Re: [asterisk-users] Top Posting
It is not a matter of preference, it is actually a rule [1]. Top-posting is also an annoying practice [2] and NOT the general accepted way to reply. And that's been the case for at least TWO DECADES. I find it amazing that this is still being argued now. -- _ -- 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
Re: [asterisk-users] Asterisk stops responding
On Jan 15, 2011, at 2:01 AM, Carlos Chavez wrote: The problem is that Asterisk simply stops responding. No calls in or out and you cannot even get to the CLI. The process seems to be running but there is simple no activity. All I see in the log files is: [Jan 14 16:30:46] WARNING[20747] pbx.c: Failed to create new channel thread [Jan 14 16:30:46] WARNING[20747] chan_sip.c: Failed to start PBX :( [Jan 14 16:30:47] WARNING[20745] pbx.c: Failed to create new channel thread [Jan 14 16:30:47] WARNING[20745] chan_dahdi.c: Unable to start PBX on DAHDI/29-1 After restarting Asterisk everything is back to normal. The time between the first failure and the second was almost a month, between the second and third a few days. Carlos, What is in the logs immediately preceding the warning you have posted here? Scan up a number of lines (more if you have a very verbose installation, like FreePBX) and see if anything pops out at you. Basically, you want to figure out what was happening on the server at the time of the crash? Incoming fax? Hangup of a Dahdi channel? Incoming Dahdi call, etc. That will likely point you in the right direction. Tom -- _ -- 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] Music on Hold not working?
I'm going out on a limb here, as I'm still pretty new to Astrisk and running my own VOIP server, however I believe there is a bug or flaw with the Music on Hold feature. I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. I have tried noload = res_timing_dahdi.so in the etc/asterisk/modules.conf file, however that doesn't work. I've tried moving that line to the very top, very bottom, and several places in the middle, and still can not get MOH to work. I don't use dahdi, nor do I use the local Telco lines. It is strictly a VOIP trunk. Yes I know there are advantages and disadvantages to this kind of set up, however, I have a failover route set up and configured with the trunk provider, so I am ok with what I have at this point. Currently AstriskNOW/FreePBX is installed on: Dell precision 490 workstation Dual xeon 5100 dual core processors 4gb ECC Buffered Ram 73GB 15krpm hard drive The Machine has Xen server installed with asterisknow running on its own VM with: 2 cores 2gb ram 40gb hard drive space I installed AsteriskNow 64bit image that is available for download from Freepbx's website, and have done all of the updates that both the freepbx interface, as well as CLI yum update command suggests with no fixing of the problem. I thank you all in advance for taking the time to read this issue and look forward to hopefully fixing my MOH. Warmest Regards, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. -- _ -- 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
Re: [asterisk-users] Top Posting
Paul Belanger wrote: It is not a matter of preference, it is actually a rule [1]. Top-posting is also an annoying practice [2] and NOT the general accepted way to reply. [1] http://www.asterisk.org/community/rules [2] http://linux.sgms-centre.com/misc/netiquette.php#toppost Thanks for pointing out the rule ([1] #5); this is a more effective way of communicating the list's etiquette than the snide asides that we generally see. It also conforms to the rule ([1] #1) that says posts should be considerate and respectful. With regard to my preference, I stand by my explanation in my first post in this thread. I think the main argument for bottom-posting is we've been doing it that way for decades. That said, of course I want to follow this list's etiquette. I've posted a couple times asking how I can interleave responses in Outlook or what other approach can I take to make it practical to stop top-posting. Any suggestions? -- --Don -- _ -- 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
Re: [asterisk-users] Top Posting
On Jan 15, 2011, at 9:29 AM, Don Kelly wrote: That said, of course I want to follow this list's etiquette. I've posted a couple times asking how I can interleave responses in Outlook or what other approach can I take to make it practical to stop top-posting. Any suggestions? Don: Outlook-QuoteFix: http://home.in.tum.de/~jain/software/outlook-quotefix/ I found that program last night after reading one of the pages linked in this thread. The program isn't supported on OL 2007 and newer, but there is a link on the page to a macro for newer versions. Wish I had known about it years ago! Also, http://mailformat.dan.info/config/outlook.html shows the general steps needed to make Outlook approximate standards. HTH, Tom -- _ -- 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
Re: [asterisk-users] Top Posting
Tom Rymes wrote: On Jan 15, 2011, at 9:29 AM, Don Kelly wrote: That said, of course I want to follow this list's etiquette. I've posted a couple times asking how I can interleave responses in Outlook or what other approach can I take to make it practical to stop top-posting. Any suggestions? Don: Outlook-QuoteFix: http://home.in.tum.de/~jain/software/outlook-quotefix/ I found that program last night after reading one of the pages linked in this thread. The program isn't supported on OL 2007 and newer, but there is a link on the page to a macro for newer versions. Wish I had known about it years ago! Also, http://mailformat.dan.info/config/outlook.html shows the general steps needed to make Outlook approximate standards. HTH, Tom Thanks. As far as making bottom posting work, http://home.in.tum.de/~jain/software/outlook-quotefix/ makes it so much simpler and better! I just installed it, and will try to keep using it. If I don't want the previous material at the top, I will delete it. But, personally, I really prefer top posting or the previous material deleted. However... Time will tell. Cary -- _ -- 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
Re: [asterisk-users] Music on Hold not working?
I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from moh show files in Asterisk? -- _ -- 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] Problem with chan_dahdi and conferencing
Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load = chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) Now chan_dahdi is indeed loaded, but I have an empty chan_dahdi.conf. What do I need to do to get recording to work? Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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
Re: [asterisk-users] Music on Hold not working?
On Sat, Jan 15, 2011 at 9:36 AM, James Miller paramedi...@gmail.com wrote: Forgive me, but how do I do moh show files? Basically what is occurring is: If you enter a queue and are waiting to be answered, you will hear the streaming MOH If you call another extension on the system, you will only hear the default MOH. I want it to stream MOH for everything. Hopefully that makes sense. Regards, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gary Allen *Sent:* Saturday, January 15, 2011 11:33 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Music on Hold not working? I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from moh show files in Asterisk? Go into Asterisk CLI (asterisk -r) and issue the command moh show files. I don't see how you can have different MOH in a queue vs. being on hold unless you have specified a specific MOH group for your call queues. -- _ -- 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
Re: [asterisk-users] Bruce B
On Fri, Jan 14, 2011 at 6:31 PM, Tim Nelson tnel...@fudnet.net wrote: You've been officially added to my kill file [1]. The lists are here to get suggestions and assistance with various issues [2]. They are *NOT* your one stop shop for everyone doing your homework [3][4][5][6][7][8][9]. You make it abundantly clear that you're making no effort whatsoever to find answers to the questions you post. And, rather than listen to answers given, or even suggestions about your list etiquette, you instead choose to ignore those suggestions and ask more questions [10]. AND, to make matters worse, this isn't the only list you actively abuse [11][12][13]. Also, since you're unable to seek information on your own, I've taken the liberty of keeping references to all of the above points for you. If I were a mod, I'd drop you from the list. But alas, pushing your useless drivel to /dev/null will have to suffice [14]. I'll just sit here listening to a very relevant song [15] while I get back to the regularly scheduled programming. --Tim [1] http://en.wikipedia.org/wiki/Kill_file [2] http://en.wikipedia.org/wiki/Mailing_list [3] http://lists.digium.com/pipermail/asterisk-users/2011-January/257684.html [4] http://lists.digium.com/pipermail/asterisk-users/2011-January/257685.html [5] http://lists.digium.com/pipermail/asterisk-users/2011-January/257762.html [6] http://lists.digium.com/pipermail/asterisk-users/2011-January/257832.html [7] http://lists.digium.com/pipermail/asterisk-users/2011-January/257888.html [8] http://lists.digium.com/pipermail/asterisk-users/2011-January/257962.html [9] http://lists.digium.com/pipermail/asterisk-users/2011-January/257992.html [10] http://lists.digium.com/pipermail/asterisk-users/2011-January/257991.html [11] http://www.mail-archive.com/support@pfsense.com/msg21300.html [12] http://www.mail-archive.com/support@pfsense.com/msg21307.html [13] http://www.mail-archive.com/support@pfsense.com/msg21119.html [14] http://en.wikipedia.org/wiki//dev/null [15] http://en.wikipedia.org/wiki/Don't_Go_Away_Mad_(Just_Go_Away) http://en.wikipedia.org/wiki/Don%27t_Go_Away_Mad_%28Just_Go_Away%29 -- _ -- 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 Hulk smash? -- _ -- 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] Sound quality issue
Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check on his network equipments, everything is fine too, no packets loss recorded on routers's interfaces ect ... We have, on our side, check and replace all the VOIP equipments (spare rocks), an reduce the configuration to its simpliest (MPLS router = ethernet cable = VOIP equipment), quality problem still there. I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? Any help would be greatly appreciated, thx. Cédric -- _ -- 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
Re: [asterisk-users] Sound quality issue
Hello, Can you record audio at different locations on its route? Our experience would suggest (of course) using intrusive or non-intrusive perceptual voice quality evaluation at different parts of the network to localize the one where it drops down. Best regards, Sevana Oy http://www.sevana.fi http://twitter.com/sevana - Original Message - From: Cédric Lemarchand cedric.lemarch...@ixcore.com To: asterisk-users@lists.digium.com Sent: Saturday, January 15, 2011 10:38 PM Subject: [asterisk-users] Sound quality issue Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check on his network equipments, everything is fine too, no packets loss recorded on routers's interfaces ect ... We have, on our side, check and replace all the VOIP equipments (spare rocks), an reduce the configuration to its simpliest (MPLS router = ethernet cable = VOIP equipment), quality problem still there. I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? Any help would be greatly appreciated, thx. Cédric -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Sound quality issue
Le 15/01/11 20:50, Sevana Oy a écrit : Hello, Can you record audio at different locations on its route? Our experience would suggest (of course) using intrusive or non-intrusive perceptual voice quality evaluation at different parts of the network to localize the one where it drops down. Yes we already do some records. We don't have access to the internal network of the provider, so the network topology is quiet simple, only 2 sides : Asterisk site = MPLS NETWORK (the provider) = remote site Sound quality problems are present on both sent and received RTP flow. Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check on his network equipments, everything is fine too, no packets loss recorded on routers's interfaces ect ... We have, on our side, check and replace all the VOIP equipments (spare rocks), an reduce the configuration to its simpliest (MPLS router = ethernet cable = VOIP equipment), quality problem still there. I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? Any help would be greatly appreciated, thx. Cédric -- _ -- 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
Re: [asterisk-users] Sound quality issue
I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? If it is possible to make a network trace in a Wireshark compatible format, Wireshark can parse all the SIP and RTP messaging and give you lots of statistics, including packet loss, jitter, etc. Check the Wireshark site (http://www.wireshark.org/) for more information. -- Andreas Sikkema -- _ -- 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] app_fax watchdog timeout
Hello all, In app_fax WATCHDOG_TOTAL_TIMEOUT is set to 30 minutes to kill a fax channel regardless of whether or not it completes. In my case I have a fax that really would take longer than 30 minutes to complete. Is there any way to disable the WATCHDOG_TOTAL_TIMEOUT so that it runs to completion? Thanks, Eric -- _ -- 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
Re: [asterisk-users] Music on Hold not working?
From the command you suggested to enter: Class: default File: /var/lib/asterisk/moh//reno_project-system File: /var/lib/asterisk/moh//macroform-robot_dity File: /var/lib/asterisk/moh//manolo_camp-morning_coffee File: /var/lib/asterisk/moh//macroform-cold_day File: /var/lib/asterisk/moh//macroform-the_simplicity Class: none File: /var/lib/asterisk/moh/.nomusic_reserved/silence Basically the queues will stream the online music, but if I call another extension on the network, it will play just the default sounds. One would think that if you have suggested the system play streaming music for everything else, it would follow suite and play streaming for ext to ext calls. I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Allen Sent: Saturday, January 15, 2011 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold not working? On Sat, Jan 15, 2011 at 9:36 AM, James Miller paramedi...@gmail.com wrote: Forgive me, but how do I do moh show files? Basically what is occurring is: If you enter a queue and are waiting to be answered, you will hear the streaming MOH If you call another extension on the system, you will only hear the default MOH. I want it to stream MOH for everything. Hopefully that makes sense. Regards, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Allen Sent: Saturday, January 15, 2011 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold not working? I have it all configured and it should work, and it did briefly several weeks ago, however now, it doesn't work at all and only plays the default hold music. If it is playing the default music, then the MOH function is working. What do you get from moh show files in Asterisk? Go into Asterisk CLI (asterisk -r) and issue the command moh show files. I don't see how you can have different MOH in a queue vs. being on hold unless you have specified a specific MOH group for your call queues. -- _ -- 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 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
Re: [asterisk-users] Problem with chan_dahdi and conferencing
On Sat, Jan 15, 2011 at 11:50:32AM -0500, cov...@ccs.covici.com wrote: Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load = chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) What is the output of: dahdi show channels Now chan_dahdi is indeed loaded, but I have an empty chan_dahdi.conf. Not even an empty '[channels]' section? -- 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 -- 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
Re: [asterisk-users] Problem with chan_dahdi and conferencing
Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sat, Jan 15, 2011 at 11:50:32AM -0500, cov...@ccs.covici.com wrote: Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load = chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) What is the output of: dahdi show channels Now chan_dahdi is indeed loaded, but I have an empty chan_dahdi.conf. Not even an empty '[channels]' section? I did put that just now, but I still get the same warning. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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
