Re: [asterisk-users] 1.6.2 beta 1 crash
On Friday 20 March 2009 20:36:38 Sebastian wrote: [Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql: Attempted to update column 'useragent' in table 'sip', but column does not exist! [Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL RealTime: Updating on column 'lastms', but that column does not exist within the table 'sip'! I would like to know if this new fields corresponds to previous ones, so I can delete deprecated ones and add this new to my tables. The useragent field should have been there previously. Now, Asterisk warns you, instead of the query silently failing. The field lastms is new (numeric). localhost*CLI *** glibc detected *** /usr/sbin/asterisk: realloc(): invalid next size: 0x08abff28 *** === Backtrace: = /lib/libc.so.6[0xdfa440] /lib/libc.so.6(realloc+0x1a7)[0xdfb377] /usr/sbin/asterisk(__ast_str_helper+0x75)[0x81403b5] This tells me that you have memory corruption and probably need to run this under valgrind. See doc/valgrind.txt for details. -- Tilghman ___ -- 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] Looking for clues to this error message
As you're using a SIP channel, likely you are not limiting the number of calls. Try setting limitonpeers and call-limit. Hope this helps, l. 2009/3/20 Cary Fitch ca...@usawide.net [Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members don't show busy.. there will be no announcements. However, I see no references to such an issue in any upgrade documents I have found. Any one have a tip? Thanks Cary Fitch ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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] OpenBTS chat with David A. Burgess
Hi, The OpenBTS Project is an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (Um) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world. Last Friday, the VoIP Users Conference had David A. Burgess as our guest for an hour of discussion of one of his projects, OpenBTS (http://openbts.sourceforge.net/) which uses asterisk as a backend for the GSM network. Bookmarks to all the URL given on OpenBTS are here: http://delicious.com/voipusersconference The recorded conference can be hear or downloaded here: http://www.talkshoe.com/talkshoe/web/talkCast.jsp?masterId=22622 Enjoy! ___ -- 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] 1.6.0-rc3 Build failure: asterisk.h: No such file or directory
Trying to build asterisk 1.6.0-rc3, it fails with this message: make[2]: Entering directory `/home/fhimpe/rpm/BUILD/asterisk-1.6.1.0-rc3/ main/editline' /bin/sh makelist -h common.c common.h /bin/sh makelist -h emacs.c emacs.h /bin/sh makelist -h vi.c vi.h /bin/sh makelist -fh common.h emacs.h vi.h fcns.h /bin/sh makelist -fc common.h emacs.h vi.h fcns.c if uname -s | /bin/grep -qi cygwin; then cat fcns.c | sed -e s/sys \.h/config.h/g fcns.c.copy; mv --force fcns.c.copy fcns.c; fi /bin/sh makelist -bh common.c emacs.c vi.c help.h /bin/sh makelist -bc common.c emacs.c vi.c help.c if uname -s | /bin/grep -qi cygwin; then cat help.c | sed -e s/sys \.h/config.h/g help.c.copy; mv --force help.c.copy help.c; fi /bin/sh makelist -e common.c emacs.c vi.c chared.c el.c hist.c key.c map.c parse.c prompt.c read.c refresh.c search.c sig.c term.c tty.c fcns.c help.c editline.c gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. editline.c -o editline.o_a gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. np/fgetln.c -o np/fgetln.o_a gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. np/vis.c -o np/ vis.o_a gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. np/unvis.c -o np/unvis.o_a gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. np/strlcpy.c -o np/strlcpy.o_a gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. np/strlcat.c -o np/strlcat.o_a gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. history.c -o history.o_a gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. tokenizer.c -o tokenizer.o_a gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. readline.c -o readline.o_a readline.c:39:22: error: asterisk.h: No such file or directory What could be wrong here? -- Frederik Himpe ___ -- 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] OpenBTS chat with David A. Burgess
On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote: Hi, The OpenBTS Project is an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (Um) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world. Last Friday, the VoIP Users Conference had David A. Burgess as our guest for an hour of discussion of one of his projects, OpenBTS (http://openbts.sourceforge.net/) which uses asterisk as a backend for the GSM network. See also http://lwn.net/Articles/320163 . Patents hit them really badly. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] OT - CID with Asterisk and Betamax
Hi, sorry for this a bit OT. I'm using VoiceTrading for some calls -premium route- and can't get CID to work despite the fact that CALLERID(num) and CALLERID(name) are setted. I ask in VT-myAccount to accept calls from my IP without checking username secret. On incoming calls the CID is setted to 01 If I accept calls from username secret and no IP relation, CID is always different and not related to something I know, doesn't matter which country is called. I read in some forums that setting secret=secret:username and username=CID should work. I tried, doesn't work. Anyway it wouldn't solve the problem, I would have to dynamicaly change the username (how to do this?), as the CID is depending on the current call. Has anyone of you have success with Voicetrading or others Betamax GWs to set there own CID? If yes, would you share? Ah, yes, already tried to contact their support, but like all Betamax Services, no answer. And yes, I know, use other GWs -what I do-, but their prices are very competitive and my account is loaded. Regards, thanks for any hint. -- Daniel ___ -- 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] 1.6.0-rc3 Build failure: asterisk.h: No such file or directory
On Saturday 21 March 2009 06:33:57 Frederik Himpe wrote: Trying to build asterisk 1.6.0-rc3, it fails with this message: snip What could be wrong here? What's wrong here is that you're compiling a release candidate that is 6 months old and the problems associated with it have been long since corrected. If you're keen on trying a release candidate, please try asterisk-1.6.0.7-rc2, or if not, try asterisk-1.6.0.6. -- Tilghman ___ -- 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] 1.6.2 beta 1 crash
If I compile with DON'T OPTIMIZE and MALLOC DEBUG realtime stop working, it says that the query was empty to consult realtime, if I go back to those parameters not checked starts working ok again. Tell me if you need that with valgrind also, or other information, debug info on the cli doesnt give me much info to trace the problem. Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: sábado, 21 de marzo de 2009 03:54 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.6.2 beta 1 crash On Friday 20 March 2009 20:36:38 Sebastian wrote: [Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql: Attempted to update column 'useragent' in table 'sip', but column does not exist! [Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL RealTime: Updating on column 'lastms', but that column does not exist within the table 'sip'! I would like to know if this new fields corresponds to previous ones, so I can delete deprecated ones and add this new to my tables. The useragent field should have been there previously. Now, Asterisk warns you, instead of the query silently failing. The field lastms is new (numeric). localhost*CLI *** glibc detected *** /usr/sbin/asterisk: realloc(): invalid next size: 0x08abff28 *** === Backtrace: = /lib/libc.so.6[0xdfa440] /lib/libc.so.6(realloc+0x1a7)[0xdfb377] /usr/sbin/asterisk(__ast_str_helper+0x75)[0x81403b5] This tells me that you have memory corruption and probably need to run this under valgrind. See doc/valgrind.txt for details. -- Tilghman ___ -- 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 Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 8.5.278 / Base de datos de virus: 270.11.21/2014 - Fecha de la versión: 03/20/09 06:59:00 ___ -- 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] OpenBTS chat with David A. Burgess
Tzafrir Cohen wrote: On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote: Hi, The OpenBTS Project is an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (Um) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world. Last Friday, the VoIP Users Conference had David A. Burgess as our guest for an hour of discussion of one of his projects, OpenBTS (http://openbts.sourceforge.net/) which uses asterisk as a backend for the GSM network. See also http://lwn.net/Articles/320163 . Patents hit them really badly. I'm not sure how many patents might still be in force on the original GSM design, but I think you'll find the issues surrounding this case are not patents. I think Martone are claiming the OpenBTS guys reused designs developed under an exclusive contract with Martone. That is, its about their trade secrets, and not about building a GSM system. Steve ___ -- 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] 1.6.2 beta 1 crash
On Saturday 21 March 2009 10:10:22 Sebastian wrote: If I compile with DON'T OPTIMIZE and MALLOC DEBUG realtime stop working, it says that the query was empty to consult realtime, if I go back to those parameters not checked starts working ok again. I need to see the exact error. Tell me if you need that with valgrind also, or other information, debug info on the cli doesnt give me much info to trace the problem. You can avoid MALLOC_DEBUG, but DONT_OPTIMIZE is pretty much required to get useful information. Also, debugging this would be more appropriate if you filed a bug on http://bugs.digium.com, instead of discussing it here. This also allows us to track the issue and ensure it doesn't fall through the cracks. -- Tilghman ___ -- 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] 1.6.0-rc3 Build failure: asterisk.h: No such file or directory
On Sat, 21 Mar 2009 09:35:34 -0500, Tilghman Lesher wrote: On Saturday 21 March 2009 06:33:57 Frederik Himpe wrote: Trying to build asterisk 1.6.0-rc3, it fails with this message: snip What could be wrong here? What's wrong here is that you're compiling a release candidate that is 6 months old and the problems associated with it have been long since corrected. If you're keen on trying a release candidate, please try asterisk-1.6.0.7-rc2, or if not, try asterisk-1.6.0.6. Err, no. I made a mistake in the subject which you would have noticed if you had read the output I posted: this is asterisk-1.6.1.0-rc3, released yesterday. -- Frederik Himpe ___ -- 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] 1.6.2 beta 1 crash
Issue: 0014716 With just DON't OPTIMIZE Works realtime but func_odbc doesn't crash. I put the message of realtime not working with debug malloc on the bug. Let me know if you need more info. Regards Sebastian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: sábado, 21 de marzo de 2009 12:58 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.6.2 beta 1 crash On Saturday 21 March 2009 10:10:22 Sebastian wrote: If I compile with DON'T OPTIMIZE and MALLOC DEBUG realtime stop working, it says that the query was empty to consult realtime, if I go back to those parameters not checked starts working ok again. I need to see the exact error. Tell me if you need that with valgrind also, or other information, debug info on the cli doesnt give me much info to trace the problem. You can avoid MALLOC_DEBUG, but DONT_OPTIMIZE is pretty much required to get useful information. Also, debugging this would be more appropriate if you filed a bug on http://bugs.digium.com, instead of discussing it here. This also allows us to track the issue and ensure it doesn't fall through the cracks. -- Tilghman ___ -- 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 Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 8.5.278 / Base de datos de virus: 270.11.21/2014 - Fecha de la versión: 03/20/09 06:59:00 ___ -- 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] 1.6.0-rc3 Build failure: asterisk.h: No such file or directory
On Sat, Mar 21, 2009 at 11:33:57AM +, Frederik Himpe wrote: Trying to build asterisk 1.6.0-rc3, it fails with this message: gcc -c -O2 -g -pipe -Wformat -Werror=format-security -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -Werror-implicit-function-declaration -O '-D__RCSID(x)=' '- D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. readline.c -o readline.o_a readline.c:39:22: error: asterisk.h: No such file or directory Here is how it was run for me: gcc -c -pthread -I../..//include -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -march=k8 -ffunction-sections -O6 -O '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. readline.c -o readline.o_a Note the extra '-I../..//include' . What is the output of: ls -l main/editline/Makefile* grep ^CFLAGS main/editline/Makefile* My wild guess: somebody tried to overrie Asterisk's, well, let's say, strange, -O6, and did it wrong. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] 1.6.0-rc3 Build failure: asterisk.h: No such file or directory
Are you bulding it from rpmbuild ? The error says it can't find the asterisk.h so it's most likely a Makefile/paths error. go to readline.c to where it's trying to #include asterisk.h and fix it there :) Martin On Sat, Mar 21, 2009 at 11:06 AM, Frederik Himpe fhi...@telenet.be wrote: On Sat, 21 Mar 2009 09:35:34 -0500, Tilghman Lesher wrote: On Saturday 21 March 2009 06:33:57 Frederik Himpe wrote: Trying to build asterisk 1.6.0-rc3, it fails with this message: snip What could be wrong here? What's wrong here is that you're compiling a release candidate that is 6 months old and the problems associated with it have been long since corrected. If you're keen on trying a release candidate, please try asterisk-1.6.0.7-rc2, or if not, try asterisk-1.6.0.6. Err, no. I made a mistake in the subject which you would have noticed if you had read the output I posted: this is asterisk-1.6.1.0-rc3, released yesterday. -- Frederik Himpe ___ -- 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
[asterisk-users] H323plus homepage down?
Anybody knows why is down? Or if has been moved to another page?? ___ -- 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] [asterisk-biz] OpenBTS chat with David A. Burgess
On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote: Hi, The OpenBTS Project is an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (Um) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world. This looks like a great project, sorry I missed the call. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- 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] [asterisk-biz] OpenBTS chat with David A. Burgess
They gave a presentation at the CCC around newyear, you might be able to find the videostream somewhere. Zoa Steve Kennedy wrote: On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote: Hi, The OpenBTS Project is an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (Um) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world. This looks like a great project, sorry I missed the call. Steve ___ -- 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] music-on-hold kicks in and disconnects/interrupt the call
I'm using Asterisk 1.4.22.1 When I'm on active call it happens many times the call gets interrupted by music-on-hold without my pressing any button. MOH just kicks in and int erupt the call and I have no way of getting the call back. Did anybody experienced anything like this? -- Joseph ___ -- 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] Digium and Sangoma Cards PCI express compatibility
Hi to All, I dont know much about PCI express slots in newer Servers, my doubt is if the Digium and Sangoma PCI express cards, are compatible with the x8 PCI express slots that come in the HP Proliant ML150 G5 server. Thanks in advance. Ricardo ___ -- 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] Asterisk on iMac G3 Debian5 (powerpc)
I've recently installed the latest Debian Linux for powerpc onto and old iMac (version A) the original iMac with a 233Mhz G3 processor and 160MB of sdram. The debian install went smooth and so the the apt-get install of Asterisk 1.4.21 It appears to have no functioning zaptel or ztdummy module. Is because of hardware? or is it because whoever built the package didn't include a full working zaptel with ztdummy? In other words is zaptel ztdummy supported on this old mac hardware? Can I re-compile from scratch and get it all to work? or is there are hardware limitation or lack leaving this with no hardware timer options? It's on a 2.6 kernel and very up to date. Having a full working asterisk pbx on and old iMac is something everyone should experience at least once in their life! :-) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Polycom BLF with Idle State meetme conference
Just watned to kick this back out here one more time. Anyone done this or ever looked into a work-around? Thanks! Steve I have meetme working with BLF on polycom phones however when meetme is not actually being used by anyone the 'status' of meetme becomes idle. Which the Polycom phone sees and produces a clock symbol and FLASHING red LED. Are there any 'tricks' or work-arounds to change this status to something that does not blink the phone's LED making it look busy when meetme is idle? With timed out SIP registrations setting a default IP address for the friend always did the trick I'm wondering if there is a nice one like that to use on meetme? Thanks!!! Steve 4...@mbb : meetme:4400 State:Idle Watchers 1 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Asterisk on iMac G3 Debian5 (powerpc)
On Sun, Mar 22, 2009 at 12:09:09AM -0400, Steve Gladden wrote: I've recently installed the latest Debian Linux for powerpc onto and old iMac (version A) the original iMac with a 233Mhz G3 processor and 160MB of sdram. The debian install went smooth and so the the apt-get install of Asterisk 1.4.21 It appears to have no functioning zaptel or ztdummy module. apt-get install zaptel-source m-a a-i zaptel -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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
