Re: [asterisk-users] Delay between answer and pickup ?
Anyone got a clue ? (he asks in desperation!) Julian On 9 July 2010 17:48, Julian Lyndon-Smith aster...@dotr.com wrote: We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing hello ? Hello ? and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of asterisk don't show this discrepancy. We are using a local channel to dial a landline through a sip provider. When the call is answered, the agent's phone is then dialled. the logs go something like this [Jul 9 13:29:26] VERBOSE[23396] logger.c: [Jul 9 13:29:26] -- SIP/provider-0001ed6e is making progress passing it to Local/somenum...@dialleroutbound-4c93,2 [Jul 9 13:29:44] VERBOSE[23396] logger.c: [Jul 9 13:29:44] -- SIP/provider-0001ed6e answered Local/01577864...@dialleroutbound-4c93,2 .. [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Executing [*00...@diallerconnected:2] Dial(Local/somenum...@dialleroutbound-4c93,1, SIP/*0086*|5|iA(autoanswer)) in new stack [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- SIP/*0086*-0001ed73 answered Local/somenum...@dialleroutbound-4c93,1 .. as you can see, the call is answered at 13:29:44 and the agent gets called (auto-answer phones) at 13:29:46, yes if you listen to the call recording, there is a 6 second gap between the person saying hello and the agent being connected. Is it possible that the call was answered 5 seconds *before* I get notification of the answer ? i.e. is the provider taking too long notifying me of the answer ? Julian -- _ -- 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] Is a device a member of a queue?
I'm looking for a function I can put in my dial plan that tells me if
a device is a member of a queue, but I can seem to find one. Basically
I want to be able to dial to join a queue and if I'm already on the
queue, leave..
exten = 4,1,GotoIf(${is_queue_member(queuename,SIP/${ext})}?leave:join)
exten = 4,n(leave),RemoveQueueMember(queuename,SIP/${ext})
exten = 4,n,Hangup
exten = 4,n(join),AddQueueMember(queuename,SIP/${ext})
or simular, If such a function exists it would be very handy
The only way I can see of doing this is to use
queue_member_list(queue) and then loop through the returned list using
cut searching for the device.
So.
1. Is there a function I'm missing to do this say..
is_queue_member(queuename,channel)
or
2. Is there some way of creating such a function.
Thanks in advanced
Peter Childs
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Re: [asterisk-users] [Dahdi 2.3.0.1] Does it need all those modules?
On Fri, 09 Jul 2010 13:45:18 -0500, Shaun Ruffell sruff...@digium.com wrote: # lsmod | grep -i wc wctc4xxp 32414 0 dahdi_transcode 5751 1 wctc4xxp wcb4xxp33905 0 wcfxo 8968 0 wctdm24xxp116684 0 wcte11xp 22995 0 wct1xxp12971 0 wcte12xp 26308 0 dahdi_voicebus 39947 2 wctdm24xxp,wcte12xp wct4xxp 230713 0 wctdm 35677 0 dahdi 197809 11 xpp,dahdi_transcode,wcb4xxp,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp,wctdm crc_ccitt 1339 3 wctdm24xxp,dahdi,hisax == Does Dahdi really need all those modules, or is there another configuration file that I missed to disable unneeded modules? /etc/dahdi/modules controls which modules /etc/init.d/dahdi will load on start. Thanks Shaun. I edited /etc/dahdi/modules thusly and will try later to see if Dahdi still works as intended: # lsmod | grep -i wc wctdm 35677 0 dahdi 197809 1 wctdm -- _ -- 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] Couple of questions about modules
On Sat, 3 Jul 2010 13:47:23 -0500, Tilghman Lesher tles...@digium.com wrote: (snip) Thanks much for the education. -- _ -- 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] Dialplan question when using a round-robin
Hi all,
i have a question regarding the dialplan when using a DNS round robin for
simple load balancing.
When i have 3 identically configured Asterisk servers and one DNS round robin
populated to the clients
Server1
Server2 - round robin voip.example.com
Server3
where all 3 servers are connected via an IAX2 trunk among each other, how can i
determine within the dialplan on which of the servers a client is actually
registered?
So when user A wants to call user B and user A is registered on server1 and
user B is registered on server2, afaik the dialplan would need to look like
Dial(IAX2/server2/${EXTEN})
But how can i determine on which physical server user B is registered?
Or is there an other, better way to achieve this? Maybe in replicating the
registrations between all 3 servers?
Thanks in advance for advise.
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Re: [asterisk-users] Is a device a member of a queue?
On Sun, Jul 11, 2010 at 5:40 AM, Peter Childs pchi...@bcs.org wrote: 1. Is there a function I'm missing to do this say.. is_queue_member(queuename,channel) *CLI core show function QUEUE_MEMBER -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Dialplan question when using a round-robin
On Sun, Jul 11, 2010 at 8:00 AM, unsero...@aol.com wrote: But how can i determine on which physical server user B is registered? Or is there an other, better way to achieve this? Maybe in replicating the registrations between all 3 servers? DUNDi is an options, same with DNS SRV records. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Is a device a member of a queue?
On 11 July 2010 14:19, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Jul 11, 2010 at 5:40 AM, Peter Childs pchi...@bcs.org wrote: 1. Is there a function I'm missing to do this say.. is_queue_member(queuename,channel) *CLI core show function QUEUE_MEMBER No function by that name registered. also its not listed on voip-info, I'm using SARK/Asterisk 1.4.21 Peter -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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 -- _ -- 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] Is a device a member of a queue?
On Sun, Jul 11, 2010 at 9:28 AM, Peter Childs pchi...@bcs.org wrote: No function by that name registered. also its not listed on voip-info, I'm using SARK/Asterisk 1.4.21 The function is in 1.6.2. Best you could do in 1.4 is: *CLI core show function QUEUE_MEMBER_LIST FYI: voip-info is terribly out of date. Always best to look in your CLI. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Dialplan question when using a round-robin
But how can i determine on which physical server user B is registered? Or is there an other, better way to achieve this? Maybe in replicating the registrations between all 3 servers? DUNDi is an options, same with DNS SRV records. -- Could you please give me some more info? Or is there a tutorial available somewhere? = -- _ -- 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] PHP can't insert - Can someone please help
Bruce, These two links may be helpful to you: PHP: SQL Injection - Manual http://www.php.net/manual/en/security.database.sql-injection.php PHP: mysql_real_escape_string - Manual http://www.php.net/manual/en/function.mysql-real-escape-string.php Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Dialplan question when using a round-robin
as a quick option you can use
Dial(IAX2/server1/${EXTEN}IAX2/server2/${EXTEN}IAX2/server3/${EXTEN})
call will connect to whichever is available answer, others simply ignored!
On Sun, Jul 11, 2010 at 8:09 PM, unsero...@aol.com wrote:
But how can i determine on which physical server user B is registered?
Or is there an other, better way to achieve this? Maybe in replicating the
registrations between all 3 servers?
DUNDi is an options, same with DNS SRV records.
--
Could you please give me some more info?
Or is there a tutorial available somewhere?
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Re: [asterisk-users] Dialplan question when using a round-robin
On Sun, Jul 11, 2010 at 8:09 PM, unsero...@aol.com wrote:
But how can i determine on which physical server user B is registered?
Or is there an other, better way to achieve this? Maybe in replicating the
registrations between all 3 servers?
DUNDi is an options, same with DNS SRV records.
--
Could you please give me some more info?
Or is there a tutorial available somewhere?
-
as a quick option you can use
Dial(IAX2/server1/${EXTEN}IAX2/server2/${EXTEN}IAX2/server3/${EXTEN})
call will connect to whichever is available answer, others simply
ignored!
-
But wouldn't this only work when the remote party is online?
What would happen if not?
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Re: [asterisk-users] Is a device a member of a queue?
On 11 July 2010 15:56, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Jul 11, 2010 at 9:28 AM, Peter Childs pchi...@bcs.org wrote: No function by that name registered. also its not listed on voip-info, I'm using SARK/Asterisk 1.4.21 The function is in 1.6.2. Best you could do in 1.4 is: *CLI core show function QUEUE_MEMBER_LIST FYI: voip-info is terribly out of date. Always best to look in your CLI. Hmm Yes but http://www.asterisk.org/docs/asterisk/trunk/functions/queue_member says that it just counts the number of members in the list, just like queue_member_count does. queue_member_penality might do what I want, depending on what it actually returns if the given interface is not a member But then I still need 1.6! Peter Childs -- _ -- 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] What‘s the best operating syst em suggest for Asterisk 1.6.2.9
That would be probably because Ubuntu became top-famous and widely used for anything, just fashion so to speak, while CentOS is probably chosen because asterisknow runs on top of centos. On 30 June 2010 12:30, Leif Madsen leif.mad...@asteriskdocs.org wrote: I'm not entirely sure I see where he implied it was. His answer refers to the question, I want to know what is the best OS for installing Asterisk...? I like both CentOS and Ubuntu. The next edition of the O'Reilly Asterisk book will cover installing Asterisk on both OS's. Leif. Tiago Geada wrote: Ubuntu is not Debian. I would recommend Debian tho, its rock solid and it jsut works (for anything) On 29 June 2010 12:29, Paul Belanger paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com wrote: On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun bit...@gmail.com mailto:bit...@gmail.com wrote: i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. Ubuntu 10.04 Server ? -- _ -- 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
[asterisk-users] LIMIT_PLAYAUDIO_CALLEE LIMIT_PLAYAUDIO_CALLER
Hi, Has anyone tried using these flags for the Dial command? I set it to no but both parties can still hear the (beep) warning sound. - Wei -- _ -- 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] need information
Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -- _ -- 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] ztdummy IVR no voice
Hi all , In my pbx ,there is no zaptel card ,so i loading ztdummy,but problem appear,when i dial the number into the pbx,sometimes i can not listen to the ivr ,and no rtp create. if i unload the ztdummy driver,the proble will disapper. I guess may be the timer source problem, but i dont't know how to solve it . anyone can give me some advices will be appreciated. asteirsk-1.4.21 and zaptel-1.4.10 Thanks in advance! -- Best regards! jordan pan Location:Shenzhen China Company:www.justcall.cn -- _ -- 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] need information
Use google or wikipedia if you don't know enough to ask a real question :) On Sun, Jul 11, 2010 at 7:29 PM, mohamed daif mohamed.d...@gmail.com wrote: Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -- _ -- 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] ztdummy IVR no voice
On 7/11/10 9:47 PM, jordan pan wrote: In my pbx ,there is no zaptel card ,so i loading ztdummy,but problem appear,when i dial the number into the pbx,sometimes i can not listen to the ivr ,and no rtp create. if i unload the ztdummy driver,the proble will disapper. I guess may be the timer source problem, but i dont't know how to solve it . anyone can give me some advices will be appreciated. asteirsk-1.4.21 and zaptel-1.4.10 So this only happens sometimes? I'm wondering if ztdummy on your system is using the RTC and that the RTC interrupt is stopping on your system as in issue 13930 (http://issues.asterisk.org/view.php?id=13930) If RTC is in use, you could try making sure that USE_RTC is not defined in ztdummy.c. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Use asterisk as a backend PBX
Hi List, We're planning to use Asterisk as our backend PBX for our legacy PBX where-in received calls from legacy PBX can be transferred to Asterisk PBX extension, is this possible? Regards, Malvin -- _ -- 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] power outage
I have found that sometimes shutting down the machine waiting a full minute while the power cable is unplugged then restarting can fix such problems if it's power related. On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis ge...@pagestation.com wrote: I have a TE205P that has been working fine for 2 years. power outage yesterday took out my everything for over an hour. Everything has come back up except the PRI. My provider has checked it to the box and says everything looks good on their end. I get this message: [Jul 9 12:40:32] WARNING[13709] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! ztcfg -vvv Zaptel Version: 1.4.12.1 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 7 channels to configure. and show status gives me condition RED of course. How do I find out whats wrong here? Jerry -- _ -- 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] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?
In my experience using height for radius works, for example if you have a 20 ft high ceiling then the coverage for one speaker would be 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft has never killed anyone, but this really depends on the power of the speaker, I usually deal with 70v speakers tapped at 16 or 8 watts depending on how many speakers I put on one amplifier and the output wattage of that amplifier. On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. I appreciate your advice. Thanks, Bruce -- _ -- 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] Asterisk Crashes - Segmentation Fault
Unfortunately m not able to get rid of the below mentioned errors. not sure on where i am missing now. On Sat, Jul 10, 2010 at 9:41 AM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Ahh here is the catch i was still using app_cbmysql for this. now i had removed and just followed the README of 4.0 for WMM and m getting following on ,my asterisk console. Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] MeetMe(SIP/492-, ) in new stack -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found [Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2: Failed to connect to meetme [Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No database handle available with the name of 'meetme' (check res_odbc.conf) -- SIP/492- Playing 'conf-invalid.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-' (Initially i installed using yum, i was getting the same issue. Than i scrapped everything and installed it manually.) On Fri, Jul 9, 2010 at 8:39 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] need information
Hello Why do not you want to use A2Billing with Asterisk? I think, for a start, it will give you everything to start a business. -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com mohamed daif wrote: Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -- _ -- 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
