using the recommended pwlib and openh323 versions? Yes. lrwxrwxrwx1 root root 12 Aug 17 20:39 pwlib - pwlib-1.4.11 lrwxrwxrwx1 root root 15 Aug 17 20:01 openh323 - openh323-1.11.7 Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 support for phone numbers via gateway?
Does chan_h323 support phone number calling via a gateway? ie., something like calling 5000 forwarded to: exten = 5000,1,Dial(h323/[EMAIL PROTECTED]) if so - what format should the exten be in? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 does not seem to send the destimation number to gateway
Continuing my problems with h323. I think I am getting closer. SJPhone works direct to the gateway - calls and answers fine on the pstn. So the gateway is working. Inbound calls from PSTN = Gateway = Asterisk = Phone work great! Outbound from Asterisk = Gateway = PSTN still remains a problem. The debug stuff on the gateway receives the call signal from asterisk - but does not receive the number to call - its errors with callID is -1 (nothing to call) Any ideas for the correct format to use within extenensions.conf for outbound phone number via chan_h323 and a gateway? h323 works fine if it is just an IP address that it is calling, ie, a softphone. Thanks for your help Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323/g729 - X100P connecting to non-Digium Partner
I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2 HSMU RAD: cmHookSend(masterSlaveDeterminationAck) 11- HSMU 0 Remote capabilities list: 0- HSMU 0 [1] g729AnnexA: Audio Receive 0- HSMU 0 Try matching local element: 0- HSMU 0 [1] g7231: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [2] g729: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [3] g711Ulaw64k: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [4] t38fax: Data Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [5] g729: Audio Receive and Transmit 0- HSMU 3 HSMU 2: Capabilities: NO MATCH FOUND! 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE - error release 1- HSMU 2 HSMU 2: abort timer in state H245 WAIT COMPLETE 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE == RELEASING 10- RADH 2 HSMU RAD: cmHookSend(endSessionCommand) 3- RAD 2 HSMU 2: cmEvCallControlStateChanged(cmControlStateTransportDisconnected, - ) Anybody has any idea? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-DigiumPartner
Hi The endpoint seems to be running Radvision h323 stack, and I know chan_h323 works with Radvision, there could be a couple of reasons!! 1) You dont have G729A in the capabilities of remote endpoint 2) The packetization interval is way off The best way would be to run ethereal or dump323 and see what is being negotiated. Also try to use fastConnect on both sides and force same packetization, (you can use my patch posted a couple of days ago to force packetization interval in G729 in chan_h323) Isamar Said I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2 HSMU RAD: cmHookSend(masterSlaveDeterminationAck) 11- HSMU 0 Remote capabilities list: 0- HSMU 0 [1] g729AnnexA: Audio Receive 0- HSMU 0 Try matching local element: 0- HSMU 0 [1] g7231: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [2] g729: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [3] g711Ulaw64k: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [4] t38fax: Data Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [5] g729: Audio Receive and Transmit 0- HSMU 3 HSMU 2: Capabilities: NO MATCH FOUND! 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE - error release 1- HSMU 2 HSMU 2: abort timer in state H245 WAIT COMPLETE 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE == RELEASING 10- RADH 2 HSMU RAD: cmHookSend(endSessionCommand) 3- RAD 2 HSMU 2: cmEvCallControlStateChanged(cmControlStateTransportDisconnected, - ) Anybody has any idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323 and a Cisco Gateway
Well depends.. what kind of problem are you having? http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml http://www.cisco.com/en/US/tech/tk652/tk701/technologies_problem_troubleshooting09186a00800c5e33.shtml Check those... I suspect one of those has nailed ya. If you have PRI and you try to terminate outbound via chan_h323 you must have bearer-cap speech on your voice-ports. Because chan_h323 isn't sending the appropriate bearer cap in the H.225 SETUP message. Hours of beating head on desk and searching... Hope this helps. Thanks, Brian On Tue, 26 Aug 2003, Steven Thomas wrote: Hi, Can anyone tell me what should be included in h323.conf to get asterisk to talk to a Cisco 2600 gateway? Any statement examples for extensions.conf would also be appreciated. Thanks. Will chan_h323 use the Cisco as a gateway anyway? Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323.c
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was
not created so no h323 support in asterisk.
Just wondering when to expect it again because I was stupid and didn't
make a backup of the asterisk code before wiping the directory for a
rebuild.
Has anyone gotten H.323 channel complied on Redhat 9.0? Every time I try,
I get a ton of errors from ptlib. I'm about ready to punt this sucker out
the door. I really like what I have seen out of asterisk so far...
Example of errors:
In file included from /usr/include/ptlib/contain.h:218,
from /usr/include/ptlib.h:137,
from ast_h323.h:29,
from ast_h323.cpp:27:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
PObject::BOOL'
/usr/include/ptlib/object.h:1214: parse error before `(' token
/usr/include/ptlib/object.h:1265: syntax error before `operator'
/usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
/usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
return
type
Any help would be appriciated, even if it's a recommendation to another
flavor of linux.
Thanks
Sean
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Re: [Asterisk-Users] chan_h323.c
I have been using Slackware 9 with the sources for ptlib_unix, pwlib_unix,
openh323, asterisk, zaptel and libpri in /root/src
1) load all 6 packages (+ ohphone if you need an h323 client) from cvs to
/root/src
2) /root/src/pwlib: configure, make, make install, ldconfig (not all that
sure why, but Slackware requires ldconfig to be run)
3) /root/src/openh323: configure, make, make install, ldconfig
4) /root/src/zaptel: make, make install (reload and reconfigure your zaptel
card)
5) /root/src/libpri: make, make install (I don't have a PRI card so I don't
do anything here)
6) /root/src/asterisk/channels/h323:
- edit Makfile
- set PWLIBDIR = $(HOME)/src/pwlib
- set OPENH323DIR = $(HOME)/src/openh323
- make, make install (installs openh323.a) (make samples if you do not
have h323.conf in /etc/asterisk when done)
7) /root/src/asterisk: make, make install, make samples
8) asterisk -vvvc
- the last section should load chan_h323
I haven't had any problems compiling this from CVS for almost a month on at
least three different systems with some version of Slackware. I have had
problems with other things like transferring calls but that's a different
issue.
John.
- Original Message -
From: Sean Figgins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 12:22 PM
Subject: Re: [Asterisk-Users] chan_h323.c
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was
not created so no h323 support in asterisk.
Just wondering when to expect it again because I was stupid and didn't
make a backup of the asterisk code before wiping the directory for a
rebuild.
Has anyone gotten H.323 channel complied on Redhat 9.0? Every time I try,
I get a ton of errors from ptlib. I'm about ready to punt this sucker out
the door. I really like what I have seen out of asterisk so far...
Example of errors:
In file included from /usr/include/ptlib/contain.h:218,
from /usr/include/ptlib.h:137,
from ast_h323.h:29,
from ast_h323.cpp:27:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
PObject::BOOL'
/usr/include/ptlib/object.h:1214: parse error before `(' token
/usr/include/ptlib/object.h:1265: syntax error before `operator'
/usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
/usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
return
type
Any help would be appriciated, even if it's a recommendation to another
flavor of linux.
Thanks
Sean
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Re: [Asterisk-Users] chan_h323.c
Great! Thanks for the recommendation. I'll beat on Redhat a little bit
longer, then try to load slackware and give that a whirl.
Thanks again.
Sean
On Wed, 20 Aug 2003, John Fortman wrote:
I have been using Slackware 9 with the sources for ptlib_unix, pwlib_unix,
openh323, asterisk, zaptel and libpri in /root/src
1) load all 6 packages (+ ohphone if you need an h323 client) from cvs to
/root/src
2) /root/src/pwlib: configure, make, make install, ldconfig (not all that
sure why, but Slackware requires ldconfig to be run)
3) /root/src/openh323: configure, make, make install, ldconfig
4) /root/src/zaptel: make, make install (reload and reconfigure your zaptel
card)
5) /root/src/libpri: make, make install (I don't have a PRI card so I don't
do anything here)
6) /root/src/asterisk/channels/h323:
- edit Makfile
- set PWLIBDIR = $(HOME)/src/pwlib
- set OPENH323DIR = $(HOME)/src/openh323
- make, make install (installs openh323.a) (make samples if you do not
have h323.conf in /etc/asterisk when done)
7) /root/src/asterisk: make, make install, make samples
8) asterisk -vvvc
- the last section should load chan_h323
I haven't had any problems compiling this from CVS for almost a month on at
least three different systems with some version of Slackware. I have had
problems with other things like transferring calls but that's a different
issue.
John.
- Original Message -
From: Sean Figgins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 20, 2003 12:22 PM
Subject: Re: [Asterisk-Users] chan_h323.c
On Mon, 18 Aug 2003, Mark Spencer wrote:
It's up one directly. It just moved.
Run make in h323 then do make install on asterisk again.
On Mon, 18 Aug 2003, John Fortman wrote:
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp,
ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was
not created so no h323 support in asterisk.
Just wondering when to expect it again because I was stupid and didn't
make a backup of the asterisk code before wiping the directory for a
rebuild.
Has anyone gotten H.323 channel complied on Redhat 9.0? Every time I try,
I get a ton of errors from ptlib. I'm about ready to punt this sucker out
the door. I really like what I have seen out of asterisk so far...
Example of errors:
In file included from /usr/include/ptlib/contain.h:218,
from /usr/include/ptlib.h:137,
from ast_h323.h:29,
from ast_h323.cpp:27:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
PObject::BOOL'
/usr/include/ptlib/object.h:1214: parse error before `(' token
/usr/include/ptlib/object.h:1265: syntax error before `operator'
/usr/include/ptlib/object.h:1214: duplicate member `PObject::BOOL'
/usr/include/ptlib/object.h:1274: ISO C++ forbids defining types within
return
type
Any help would be appriciated, even if it's a recommendation to another
flavor of linux.
Thanks
Sean
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[Asterisk-Users] chan_h323.c
What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp, ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was not created so no h323 support in asterisk. Just wondering when to expect it again because I was stupid and didn't make a backup of the asterisk code before wiping the directory for a rebuild.
Re: [Asterisk-Users] chan_h323.c
It's up one directly. It just moved. Run make in h323 then do make install on asterisk again. Mark On Mon, 18 Aug 2003, John Fortman wrote: What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp, ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was not created so no h323 support in asterisk. Just wondering when to expect it again because I was stupid and didn't make a backup of the asterisk code before wiping the directory for a rebuild. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 one way audio
Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323 one way audio
i also encountered this problem i'm not too sure either but i don't think codec has to do anything with it for i tried mix and matching but to no avail. so for the meantime, try adjusting the tos for oh323 and i think you could live with it by the way, are you running cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:56 PM Subject: [Asterisk-Users] Chan_h323 one way audio Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323 one way audio
not sure what you mean by 'are you running cvs'? What does the TOS setting do? Regards, Steven Thomas Kelvin Chua [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Chan_h323 one way audio .digium.com 18-08-03 12:19 PM Please respond to asterisk-users i also encountered this problem i'm not too sure either but i don't think codec has to do anything with it for i tried mix and matching but to no avail. so for the meantime, try adjusting the tos for oh323 and i think you could live with it by the way, are you running cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:56 PM Subject: [Asterisk-Users] Chan_h323 one way audio Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323 one way audio
set ipTos=lowdelay in oh323.conf and try to see what happens. (of course this would mean your switch should have the ability to detect TOS bits in the packet headers) what version of * are you using? did you check against cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 11:30 AM Subject: Re: [Asterisk-Users] Chan_h323 one way audio not sure what you mean by 'are you running cvs'? What does the TOS setting do? Regards, Steven Thomas Kelvin Chua [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Chan_h323 one way audio .digium.com 18-08-03 12:19 PM Please respond to asterisk-users i also encountered this problem i'm not too sure either but i don't think codec has to do anything with it for i tried mix and matching but to no avail. so for the meantime, try adjusting the tos for oh323 and i think you could live with it by the way, are you running cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:56 PM Subject: [Asterisk-Users] Chan_h323 one way audio Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323.so native?
Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or do you have to patch or add it in to the source directory structure before compiling? Can / and maybe how can this be added after? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323.so native?
Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or do you have to patch or add it in to the source directory structure before compiling? Can / and maybe how can this be added after? H.323 is coming into asterisk cvs, and i think is trying to find if you have openh323, anyway is enought to go to h323 directory from channel directory in asterisk source and make, and then make install. Thanks. Regards, Steven Thomas Diana the skinny one : ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323, Asterisk and DTMF issue
We use rfc2833 for everything and have no trouble. Make sure your 7960 is sending the right indications. Jeremy McNamara Jay Sakata wrote: I have the same problem that Michael describes below does anyone have any recommendations? Jay __ Hi folks, Im using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info wont work with Asterisks voicemail system. Im using the g.729 codec for h323 and Asterisk. Im told dtmfmode=inband wont work with g.729. Is it possible to use dtmfmode=info with h323 and access my Asterisk voicemail? Summary: dtmfmode = info ; works with h323 not with Asterisk Voicemail dtmfmode = inband; works with h323 (with a flood of warnings) not with Asterisk Voicemail dtmfmode = rfc2833 ; works with Asterisk Voicemail not with h323 Any help would be greatly appreciated. Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323, Asterisk and DTMF issue
One variance to the configuration that was described is that I am using a Cisco ATA186 rather than a 7960 IP phone. I have tried configuring the ATA with in-band DTMF and out-of-band DTMF both where unsuccessful. Jay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Tuesday, August 12, 2003 11:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_h323, Asterisk and DTMF issue We use rfc2833 for everything and have no trouble. Make sure your 7960 is sending the right indications. Jeremy McNamara Jay Sakata wrote: I have the same problem that Michael describes below does anyone have any recommendations? Jay ___ ___ Hi folks, I'm using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info won't work with Asterisk's voicemail system. I'm using the g.729 codec for h323 and Asterisk. I'm told dtmfmode=inband won't work with g.729. Is it possible to use dtmfmode=info with h323 and access my Asterisk voicemail? Summary: dtmfmode = info ; works with h323 not with Asterisk Voicemail dtmfmode = inband; works with h323 (with a flood of warnings) not with Asterisk Voicemail dtmfmode = rfc2833 ; works with Asterisk Voicemail not with h323 Any help would be greatly appreciated. Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323
Having problems to connect another device using chan_h323. When G723.1 or G711: log says: NOTICE[15376]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 64 NOTICE[15376]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 4 to 1 WARNING[15376]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 4, while native formats is 1 (read/write = 64/4) WARNING[15376]: File app_dial.c, Line 299 (wait_for_answer): Unable to forward voice == No one is available to answer at this time But it works using chan_oh323. I appreciate any help. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_H323, G729 (minor problem)
ok ... I removed the dtmfmode=inband from the h323.conf file which resulted in the error messages vanishing .. ya I thought ... alas DTMF tones sent to an IVR at the other end of the connection do not work either!!! My incoming calls are coming from PSTN lines through an E1 so DTMF must be inline .. THe (thousands of) error messages aren't really a problem, just annoying. Dave - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 15, 2003 4:28 PM Subject: Re: [Asterisk-Users] Chan_H323, G729 (minor problem) You're trying to detect inband dtmfs from the codec stream. Martin On Tue, 15 Jul 2003, Dave Alan Caruana wrote: hi .. I have finally managed to get Chan_H323 G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib oH323 with versions taken from nufone's site (http://www.nufone.net/downloads) since the latest versions do not have support for G729. Remember to set the environment versions as described in the Readme files. 4. Modify the makefile of chan_h323 (which is in /usr/src/asterisk/channels/h323) to re-enable the G729 code. 5. in h323.conf put in allow=g729 and you should have a working configuration .. now for my question .. during G729 calls I am getting repeatedly the message WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames this scrolls up the screen at a very high rate of knots.. the call is unaffected and goes through normally. Is this something wrong? normal? can it be fixed/suppressed? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_H323, G729 (minor problem)
dtmfmode=rfc2833 or dtmfmode=info try that instead Martin On Thu, 17 Jul 2003, Dave Alan Caruana wrote: ok ... I removed the dtmfmode=inband from the h323.conf file which resulted in the error messages vanishing .. ya I thought ... alas DTMF tones sent to an IVR at the other end of the connection do not work either!!! My incoming calls are coming from PSTN lines through an E1 so DTMF must be inline .. THe (thousands of) error messages aren't really a problem, just annoying. Dave - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 15, 2003 4:28 PM Subject: Re: [Asterisk-Users] Chan_H323, G729 (minor problem) You're trying to detect inband dtmfs from the codec stream. Martin On Tue, 15 Jul 2003, Dave Alan Caruana wrote: hi .. I have finally managed to get Chan_H323 G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib oH323 with versions taken from nufone's site (http://www.nufone.net/downloads) since the latest versions do not have support for G729. Remember to set the environment versions as described in the Readme files. 4. Modify the makefile of chan_h323 (which is in /usr/src/asterisk/channels/h323) to re-enable the G729 code. 5. in h323.conf put in allow=g729 and you should have a working configuration .. now for my question .. during G729 calls I am getting repeatedly the message WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames this scrolls up the screen at a very high rate of knots.. the call is unaffected and goes through normally. Is this something wrong? normal? can it be fixed/suppressed? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_H323, G729 (minor problem)
hi .. I have finally managed to get Chan_H323 G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib oH323 with versions taken from nufone's site (http://www.nufone.net/downloads) since the latest versions do not have support for G729. Remember to set the environment versions as described in the Readme files. 4. Modify the makefile of chan_h323 (which is in /usr/src/asterisk/channels/h323) to re-enable the G729 code. 5. in h323.conf put in allow=g729 and you should have a working configuration .. now for my question .. during G729 calls I am getting repeatedly the message WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames this scrolls up the screen at a very high rate of knots.. the call is unaffected and goes through normally. Is this something wrong? normal? can it be fixed/suppressed? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_H323, G729 (minor problem)
You're trying to detect inband dtmfs from the codec stream. Martin On Tue, 15 Jul 2003, Dave Alan Caruana wrote: hi .. I have finally managed to get Chan_H323 G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib oH323 with versions taken from nufone's site (http://www.nufone.net/downloads) since the latest versions do not have support for G729. Remember to set the environment versions as described in the Readme files. 4. Modify the makefile of chan_h323 (which is in /usr/src/asterisk/channels/h323) to re-enable the G729 code. 5. in h323.conf put in allow=g729 and you should have a working configuration .. now for my question .. during G729 calls I am getting repeatedly the message WARNING[311314]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames this scrolls up the screen at a very high rate of knots.. the call is unaffected and goes through normally. Is this something wrong? normal? can it be fixed/suppressed? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323, Asterisk and DTMF issue
Hi folks, Im using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info wont work with Asterisks voicemail system. Im using the g.729 codec for h323 and Asterisk. Im told dtmfmode=inband wont work with g.729. Is it possible to use dtmfmode=info with h323 and access my Asterisk voicemail? Summary: dtmfmode = info ; works with h323 not with Asterisk Voicemail dtmfmode = inband ; works with h323 (with a flood of warnings) not with Asterisk Voicemail dtmfmode = rfc2833 ; works with Asterisk Voicemail not with h323 Any help would be greatly appreciated. Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323.c compile error
Hello all,
I got the following error compiling h323 support in the latest cvs. Below
the error is a diff to the file that I got to make it work. I took an
example out of sip as far as the syntax for ast_rtp_new. Not sure if it is
correct or not, but it seems to work. Please correct me if I am wrong in
the additional 2 arguements.
Regards,
Scott
cc -g -pg -c -o chan_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes
-Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE
-DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING
-DP_USE_PRAGMA -I/usr/local/pwlib/include/ptlib/unix
-I/usr/local/pwlib/include -I/usr/local/openh323/include
-Wno-missing-prototypes -Wno-missing-declarations chan_h323.c
chan_h323.c: In function `oh323_alloc':
chan_h323.c:687: too few arguments to function `ast_rtp_new'
chan_h323.c: At top level:
chan_h323.c:1601: warning: initialization from incompatible pointer type
make: *** [chan_h323.o] Error 1
--- chan_h323.c 2003-07-01 08:09:33.0 -0400
+++ chan_h323.c.mod 2003-06-30 10:25:30.0 -0400
@@ -684,7 +684,7 @@
/* Keep track of stuff */
memset(p, 0, sizeof(struct oh323_pvt));
- p-rtp = ast_rtp_new(NULL, NULL);
+ p-rtp = ast_rtp_new(NULL, NULL, 1, 0);
if (!p-rtp) {
ast_log(LOG_WARNING, Unable to create RTP session: %s\n,
strerror(errno));
free(p);
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[Asterisk-Users] chan_h323 woes
I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled, and chan_h323 module does not load with undefined symbol _ZTI19H323AudioCapability. What could be the problem? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 woes
This is covered in asterisk/channels/h323/README RTFM Jeremy McNamara Peter Zeltins wrote: I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled, and chan_h323 module does not load with undefined symbol _ZTI19H323AudioCapability. What could be the problem? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 - pwlib 1.4.11, openh 1.11.7 comiple problems
I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [EMAIL PROTECTED] h323]# make clean rm -f *.o *.so core.* [EMAIL PROTECTED] h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations chan_h323.c g++ -g -pg -c -o ast_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations ast_h323.cpp chan_h323.h:30: warning: `sockaddr_in bindaddr' defined but not used g++ -g -pg -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L/usr/src/pwlib/lib -lpt_linux_x86_r -L/usr/src/openh323/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat /usr/bin/ld: cannot find -lpt_linux_x86_r collect2: ld returned 1 exit status make: *** [chan_h323.so] Error 1 [EMAIL PROTECTED] h323]# Any ideas? Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 - pwlib 1.4.11, openh 1.11.7 comipleproblems
you need to build pwlib and/or setup your environment properly. See asterisk/channels/h323/README Jeremy McNamara [EMAIL PROTECTED] wrote: I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [EMAIL PROTECTED] h323]# make clean rm -f *.o *.so core.* [EMAIL PROTECTED] h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations chan_h323.c g++ -g -pg -c -o ast_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations ast_h323.cpp chan_h323.h:30: warning: `sockaddr_in bindaddr' defined but not used g++ -g -pg -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L/usr/src/pwlib/lib -lpt_linux_x86_r -L/usr/src/openh323/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat /usr/bin/ld: cannot find -lpt_linux_x86_r collect2: ld returned 1 exit status make: *** [chan_h323.so] Error 1 [EMAIL PROTECTED] h323]# Any ideas? Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323 problems
I found the problem. A 'make opt' doesn't create the pwlib/lib directory when compiling pwlib. You have to do a 'make'. I did a 'make install' for h323 but I get a Segmentation Fault when I start Asterisk with chan_h323. A backtrace shows the following: (gdb) bt #0 0x42029241 in kill () from /lib/i686/libc.so.6 #1 0x46bfd5b4 in PAssertFunc () from /data/gnugk/pwlib/lib/libpt_linux_x86_r.so.1 #2 0x46c11e02 in PAssertFunc () from /data/gnugk/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x4741d991 in H323EndPoint::SetLocalUserName () from /data/gnugk/openh323/lib/libh323_linux_x86_r.so.1 #4 0x47488aff in H323Gatekeeper::SetPassword () from /data/gnugk/openh323/lib/libh323_linux_x86_r.so.1 #5 0x4741798b in H323EndPoint::InternalCreateGatekeeper () from /data/gnugk/openh323/lib/libh323_linux_x86_r.so.1 #6 0x47417634 in H323EndPoint::SetGatekeeper () from /data/gnugk/openh323/lib/libh323_linux_x86_r.so.1 #7 0x41fb014c in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x41fb8800 65.39.220.195, secret=0x41fb8880 ) at ast_h323.cpp:915 #8 0x41fab366 in load_module () at chan_h323.c:1646 #9 0x08053db6 in ast_load_resource (resource_name=0x80cbdab chan_h323.so) at loader.c:298 #10 0x080541ec in load_modules () at loader.c:393 #11 0x0807a39a in main (argc=2, argv=0xb894) at asterisk.c:1330 #12 0x42017499 in __libc_start_main () from /lib/i686/libc.so.6 (gdb) Regards, Micahel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, June 16, 2003 11:27 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_h323 - pwlib 1.4.11, openh 1.11.7 comiple problems I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [EMAIL PROTECTED] h323]# make clean rm -f *.o *.so core.* [EMAIL PROTECTED] h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations chan_h323.c g++ -g -pg -c -o ast_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations ast_h323.cpp chan_h323.h:30: warning: `sockaddr_in bindaddr' defined but not used g++ -g -pg -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L/usr/src/pwlib/lib -lpt_linux_x86_r -L/usr/src/openh323/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat /usr/bin/ld: cannot find -lpt_linux_x86_r collect2: ld returned 1 exit status make: *** [chan_h323.so] Error 1 [EMAIL PROTECTED] h323]# Any ideas? Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 problems
I've done this, with the exact versions you state, 3 times today - every one does the full , proper thing. I did: cd pwlib;make clean;make opt;make install cd ../openh323;make clean;make opt;make install cd ../asterisk/asterisk/channels/h323;make clean;make install;make samples works every time on a clean RedHat 7.2 100% install I hope something in there helps... - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 8:20 PM Subject: RE: [Asterisk-Users] chan_h323 problems I did RTFM. It looks like the instructions conflict each other. Here's what it says: 4. Build the debug and release versions of the PWLib library as follows: cd $PWLIBDIR make both Your README under channels/h323/README says: cd /path/to/pwlib make clean opt Which one do I follow? If I do a 'make opt' it won't build the libs in pwlib. I tried it twice, 'make opt' won't build it but 'make both' will. I'm using PWLib 1.4.11 and Openh323 1.11.7. If I've misread something, please let me know. Asterisk now loads without core dumping (chan_oh323 was installed, it's been removed now). Although, the outgoing quality of the call is very choppy. Incoming works fine, no problems. Any idea what would cause outgoing calls to have problems? I'm sending these calls to GnuGK which then sends the calls to a Quintum or Cisco H323 Gateway (both are having the same problem). Regards, Michael No.. you MUST do a make opt. RTFM http://www.openh323.org/build.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 + openh323 CVS = no go?
Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external chan_oh323. That setup seems to drop small audio snippets like VoiceMail's password prompt, though. So I'm trying to give chan_h323 another chance. However, I get: ast_h323.cpp: In function `int h323_set_capability(int, int)': ast_h323.cpp:780: `H323_G729ACapability' undeclared (first use this function) ast_h323.cpp:780: (Each undeclared identifier is reported only once ast_h323.cpp:780: for each function it appears in.) ast_h323.cpp:780: `g729aCap' undeclared (first use this function) ast_h323.cpp:781: parse error before `)' ast_h323.cpp: At top level: chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 This is both with openh323-1.12.0 and their current CVS. (using current CVS snapshot of asterisk, too) Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
Hello, I've been working with the chan_h323 myself, and I had several problems, but finally got it working. I had to do things in the following order: (1) build and installed asterisk as root (2)I built pwlib and openh323 into my home directory (not root) and built them there as me, I downloaded the tarballs and compiled them.(you could probably do the same thing as root) I did not yet have the os install of the libraries on the system, as this seemed to mess me up. (3) I built the chan_h323 object as myself. (4) I installed the chan_h323.so (make install) as root (5) finally, I installed the system libraries for pwlib and openh323 After all of that, it seemed to work. Good luck, Kelly. On Tue, 2003-06-10 at 12:40, Siggi Langauf wrote: Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external chan_oh323. That setup seems to drop small audio snippets like VoiceMail's password prompt, though. So I'm trying to give chan_h323 another chance. However, I get: ast_h323.cpp: In function `int h323_set_capability(int, int)': ast_h323.cpp:780: `H323_G729ACapability' undeclared (first use this function) ast_h323.cpp:780: (Each undeclared identifier is reported only once ast_h323.cpp:780: for each function it appears in.) ast_h323.cpp:780: `g729aCap' undeclared (first use this function) ast_h323.cpp:781: parse error before `)' ast_h323.cpp: At top level: chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 This is both with openh323-1.12.0 and their current CVS. (using current CVS snapshot of asterisk, too) Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Kelly McDonald [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
If you would have followed the build instructions laid out by the Open H.323 folks you wouldn't have had to go thru all of that. http://www.openh323.org/build.html (Notice they NEVER tell you to make install ANYTHING, there is a reason for that) Jeremy McNamara Kelly McDonald wrote: Hello, I've been working with the chan_h323 myself, and I had several problems, but finally got it working. I had to do things in the following order: (1) build and installed asterisk as root (2)I built pwlib and openh323 into my home directory (not root) and built them there as me, I downloaded the tarballs and compiled them.(you could probably do the same thing as root) I did not yet have the os install of the libraries on the system, as this seemed to mess me up. (3) I built the chan_h323 object as myself. (4) I installed the chan_h323.so (make install) as root (5) finally, I installed the system libraries for pwlib and openh323 After all of that, it seemed to work. Good luck, Kelly. On Tue, 2003-06-10 at 12:40, Siggi Langauf wrote: Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external chan_oh323. That setup seems to drop small audio snippets like VoiceMail's password prompt, though. So I'm trying to give chan_h323 another chance. However, I get: ast_h323.cpp: In function `int h323_set_capability(int, int)': ast_h323.cpp:780: `H323_G729ACapability' undeclared (first use this function) ast_h323.cpp:780: (Each undeclared identifier is reported only once ast_h323.cpp:780: for each function it appears in.) ast_h323.cpp:780: `g729aCap' undeclared (first use this function) ast_h323.cpp:781: parse error before `)' ast_h323.cpp: At top level: chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 This is both with openh323-1.12.0 and their current CVS. (using current CVS snapshot of asterisk, too) Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
On Tue, 10 Jun 2003, Jeremy McNamara wrote: trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) Open H.323 1.11.7 works perfectly in all of my installations. I babysit 15 different chan_h323 based systems. One way audio usually means you have codec problems or are trying to traverse NAT. Nope, things were more weird in this case: This installation has a few hundred Cisco 79xx phones running in Skinny mode babysitted by a Cisco CallManager (actually two CCMs, if you count the fallback machine). Asterisk is used as a voicemail box attached to the CCM as an H.323 gateway. So what happens is: the CCM builds all connections to asterisk but negotiates via H.245 that the actual voice streams should be sent directly to the phone. For some reason, OpenH323 1.11.7 would ignore this and just send packets to the CCM instead, which would just drop them. Hence silence on the phone. The codec is G.711, so no problems here. and everything's running in one big private class B net without any outside connection, so no NAT. [...] It looks like the Open H.323 folks either forgot to include the G.729 Capability stubb or were forced to pull it by their legal department. I will look into this. It's still there, but skipped during compilation. Why, I can't tell. Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? We have never tested the latest cvs -HEAD of Open H.323 and PWLib, as there have been major changes, so we are giving those guys some time to make sure everything is stable before we dive in to new, untested code. I see. So I'll have to stick to chan_oh323 for now. Thanks, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
