Re: [asterisk-users] CDR - mysql with asterisk 1.2.12 not working
Joseph wrote: On Mon, 2006-10-09 at 01:45 -0300, Hermann Wecke wrote: On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote: I have bind-address = 127.0.0.1 in my.cnf the cdr was working find with asterisk 1.0.1 just after upgrade something is not connecting. I don't know if asterisk will use the localhost or the network IP to connect. Just try to comment your line and see what happens. This is really a guess... Make no difference if I use IP or localhost it is still not connecting; it could be something with the cdr_addon_mysql.so Anybody has any other ideas / suggestions? Have you tried turning on debug in logger.conf. You should be able to see what is wrong from there. Kind Regards Garth ___ --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] Kind of OT : Europeans going to Astricon
Greetings ! Its kind of OT, but if there are any Europeans going to Astricon in Dallas, please send a message of-list. It's possible we will be on the same flight,(i am flying from Frankfurt) ;) so it will be a good way to know it's other and spend some of the 10 hours + flight time . Regards Stelios ___ --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] Kind of OT : Europeans going to Astricon
I will be also on a flight from frankfurt (lufthansa), but a few days early. Zoa. Stelios Koroneos wrote: Greetings ! Its kind of OT, but if there are any Europeans going to Astricon in Dallas, please send a message of-list. It's possible we will be on the same flight,(i am flying from Frankfurt) ;) so it will be a good way to know it's other and spend some of the 10 hours + flight time . Regards Stelios ___ --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] Re: Echo Cancel Cards
On 2006-10-09 22:05:06 -0700, Joseph [EMAIL PROTECTED] said: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. My question: did they artificially introduced echo to sell more hardware? What a preposterous suggestion! Maybe you should try to build/configure your system properly and stop trying to make up silly conspiracy theories? Just a thought. ___ --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] Re: Home Hardware SIP Proxy with use of POTS in Emergency
On 2006-10-09 15:53:36 -0700, Brandon Galbraith [EMAIL PROTECTED] said: Does anyone know of any ATA devices (Linksys, Dlink, Cisco, etc) that will fail over to POTS for an emergency call? I'd like to route any call except a 911 call over SIP or IAX, but any 911 call should be routed out over POTS. If this is not an option, I'm also open to devices that will fail over to GSM to make the emergency call. I apologize if this topic has already been covered before. -brandon The AG168V, which has actually become a pretty nice ATA at this point (thanks to continued firmware updates), can do this for sure. Not only will it fail over nicely if there is a power off, but it also has a built in function that can trap particular calls (ie 911) and send them over PSTN. ___ --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] Re: Asterisk 1.2.12 - Can NOT make call out / Asterisk terminate
On 2006-10-09 05:31:30 -0700, Benny Amorsen [EMAIL PROTECTED] said: PB == Peter Bowyer [EMAIL PROTECTED] writes: PB Fair enough - that's a bit different to 'Asterisk 1.2 is not ready PB for PRIME TIME' though, isn't it? There are plenty of stable 1.2 PB releases, all of which have many fewer bugs than your 1.0.x PB version. Unfortunately they also have security issues. It would be nice if someone made a 1.2.7.2 with the security issues fixed. Either way it is rather unfortunate that the latest version of 1.2 is unstable. That depends on your configuration and usage. Works fine for me on a couple of systems so far... (hope I am not spoiling my luck). ___ --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] Re: SIP stuck channel soft hangup?
On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said: Martin Joseph wrote: I am seeing occasional stuck SIP channels that seem to occur when the fricking Nokia E60 drifts out of WIFI range in the midst of a call. This is particularly annoying when the stuck channels include my PSTN gateway (wellgate 3701a), which leaves incoming and outgoing calls a busy signal. I see by googling that soft hangup is a good way to kill these channels and that works fine for me. I wonder if there is some way to automatically soft hangup these channels when the qualify fails? Take a look at rtptimeout in sip.conf - that might do what you need. Wow! A response! I am thrilled beyond belief ;~) Thanks, I will attempt to look into that. ___ --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] [EMAIL PROTECTED] problems
Probably best change the login and password from the defaults now you've posted this - your admin interface is wide open On 09/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Alex...I do not have FreePBX. What I have is this: http://70.89.124.237/ Ed ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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] Asterisk on 64bit xeon
Hi All, Would asterisk and zaptel compile on 64bit dual xeon hardware?? Rgds From: Martin Joseph [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out /Asterisk terminate Date: Tue, 10 Oct 2006 00:49:30 -0700 On 2006-10-09 05:31:30 -0700, Benny Amorsen [EMAIL PROTECTED] said: PB == Peter Bowyer [EMAIL PROTECTED] writes: PB Fair enough - that's a bit different to 'Asterisk 1.2 is not ready PB for PRIME TIME' though, isn't it? There are plenty of stable 1.2 PB releases, all of which have many fewer bugs than your 1.0.x PB version. Unfortunately they also have security issues. It would be nice if someone made a 1.2.7.2 with the security issues fixed. Either way it is rather unfortunate that the latest version of 1.2 is unstable. That depends on your configuration and usage. Works fine for me on a couple of systems so far... (hope I am not spoiling my luck). ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] SIP vz IAX...
On 9 Oct 2006, at 11:49, raviprakash sunkara wrote: Hello Users. I'm in Dilemma with the performance on SIP and IAX Can any one help ... 1) Difference between the SIP and IAX... which one is Best... in VOIP service I'm using only SIP protocol for my VOIP in OpenSER... And Also I using Asterisk in SIP we can Communicate the SIP and IAX by below scenario SIP (UA) OPENSER - ASTERISK IAX (UA)... this I can do... IAX --- OPENSER - ASTERISK - SIP/IAX. But main problem is ... Suppose IAX -- ASTERISK--- openSER SIP / IAX ... How ? You can't - SER only talks SIP - so the devices on either side of SER in your diagram have to support SIP. you can have IAX -- ASTERISK--- openSER +--- SIP | +---ASTERISK2- IAX I've wondered about the value of a IAX-SIP gateway program, that just acted as a protocol converter, but in the end decided that a cut down asterisk running on an embedded device was easier to deal with. Tim Panton www.mexuar.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] Asterisk RT on Disk On Module Performance andDurability
On 9 Oct 2006, at 21:19, Douglas Garstang wrote: I'm just going to jump in here, and ask a stoopid question. How could you possibly write a multi-user front end in AJAX without using a database backend like MySQL? Wandering off topic here, but I'll bite There are a few options for backing store when a full database is overkill. 0) in-memory - load it all into a hash table at start-up and query that (like asterisk in static mode) 1) carefully structured filesystem, using directory trees for navigation, files for records etc (you can even use symlinks for foreign keys) - you'd be amazed how good the filesystem is as a database :-) 3) xml files with xsl to query the data. This works really well for smallish files that don't change much. Each file change requires a re-write of the whole file, so it is inefficient for writes 4) variation on 0 - one of the in-memory relational databases I used (don't now) to take the view that anyone who didn't need oracle didn't need a real database, so one of the above would be adequate The option discussed earlier of mysql on a ram disk is a variation of 4 I guess. Tim Panton www.mexuar.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] connecting multiple servers with iax - authentication fails
On 9 Oct 2006, at 17:36, Benko wrote: Hello! I'm having a problem which actually looks banal. I'm trying to connect 3 servers via iax with each other. However, i've not been successfull so far. Asterisk always tries to authenticate the calling user with the credentials of the last entry in iax.conf, not the ones that would actually belong to the calling user. e.g. Server1 has peer/user entries for Server2 and Server3(in this order), Server2 now tries to call Server1, but is asked for the credentials of Server3(Because Server3 is the last entry in iax.conf), which doesn't work of course. The IAX debug for this example is attached(iax_server2.txt). Please also take a look at the attached iax.conf-files for each server, maybe i've missed some setting... Currently i workaround this issue by using the same secret for all servers, this is not very practicable however... The asterisk versions in use are 1.2.9.1 on server3 and server2 and 1.4.0-beta2 on server1. This guy seems to have had the same problem, unfortunately he received no answer: http://lists.digium.com/pipermail/asterisk-users/2003-August/ 011960.html thx christian iax.conf.server1.txt iax.conf.server2.txt iax.conf.server3.txt iax_debug_server2.txt It is a bit hard to tell what is going on because you have blanked the IP addresses in the config files to all the same value. If you specify an IP address in the host= line , asterisk will use the from IP address of a 'new' to try and find a matching entry, and ignore the username sent in the message. At a guess you have the IP addresses of servers 1 or 3 wrong in iax.conf on server2 Tim. Tim Panton www.mexuar.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] Asterisk on 64bit xeon
Yes Akpome Akpoguma wrote: Hi All, Would asterisk and zaptel compile on 64bit dual xeon hardware?? Rgds From: Martin Joseph [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out /Asterisk terminate Date: Tue, 10 Oct 2006 00:49:30 -0700 On 2006-10-09 05:31:30 -0700, Benny Amorsen [EMAIL PROTECTED] said: PB == Peter Bowyer [EMAIL PROTECTED] writes: PB Fair enough - that's a bit different to 'Asterisk 1.2 is not ready PB for PRIME TIME' though, isn't it? There are plenty of stable 1.2 PB releases, all of which have many fewer bugs than your 1.0.x PB version. Unfortunately they also have security issues. It would be nice if someone made a 1.2.7.2 with the security issues fixed. Either way it is rather unfortunate that the latest version of 1.2 is unstable. That depends on your configuration and usage. Works fine for me on a couple of systems so far... (hope I am not spoiling my luck). ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] OT: Hand free solution recommandation
Morning all, We're looking for hand free solution to use with Asterisk beside BT headsets. I was thinking on Sipura 841 but it seems that the headset jack connector is not carrying voice (microphone), only audio. Ideal would be a headset audio+microphone with RJ11 4p female that we could plug into the handset cable of any IP phone, or a converter 2xjack2,5mm female RJ11 4p female -which seems not to exist-. What are you recommanding/using/installing in such case? Regards -- Daniel ___ --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] Asterisk on 64bit xeon
Hi All, Would asterisk and zaptel compile on 64bit dual xeon hardware?? Rgds From: Martin Joseph [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk 1.2.12 - Can NOT make call out /Asterisk terminate Date: Tue, 10 Oct 2006 00:49:30 -0700 On 2006-10-09 05:31:30 -0700, Benny Amorsen [EMAIL PROTECTED] said: PB == Peter Bowyer [EMAIL PROTECTED] writes: PB Fair enough - that's a bit different to 'Asterisk 1.2 is not ready PB for PRIME TIME' though, isn't it? There are plenty of stable 1.2 PB releases, all of which have many fewer bugs than your 1.0.x PB version. Unfortunately they also have security issues. It would be nice if someone made a 1.2.7.2 with the security issues fixed. Either way it is rather unfortunate that the latest version of 1.2 is unstable. That depends on your configuration and usage. Works fine for me on a couple of systems so far... (hope I am not spoiling my luck). ___ --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 _ Don't just search. Find. Check out the new MSN Search! http://search.msn.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] Lots and lots of log files
On Mon, Oct 09, 2006 at 01:50:11PM -0400, J. Oquendo wrote:
Edit /etc/asterisk/logger.conf then asterisk -rx logger reload or in
cron the dirty way:
0 * * * * ls -ltha /var/log/asterisk/|awk '{print $5,$9}'|grep 0
|grep -v [1-9]|xargs rm -rf
Huh?
Is it supposed to pick files in the csv dirs?
No: His original post:
In my /var/log/asterisk directory I have 492,018 log files, most of which
are empty.
event_log.XXX queue_log.XXX messages.XXX where XXX is an integer.
In this case, using xargs would exceed the maximal command-line length
(about 128kb on Linux).
Use a loop: for file in messages.* queue_log.*; do rm file; done
Slower, but would work.
One hint: check the log rotation configuration. Don't simply rotate
/var/log/asterisk/*' . Is this the case?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
___
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Re: [asterisk-users] Asterisk on 64bit xeon
When you want to ask a new message, rather than replying to an existing one, please write a new message and don't reply to an existing one. (and don't even reply to an existing one and delete its contets. This is not the same). Posting the same question twice is also not a good habit. See reply inline, On Tue, Oct 10, 2006 at 09:05:01AM +, Akpome Akpoguma wrote: Hi All, Would asterisk and zaptel compile on 64bit dual xeon hardware?? It compiles even faster when use use -j3 (-j4?) with make ;-) Not only does it compiles, it also runs. In both 32 bit mode and 64 bit modes. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [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] Echo Cancel Cards
Joseph wrote: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. I've got an SPA-3000 at home that is constantly crashing, echoey and is almost unusable. (The CS4660-based ATA and PA1688-based handsets have otherwise been fine, as were the the Cisco 468 and Linysys PAP2 when they were in use). ___ --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] single conference, multiple numbers
Hi, Is it within the realms of possibility to have a single conference with multiple numbers? I'm thinking of getting PSTN numbers in a number of different countries so that people in those countries only pay for a local call. At this stage doing it with VoIP is out of the question. Thanks -- Mike Williams ___ --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] single conference, multiple numbers
Yes you can. If you are dealing striclty with non-voip you may have a bit of a challenge getting the calls to the same server but if you had a way of getting all the calls to the same box, or have boxes all over and have them all conntect to one main box then it would be possible. - Original Message - From: Mike Williams [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 12:37 PM Subject: [asterisk-users] single conference, multiple numbers Hi, Is it within the realms of possibility to have a single conference with multiple numbers? I'm thinking of getting PSTN numbers in a number of different countries so that people in those countries only pay for a local call. At this stage doing it with VoIP is out of the question. Thanks -- Mike Williams ___ --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] Echo Cancel Cards
I have never used T1 cards but as far as POTS line cards I would say that I like sangoma better. It is a little bit harder to set up but works wonders. - Original Message - From: Thomas Kenyon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 12:31 PM Subject: Re: [asterisk-users] Echo Cancel Cards Joseph wrote: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. I've got an SPA-3000 at home that is constantly crashing, echoey and is almost unusable. (The CS4660-based ATA and PA1688-based handsets have otherwise been fine, as were the the Cisco 468 and Linysys PAP2 when they were in use). ___ --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] help this....
Hello UsersHelp me ... the below error [codec_lpc10.so] = (LPC10 2.4kbps (signed linear) Voice Coder) == Parsing '/etc/asterisk/codecs.conf': Found -- Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded. -- codec_lpc10: using generic PLC == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 41 == Registered translator 'lintolpc10' from format slin to lpc10, cost 2 Bolded one... is occuring in difference configuring files...Help what this means... :P-- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.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] single conference, multiple numbers
On Tue, Oct 10, 2006 at 11:37:49AM +0100, Mike Williams wrote: Hi, Is it within the realms of possibility to have a single conference with multiple numbers? exten = 1234,1,Meetme() exten = 5676,1,Meetme() ? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [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
[asterisk-users] Tutorial: Simple queue and agent debug monitoring
Hi list, out of pure frustration I have prepared another tutorial (must be the season) about how to filter the various outputs of Asterisk in order to keep track of what is going on in realtime in a call-center, to avoid being swamped by too many logging and information on the * side. http://astrecipes.net/index.php?n=209 Any comments or corrections are welcome! l. -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --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] how to play pre-recorded file in meetme conference
I don't know if there is a better way to do this with meetme itself, but you could use the manager interface (or even the file method described in http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out). You can pass a Data argument with the filename, to an extension that simply plays a file into the conference. You may also be able to do something with the 'b' argument to MeetMe. --Brian On Mon, Oct 09, 2006 at 04:42:02PM -0400, Barry D. Hassler wrote: Hey folks, Is it possible to play a pre-recorded file in a meetme conference? That is, I'd like to get everyone into a conference, then somehow play a previously recorded file (in this case, a podcast). This isn't for individuals to call into to listen to the cast, but for it to be played simultaneously for all in the conference. This would be handy for me! ___ --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] CDR - mysql with asterisk 1.2.12 not working
I know that this is a silly suggestion but you should check to make sure that you actually have the cdr_mysql module, because at some point (I believe at the 1.2 release or shortly thereafter), it was moved into asterisk-addons. --Brian On Tue, Oct 10, 2006 at 08:31:43AM +0200, Garth van Sittert wrote: Joseph wrote: On Mon, 2006-10-09 at 01:45 -0300, Hermann Wecke wrote: On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote: I have bind-address = 127.0.0.1 in my.cnf the cdr was working find with asterisk 1.0.1 just after upgrade something is not connecting. I don't know if asterisk will use the localhost or the network IP to connect. Just try to comment your line and see what happens. This is really a guess... Make no difference if I use IP or localhost it is still not connecting; it could be something with the cdr_addon_mysql.so Anybody has any other ideas / suggestions? Have you tried turning on debug in logger.conf. You should be able to see what is wrong from there. Kind Regards Garth ___ --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] single conference, multiple numbers
Absolutely, the MeetMe command just takes a conference number. You could have as many extensions invoke it as you would like. --Brian On Tue, Oct 10, 2006 at 11:37:49AM +0100, Mike Williams wrote: Hi, Is it within the realms of possibility to have a single conference with multiple numbers? I'm thinking of getting PSTN numbers in a number of different countries so that people in those countries only pay for a local call. At this stage doing it with VoIP is out of the question. Thanks -- Mike Williams ___ --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] alive check for HA constellation
Hi, I have setup two asterisks with ucarp, to build a HA cluster. Everything works fine, if one of the machines is going to die completely. But if the asterisk software is running, but behaving not correctly, this cannot be detected by the ucarp software. I think I need a script that periodically checks the master, and if the answer is not the expected one, the slave shall try to take over the master. I can imagine this will work when I try to check whether I can successfully authenticate via SIP to the asterisk. Just pipe it through netcat, and wait for the answer. but I have the feeling that I am not the first one with that problem, so I want to ask for more easily/robust tests to make sure the master is running or not. any suggestions are appreciated. kind regards Sebastian ___ --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] alive check for HA constellation
Hi Sebastian, This url http://underlinux.com.br/content/view/6330/70/ have some thinks that you need.Leonardo Silva 2006/10/10, Sebastian Reitenbach [EMAIL PROTECTED]: Hi,I have setup two asterisks with ucarp, to build a HA cluster. Everything worksfine, if one of the machines is going to die completely. But if the asterisksoftware is running, but behaving not correctly, this cannot be detected by the ucarp software.I think I need a script that periodically checks the master, and if the answeris not the expected one, the slave shall try to take over the master.I can imagine this will work when I try to check whether I can successfully authenticate via SIP to the asterisk. Just pipe it through netcat, and waitfor the answer. but I have the feeling that I am not the first one with thatproblem, so I want to ask for more easily/robust tests to make sure the master is running or not.any suggestions are appreciated.kind regardsSebastian___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Leonardo Silvafone: 16 8143-1146 ___ --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] single conference, multiple numbers
On Tuesday 10 October 2006 12:24, Brian Rogan wrote: Absolutely, the MeetMe command just takes a conference number. You could have as many extensions invoke it as you would like. Thanks all. -- Mike Williams ___ --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] whisper paging
Does anyone have a quick howto and a sample to get whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2 Thanks for your help! ___ --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] T1 Passthrough
Assuming the outgoing T1 to the Norstar is a standard T1 that accepts ANI and
DNIS all have to do is
exten = _XXX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
This will redial the caller id (ANI) and the 3 digits Dialed (DNIS) to the
Norstar T1 in the formst *ANI*DNIS*
I did the same thing for a while to convert and PRI to a T1 into a Mitel system
that could no do PRI without a
very expensive upgrade.
--
From: Forrest Beck[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, October 09, 2006 10:18 PM
To: Asterisk Users List
Subject:[asterisk-users] T1 Passthrough
I want to setup a asterisk server with two T1 spans (two TE110P
cards). The server will have one card connected to the PRI and the
other will connect to our Norstar Meridian ICS system. I want to have
a very simple dial plan for the context that the PRI card will be
assigned to something like this. Note that our telecom provider sends
final three digits of the phone number:
SPAN 1
Channels 1-23
g1
context: pri_incoming
SPAN 2
Channels 25-48
g2
context: norstar_ics
[pri_incoming]
exten = _XXX,1,Dial,ZAP/g2/${EXTEN}
My questions are:
Will I need to set the callerid before routing to the next span, or
will the three digits remain intact.?
and
Has anyone tried this? and if so do you forsee any problems i will run into?
This is all theroey in my head right now, since I am awaiting the
second cards arrival.
Thanks.
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Re: [asterisk-users] OT: Hand free solution recommandation
Ideal would be a headset audio+microphone with RJ11 4p female that we could plug into the handset cable of any IP phone, or a converter 2xjack2,5mm female RJ11 4p female -which seems not to exist-. What are you recommanding/using/installing in such case? I don't know if it would work on any phone, but it works on Cisco 7940/7960 : http://www.mml.uni-hannover.de/einhorn/headset/index_e.html hth ___ --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] Re: Range Operator
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG How can I check a number is within a specified range in the
DG dialplan? What's the greater than operator? How would I use a
DG combination of greater than and less than in conjection with
DG GotoIf()? The following seems to break the dialplan. I need to
DG check callerid is _5XXX.
DG _X./_5XXX,1,Set(CALLERID(number)=5551212)
DG _X./_5XXX,n,NoOp(Dialplan dies before here)
DG Presumably it's because we just changed the callerid number and
DG the dialplan now has nowhere to go.
How about simply:
_X./_5XXX,1,Goto(handle5xxx,${EXTEN},1)
[handle5xxx]
_X.,1,Set(CALLERID(number)=5551212)
_X.,n,NoOp(foo)
If you want the other one, you can:
_X./_5XXX,1,Set(CALLERID(number)=5551212)
_X./_5551212,2,NoOp(foo)
(n probably works here too, but since n always increments, I'm wary of
using n with exgirlfriend logic)
/Benny
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Re: [asterisk-users] Echo Cancel Cards
The Sangoma single port T1 card also works well, and , along with the 5 year warranty, works with just about any MB that it will fit, makes it a no brainer choice over the Digium products. Sangoma just doesn't say try another motherboard. If their product doesn't work, they find out why and fix it! John Novack Dovid B wrote: I have never used T1 cards but as far as POTS line cards I would say that I like sangoma better. It is a little bit harder to set up but works wonders. - Original Message - From: Thomas Kenyon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 12:31 PM Subject: Re: [asterisk-users] Echo Cancel Cards Joseph wrote: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. I've got an SPA-3000 at home that is constantly crashing, echoey and is almost unusable. (The CS4660-based ATA and PA1688-based handsets have otherwise been fine, as were the the Cisco 468 and Linysys PAP2 when they were in use). ___ --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 ___ --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] alive check for HA constellation
Hi Leonardo, I had the problem, asterisk was running, the port was open, but I misconfigured Asterisk that way, that it was impossible, to register on the asterisk. As it seems to me, hapm can only check whether a port is closed or open. unfortunately I do not understand that brazilian portugese. Sebastian Leonardo Silva [EMAIL PROTECTED] wrote: Hi Sebastian, This url http://underlinux.com.br/content/view/6330/70/ have some thinks that you need. Leonardo Silva 2006/10/10, Sebastian Reitenbach [EMAIL PROTECTED]: Hi, I have setup two asterisks with ucarp, to build a HA cluster. Everything works fine, if one of the machines is going to die completely. But if the asterisk software is running, but behaving not correctly, this cannot be detected by the ucarp software. I think I need a script that periodically checks the master, and if the answer is not the expected one, the slave shall try to take over the master. I can imagine this will work when I try to check whether I can successfully authenticate via SIP to the asterisk. Just pipe it through netcat, and wait for the answer. but I have the feeling that I am not the first one with that problem, so I want to ask for more easily/robust tests to make sure the master is running or not. any suggestions are appreciated. kind regards Sebastian ___ --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] Echo Cancel Cards
I´m about to acquire an E1 interface. I was reading about TE110P and hardware incompatibilities issues with some boards, servers and chipsets. I also read a lot of compliments about Sangoma Hardware (specially for E1/T1 interfaces) and I was wondering if A101 from Sagoma is a better choice (technically speaking) than Digium TE110P. I read now, on this post, an opinion about Sangoma interfaces and echo cancellation issues.. I have a PC with Asus P5LD2 board (Intel 945P chipset). I asked Digium support for how compatible is the TE110P with my box.. and they said that no incompatibility issues had been reported with the chipset I use.. BUT, they had no test TE110P with this chipset... I´m not a Sangoma or Digium fan... I´m just a newbie who don´t want to get the wrong piece of hardware. I really appreciate any advice from people with a lot of experience and skills on this topic. Thanks in advance R.R. Libera Dovid B escribió: I have never used T1 cards but as far as POTS line cards I would say that I like sangoma better. It is a little bit harder to set up but works wonders. - Original Message - From: Thomas Kenyon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 12:31 PM Subject: Re: [asterisk-users] Echo Cancel Cards Joseph wrote: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. I've got an SPA-3000 at home that is constantly crashing, echoey and is almost unusable. (The CS4660-based ATA and PA1688-based handsets have otherwise been fine, as were the the Cisco 468 and Linysys PAP2 when they were in use). ___ --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 ___ --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] Error loading Unicall
Try using testcall tool included with Unicall to debug, as shown in this document I wrote a couple of months ago. It also shows how to use zttool to detect problems in the E1 layer. http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf The verbosity level in testcall.c must be at highest level to be able to see the problem clearly. If you have more that 1 port in your PCI cards, try using a loop, like explained in the document, to discard problems on your side. Regards On 10/9/06, Carlos Chavez [EMAIL PROTECTED] wrote: On Mon, 2006-10-09 at 19:39 -0500, Moises Silva wrote: Same problem as your other post. dtmf_put is no longer available in newer spandsp versions, the solutions is the same as with libmfcr2, downgrade spandsp, or upgrade chan_unicall (not always a matching Ok, I downgraded all the programs and now everything compiles. Now I cannot make or receive any calls. When I dial a Unicall channel I can hear a crack on the phone and after a few seconds it gives me a busy tone. I have never had this much trouble installing mfcr2 but I usually use CentOS instead of FC5. I had to use FC5 because the drivers for Xorcom do not compile in CentOS. -- Executing Set(SIP/139-091cd588, TIMEOUT(absolute)=900) in new stack -- Channel will hangup at 2006-10-10 01:22:54 UTC. -- Executing Dial(SIP/139-091cd588, Unicall/g1/0445529613670) in new stack Oct 9 20:07:54 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(1) Oct 9 20:07:54 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Make call Oct 9 20:07:54 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32770 Oct 9 20:07:54 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0001 - [1/ 1/Idle /Idle ] -- Called g1/0445529613670 Oct 9 20:07:54 WARNING[19096]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Dialing Oct 9 20:07:54 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Oct 9 20:07:54 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 4 on [2/ 40/Group I /DNIS ] Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [2/ 40/Group I /DNIS ] Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 4 off [2/ 40/Group I /DNIS ] Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Far end disconnected(cause=Switching equipment congestion [42]) - state 0x40 Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Far end disconnected Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:2930 handle_uc_event: CRN 32770 - far disconnected cause=Switching equipment congestion [42] -- Channel 0 got hangup -- UniCall/1-1 is circuit-busy Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel gains Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel switching Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(6) Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Clearing fwd Oct 9 20:08:12 WARNING[19096]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [2/ 800/Clear fwd B /DNIS ] -- Hungup 'UniCall/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion(SIP/139-091cd588, ) in new stack == Spawn extension (oficina-todo, 90445529613670, 3) exited non-zero on 'SIP/139-091cd588' Oct 9 20:08:12 WARNING[19062]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 800/Clear fwd D /Idle ] Oct 9 20:08:12 WARNING[19062]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call disconnected(cause=Switching equipment congestion [42]) - state 0x800 Oct 9 20:08:12 WARNING[19062]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Drop call Oct 9 20:08:12 WARNING[19062]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(7) Oct 9 20:08:12 WARNING[19062]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release call Oct 9 20:08:12 WARNING[19062]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Destroying call with CRN 32770 Oct 9 20:08:12 WARNING[19062]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Release call -- Unicall/1 released Oct 9 20:08:12 WARNING[19062]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel -- Carlos Chavez
Re: [asterisk-users] Echo Cancel Cards
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I'm quite happy with sangoma cards, no issues so far, plus their installation/setup software makes it a breeze to get everything working. R.R. Libera wrote: I´m about to acquire an E1 interface. I was reading about TE110P and hardware incompatibilities issues with some boards, servers and chipsets. I also read a lot of compliments about Sangoma Hardware (specially for E1/T1 interfaces) and I was wondering if A101 from Sagoma is a better choice (technically speaking) than Digium TE110P. I read now, on this post, an opinion about Sangoma interfaces and echo cancellation issues.. I have a PC with Asus P5LD2 board (Intel 945P chipset). I asked Digium support for how compatible is the TE110P with my box.. and they said that no incompatibility issues had been reported with the chipset I use.. BUT, they had no test TE110P with this chipset... I´m not a Sangoma or Digium fan... I´m just a newbie who don´t want to get the wrong piece of hardware. I really appreciate any advice from people with a lot of experience and skills on this topic. Thanks in advance R.R. Libera -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFK6YM2QVs8jsa1mQRAklcAJ9sEpXIumEug6zulq+v6q72pgee9ACfU7t/ utDsPjAfJXNWCW07Q0Q8q5U= =Rmc0 -END PGP SIGNATURE- ___ --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] Re: PRI issues
On Mon, Oct 09, 2006 at 06:42:06PM -0400, Doug Lytle wrote: Doug Lytle wrote: Jay R. Ashworth wrote: From the A102 spec sheet: * DSU/CSU set up entirely in software. I guess I need to learn to read a little more carefully. Looks like it's 'set up' in software. Well, I was working on my own snarky reply, when I discovered it's insanely difficult to find an online reference that describes in proper technical detail what a CSU actually *does* -- hell, some of the writeups can't even expand the initialism correctly. :-) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] External Custom Extension Timeout
[EMAIL PROTECTED] wrote: Hello, I'm having trouble getting this to work: I have a ring group that dials an extension and if no answer dials a cell phone. If the cell phone doesn't answer I want to go to voicemail or another extension. I have set the timeout to 15 seconds but it never actually works, it will just ring until the cell voice mail picks up. I'm using [EMAIL PROTECTED] 2.8 and a TDM400P card. Please, any help is greatly appreciated! Craig I'm running Asterisk 1.2.12.1 and Freepbx 2.1.3 and have this problem also. Also on a TDM400P card. I've tried setting up a queue, ring group, followme, none of the timeouts are obeyed. JD ___ --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] whisper paging
AFAIK it's not possible. On 10/10/06, Hall, Eric M. [EMAIL PROTECTED] wrote: Does anyone have a quick howto and a sample to get whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2 Thanks for your help! ___ --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] Echo Cancel Cards
I am using the TE110P with the Intel 945P chipset, and I don't have any issues with compatibility. The 945P chipset is a very common chipset for the D and 4 processor. Works quite well. On 10/10/06, R.R. Libera [EMAIL PROTECTED] wrote: I´m about to acquire an E1 interface. I was reading about TE110P and hardware incompatibilities issues with some boards, servers and chipsets. I also read a lot of compliments about Sangoma Hardware (specially for E1/T1 interfaces) and I was wondering if A101 from Sagoma is a better choice (technically speaking) than Digium TE110P. I read now, on this post, an opinion about Sangoma interfaces and echo cancellation issues.. I have a PC with Asus P5LD2 board (Intel 945P chipset). I asked Digium support for how compatible is the TE110P with my box.. and they said that no incompatibility issues had been reported with the chipset I use.. BUT, they had no test TE110P with this chipset... I´m not a Sangoma or Digium fan... I´m just a newbie who don´t want to get the wrong piece of hardware. I really appreciate any advice from people with a lot of experience and skills on this topic. Thanks in advance R.R. Libera Dovid B escribió: I have never used T1 cards but as far as POTS line cards I would say that I like sangoma better. It is a little bit harder to set up but works wonders. - Original Message - From: Thomas Kenyon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 12:31 PM Subject: Re: [asterisk-users] Echo Cancel Cards Joseph wrote: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. I've got an SPA-3000 at home that is constantly crashing, echoey and is almost unusable. (The CS4660-based ATA and PA1688-based handsets have otherwise been fine, as were the the Cisco 468 and Linysys PAP2 when they were in use). ___ --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 ___ --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] FYI - Polycom SoundPoint IP 301 Denial of Service]
FYI. TITLE: Polycom SoundPoint IP 301 Denial of Service SECUNIA ADVISORY ID: SA22266 VERIFY ADVISORY: http://secunia.com/advisories/22266/ CRITICAL: Less critical IMPACT: DoS WHERE: From local network OPERATING SYSTEM: Polycom SoundPoint IP 301 http://secunia.com/product/12229/ DESCRIPTION: A vulnerability has been reported in the Polycom SoundPoint IP 301 VoIP Desktop Phone, which can be exploited by malicious people to cause a DoS (Denial of Service). Sending a long URL to the embedded HTTP server or using the Nessus http_fingerprinting_hmap.nasl script can cause the phone to reboot. Additional, it has been reported that the TCP port 42 is open and accepting connections. The vulnerabilities have been reported in firmware version 1.4.1.0040. Other versions may also be affected. SOLUTION: Reportedly, this does not affect the firmware version 2.0.1. PROVIDED AND/OR DISCOVERED BY: Shawn Merdinger -- About: This Advisory was delivered by Secunia as a free service to help everybody keeping their systems up to date against the latest vulnerabilities. Subscribe: http://secunia.com/secunia_security_advisories/ ___ --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] OT: Hand free solution recommandation
Plantronics makes something like this...designed to go inline with handset cable, with 2 2.5mm audio connectors for connection to PC. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Tuesday, October 10, 2006 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Hand free solution recommandation Ideal would be a headset audio+microphone with RJ11 4p female that we could plug into the handset cable of any IP phone, or a converter 2xjack2,5mm female RJ11 4p female -which seems not to exist-. What are you recommanding/using/installing in such case? I don't know if it would work on any phone, but it works on Cisco 7940/7960 : http://www.mml.uni-hannover.de/einhorn/headset/index_e.html hth ___ --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] single conference, multiple numbers
Mike Williams wrote: Hi, Is it within the realms of possibility to have a single conference with multiple numbers? I'm thinking of getting PSTN numbers in a number of different countries so that people in those countries only pay for a local call. At this stage doing it with VoIP is out of the question. This is very simple to do with voip, It is worth noting that with some countries you cannot yet buy a number (Dominican republic and Isle of Man spring to mind). ___ --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] Echo Cancel Cards
R.R. Libera wrote: I´m about to acquire an E1 interface. I was reading about TE110P and hardware incompatibilities issues with some boards, servers and chipsets. I also read a lot of compliments about Sangoma Hardware (specially for E1/T1 interfaces) and I was wondering if A101 from Sagoma is a better choice (technically speaking) than Digium TE110P. I read now, on this post, an opinion about Sangoma interfaces and echo cancellation issues.. I've just put the A102 from Sangoma into a system here - their install routine did almost everything for me. Just some minor Asterisk configuration needed before I could plug it into our legacy PBX and have the two talking like friends! [EMAIL PROTECTED] :o) ___ --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] Inbound Callcenter with multiple DIDs
I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like accounts 1 - 100 are in group 1, 101 - 200 are in group two. Some agents can get calls from group 1, some from group 2 and some from both groups. Most solutions I have found are meant for inbound call centers that handle only a few types of calls and have little need to make large distinctions between different DIDs. I have played around with QueueMetrics and it is a good piece of software, but does not handle the DIDs the way I need. Really any recommendations for software to go with asterisk that inbound call centers are using and find useful would be great. -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --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] help this....
Hello. It would appear that the voicemail module is not loaded. If this is a new install, did you install the sample config files? Specifically voicemail.conf. -Ejay From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of raviprakash sunkaraSent: Tuesday, October 10, 2006 6:07 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] help this Hello UsersHelp me ... the below error [codec_lpc10.so] = (LPC10 2.4kbps (signed linear) Voice Coder) == Parsing '/etc/asterisk/codecs.conf': Found -- Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded. -- codec_lpc10: using generic PLC == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 41 == Registered translator 'lintolpc10' from format slin to lpc10, cost 2 Bolded one... is occuring in difference configuring files...Help what this means... :P-- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED]www.hyperion-tech.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] whisper paging
I thought whisper paging was implemented in 1.4? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] whisper paging AFAIK it's not possible. On 10/10/06, Hall, Eric M. [EMAIL PROTECTED] wrote: Does anyone have a quick howto and a sample to get whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2 Thanks for your help! ___ --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 ___ --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] Xorcom TS-1 and Digium TE110P or TE210P
Hi! Does anyone have some experience with a Xorcom TS-1 and a 1 or 2 port Digium PRI card? I am looking for a SIP/IAX to ISDN gateway and this combination could by interesting. But Xorcom writes that the TS-1 is compatible with Digium PRI cards but that it has not been tested much.-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --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] whisper paging
Then correct me. On 10/10/06, Douglas Garstang [EMAIL PROTECTED] wrote: I thought whisper paging was implemented in 1.4? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] whisper paging AFAIK it's not possible. On 10/10/06, Hall, Eric M. [EMAIL PROTECTED] wrote: Does anyone have a quick howto and a sample to get whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2 Thanks for your help! ___ --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 ___ --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] Asterisk 1.2.12.1 and snom 360 6.2.3 no audio
Hello, when trying to use a snom 360 (Firmware 6.2.3) with Asterisk 1.2.12, I receive no audio. Asterisk 1.4.0 b2 works fine though. I´d upgrade to 1.4 if I hadn´t just bought a Junghanns OctoBRI that apparently only works with bristuff, which is stuck at the 1.2 series. Is this a known problem? Can it be fixed by downgrading the phones software? Thank you, Holger von Ameln -- Holger von Ameln Aseko GmbH Co. KG Prinzenstr. 10a 30159 Hannover Tel: +49 511 220626 22 Fax: +49 511 220626 66 ___ --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] whisper paging
Seems that you guys are right, sorry. http://www.digium.com/en/mediacenter/news/viewpress.php?id=Asterisk1.4 On 10/10/06, C F [EMAIL PROTECTED] wrote: Then correct me. On 10/10/06, Douglas Garstang [EMAIL PROTECTED] wrote: I thought whisper paging was implemented in 1.4? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] whisper paging AFAIK it's not possible. On 10/10/06, Hall, Eric M. [EMAIL PROTECTED] wrote: Does anyone have a quick howto and a sample to get whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2 Thanks for your help! ___ --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 ___ --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] Voicemail Press '0'
Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention Whatever that means... My dialplan has: [foo] exten = 556,1,Answer exten = 556,n,Voicemail([EMAIL PROTECTED]) [default] exten = o,n,Playback(tt-monkeys) My voicemail.conf has: [general] operator=yes [default] 3254101 = 1234,Foo When I press '0' nothing happens. Nothing is displayed on the console to indicate any attempt to dial 'o'. Doug. ___ --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] whisper paging
According to this: http://bugs.digium.com/view.php?id=8019 it seems that it's part of Chan_spy, what does show application chanspy on the cli give you? On 10/10/06, C F [EMAIL PROTECTED] wrote: Seems that you guys are right, sorry. http://www.digium.com/en/mediacenter/news/viewpress.php?id=Asterisk1.4 On 10/10/06, C F [EMAIL PROTECTED] wrote: Then correct me. On 10/10/06, Douglas Garstang [EMAIL PROTECTED] wrote: I thought whisper paging was implemented in 1.4? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] whisper paging AFAIK it's not possible. On 10/10/06, Hall, Eric M. [EMAIL PROTECTED] wrote: Does anyone have a quick howto and a sample to get whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2 Thanks for your help! ___ --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 ___ --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] whisper paging
That's what we got told at the Asterisk bootcamp training in Kansas City a few weeks ago... :) -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] whisper paging Then correct me. On 10/10/06, Douglas Garstang [EMAIL PROTECTED] wrote: I thought whisper paging was implemented in 1.4? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] whisper paging AFAIK it's not possible. On 10/10/06, Hall, Eric M. [EMAIL PROTECTED] wrote: Does anyone have a quick howto and a sample to get whisper paging to work? I'm running sterisk Asterisk 1.4.0-beta2 Thanks for your help! ___ --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 ___ --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 ___ --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] whisper paging
C F wrote: According to this: http://bugs.digium.com/view.php?id=8019 it seems that it's part of Chan_spy, what does show application chanspy on the cli give you? I would wait until the next beta before giving whispering a try, it underwent some major changes. Alternatively you can grab the 1.4 branch instead and try it if you are interested. (Any of you). -- Joshua Colp Software Developer Digium, Inc. ___ --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] connecting multiple servers with iax - authentication fails
Tim Panton wrote: On 9 Oct 2006, at 17:36, Benko wrote: Hello! I'm having a problem which actually looks banal. I'm trying to connect 3 servers via iax with each other. However, i've not been successfull so far. Asterisk always tries to authenticate the calling user with the credentials of the last entry in iax.conf, not the ones that would actually belong to the calling user. e.g. Server1 has peer/user entries for Server2 and Server3(in this order), Server2 now tries to call Server1, but is asked for the credentials of Server3(Because Server3 is the last entry in iax.conf), which doesn't work of course. The IAX debug for this example is attached(iax_server2.txt). Please also take a look at the attached iax.conf-files for each server, maybe i've missed some setting... Currently i workaround this issue by using the same secret for all servers, this is not very practicable however... The asterisk versions in use are 1.2.9.1 on server3 and server2 and 1.4.0-beta2 on server1. This guy seems to have had the same problem, unfortunately he received no answer: http://lists.digium.com/pipermail/asterisk-users/2003-August/011960.html thx christian iax.conf.server1.txt iax.conf.server2.txt iax.conf.server3.txt iax_debug_server2.txt It is a bit hard to tell what is going on because you have blanked the IP addresses in the config files to all the same value. If you specify an IP address in the host= line , asterisk will use the from IP address of a 'new' to try and find a matching entry, and ignore the username sent in the message. At a guess you have the IP addresses of servers 1 or 3 wrong in iax.conf on server2 Tim. Tim Panton www.mexuar.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 I would also like to point out that it is good practice to specify the username you want to authenticate as. If it is not given (ie: not given by the username option in peer, or on the Dial line) then the remote Asterisk box will guess who you want to authenticate as which may be incorrect. This will cause an authentication failure. -- Joshua Colp Software Developer Digium, Inc. ___ --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] Voicemail Press '0'
Douglas Garstang wrote:
Crikey. I can't get this to work!
[foo]
exten = 556,1,Answer
exten = 556,n,Voicemail([EMAIL PROTECTED])
I believe it needs to be in the same context as your voicemail. Mine is:
[voice-mail]
exten = s,1,Set(CALLBACK=${DB(vmcallback/${CALLERIDNUM})})
exten = s,2,GotoIf($[${CALLBACK} = YES]?3:4)
exten = s,3,System(/usr/local/bin/vm-callout-delete.sh ${CALLERIDNUM})
exten = s,4,Set(TIMEOUT(response)=15)
exten = s,5,Set(TIMEOUT(digit)=4)
exten = s,6,VoicemailMain(@sip)
exten = s,7,Hangup()
exten = a,1,Goto(incoming,s,2) ; (* pressed to break out of directory,
goto incoming context)
exten = o,1,Goto(incoming,s,2) ; (Zero pressed for operator, goto
incoming context)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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RE: [asterisk-users] Voicemail Press '0'
If you get an answer for this please post it here on forum as I and at least one other I've talked to have this same problem. I found it was only a problem from external calls though not internally. Same for you? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, October 10, 2006 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail Press '0' Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention Whatever that means... My dialplan has: [foo] exten = 556,1,Answer exten = 556,n,Voicemail([EMAIL PROTECTED]) [default] exten = o,n,Playback(tt-monkeys) My voicemail.conf has: [general] operator=yes [default] 3254101 = 1234,Foo When I press '0' nothing happens. Nothing is displayed on the console to indicate any attempt to dial 'o'. Doug. ___ --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] Voicemail Press '0'
Douglas Garstang wrote: Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention Whatever that means... My dialplan has: [foo] exten = 556,1,Answer exten = 556,n,Voicemail([EMAIL PROTECTED]) [default] exten = o,n,Playback(tt-monkeys) My voicemail.conf has: [general] operator=yes [default] 3254101 = 1234,Foo When I press '0' nothing happens. Nothing is displayed on the console to indicate any attempt to dial 'o'. Doug. You'll want the 'o' extension to be in the same context where voicemail is called from. Try that and see if it works. -- Joshua Colp Software Developer Digium, Inc. ___ --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] Voicemail Press '0'
Dang it. Thanks. Blindly trusting the voip-wiki is bad
When using the zero '0' and star '*' it's important to note that the context
you placed the application voicemail in is irrelvant...
Doug.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Press '0'
Douglas Garstang wrote:
Crikey. I can't get this to work!
[foo]
exten = 556,1,Answer
exten = 556,n,Voicemail([EMAIL PROTECTED])
I believe it needs to be in the same context as your
voicemail. Mine is:
[voice-mail]
exten = s,1,Set(CALLBACK=${DB(vmcallback/${CALLERIDNUM})})
exten = s,2,GotoIf($[${CALLBACK} = YES]?3:4)
exten = s,3,System(/usr/local/bin/vm-callout-delete.sh
${CALLERIDNUM})
exten = s,4,Set(TIMEOUT(response)=15)
exten = s,5,Set(TIMEOUT(digit)=4)
exten = s,6,VoicemailMain(@sip)
exten = s,7,Hangup()
exten = a,1,Goto(incoming,s,2) ; (* pressed to break out of
directory,
goto incoming context)
exten = o,1,Goto(incoming,s,2) ; (Zero pressed for operator, goto
incoming context)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
little Temporary Safety, deserve neither Liberty nor Safety.
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[asterisk-users] bristuff problem?
Hi Kape, With latest asterisk 1.2.12.1, zaptel 1.2.9.1 and bristuff 0.3.1s after a while calls become stuck: either the caller or callee can't hear the other party, or heavy static is heard. An asterisk restart fixes it for a short while only. This doesn't happen with our older installs (asterisk 1.2.9, zaptel 1.2.7, bristuff 0.3.1q). Are you aware of that problem? 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] Re: Where is the PlayDTMF command?
Jan, im sorry to get back to you so late, ive been busy. It seems i sent you an incorrect patch I was testing, but I have found the correct patch in mantis: http://bugs.digium.com/view.php?id=6682 Please be aware that the patch I sent you initially used a funciton that received 1 or more DTMF digits, and thats why it fails, because the operation need to be fast enough to not lock the channel more time than allowed, so the patch you can find now in mantis, use a function that only accepts 1 DTMF digit at time, so PlayDTMF only accepts 1 digit to, you need to call it several times to send a DTMF stream. Regards On 10/9/06, Jan du Toit [EMAIL PROTECTED] wrote: So I patch my asterisk (version 1.2.12.1) with the patch given by Moises. http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch Thanks Moises. When I type in show manager command PlayDTMF it is their. With the show manager commands it is not within the list containing all the commands. When I execute the manager PlayDTMF action, the manager response says DTMF successfully queued. I don't hear anything on the phone, when I look at the CLI I see the following warning message. Its produced everytime I execute the PlayDTMF action. Oct 6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 360468 in procedure ast_waitfor_nandfds Am I doing something wrong? Is this a bug? Please help, I need this to work as soon as possible... Thanks for all the help. ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Voicemail Press '0'
That was it for me as well. Couldn't get that answer the last time I
asked. Thanks guys.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, October 10, 2006 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Voicemail Press '0'
Dang it. Thanks. Blindly trusting the voip-wiki is bad
When using the zero '0' and star '*' it's important to note that the
context you placed the application voicemail in is irrelvant...
Doug.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Press '0'
Douglas Garstang wrote:
Crikey. I can't get this to work!
[foo]
exten = 556,1,Answer
exten = 556,n,Voicemail([EMAIL PROTECTED])
I believe it needs to be in the same context as your
voicemail. Mine is:
[voice-mail]
exten = s,1,Set(CALLBACK=${DB(vmcallback/${CALLERIDNUM})})
exten = s,2,GotoIf($[${CALLBACK} = YES]?3:4)
exten = s,3,System(/usr/local/bin/vm-callout-delete.sh
${CALLERIDNUM})
exten = s,4,Set(TIMEOUT(response)=15)
exten = s,5,Set(TIMEOUT(digit)=4)
exten = s,6,VoicemailMain(@sip)
exten = s,7,Hangup()
exten = a,1,Goto(incoming,s,2) ; (* pressed to break out of
directory,
goto incoming context)
exten = o,1,Goto(incoming,s,2) ; (Zero pressed for operator, goto
incoming context)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
little Temporary Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] VOIP with PSTN backup
Brian Candler wrote: I'm looking for a way to set up a VOIP network in branch offices where one or more phones have lifeline capability, i.e. can place calls if the IP network or VOIP service dies, or even if power goes down. (I'm thinking of business continuity here, not just emergency services) This seems to limit my choice of products somewhat, and I was wondering if anyone had recommendations for use in this scenario. The approaches I'm thinking of are: (1) Use an ATA with PSTN passthrough or FXO port, and connect an old analogue telephone to the FXS port. In this case, the analogue phone has lifeline. If there's a true FXO port then PSTN calls can in principle be routed to/from other VOIP phones in the office (but see below) There seem to be a few to choose from, although far fewer with a true FXO port. (2) Find a VOIP phone with integrated PSTN or FXO port In this case, the only one I have found so far by searching the web is Clipcomm CP101. I have also read that many FXO devices tend to be badly implemented; in particular, on seeing ringing voltage, they actually pick up and answer the call, instead of sending off a SIP INVITE and waiting for the OK before connecting. I'd certainly like the device to behave properly in this regard. As a second part of this question, it would be extremely desirable if the backup PSTN service were available to all the phones in the office. That means: (a) incoming PSTN calls could ring *all* the VOIP phones in the office, not just the one phone or ATA connected to the PSTN line; and (b) any VOIP phone could route a call out over the LAN to the local FXO PSTN port, e.g. by dialling a prefix to access it. This isn't so essential but it's definitely desirable. Any recommendations for how to do this too? A large number of offices is going to be involved, and I want to keep as much switching intelligence centralised as possible, both for ease of management and to keep the cost down. That is, I don't want to install a PC + TMD400P + Asterisk in each location, but just a small media gateway or VOIP phone. However I can see that the incoming ringing issue will require call forking, so I am happy to install an OpenWrt box running Asterisk or siproxd or whatever in each site. Being diskless and low power should mean little maintenance is required. But such a box isn't going to be able to take an FXS/FXO card, so I'll still rely on an ATA or VOIP phone to present a PSTN interface. So that's the key part I'm looking for. Finally, the devices must be robust (i.e. not need power cycling every 24 hours) and centrally manageable. I think that's about it - many thanks for your ideas and experience! If you get real serious about this, then do a risk assessment for each component involved in the end-to-end communications system. The risk assessment should include an analysis of each component answering questions like: 1. What's a reasonable business down time for the communications system (and that answer is not zero) 2. How important is the component (high, medium, low) 3. What's the likely restoration time for the component 4. What are some of the potential causes for a component failure etc, etc. Once that is done, I think you'll find that you can prioritize which assets need to be addressed in what order. For example, a fiber seeking backhoe will likely disable all forms of communications (eg, analog and digital). Therefore, trying to locate a phone (or ATA) with an analog fxo port is of no value. Finding an alternative carrier maybe based on some form of wireless service, cable broadband, etc, might be a reasonable approach. Some companies will actually bury telecomm communications facilities into a building, arriving from two distinct locations, thus reducing the exposure to the fiber seeking backhoe. ___ --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] Inbound Callcenter with multiple DIDs
Hi Michael, If you want something very basic: http://www.micpc.com/eventmonitor will pop up a menu for an incoming call to an agent. It is a very basic system but i wrote it as such to be both functional and a framework to build from. You would need to enhance it (for your specific needs), however, since it has all of the asterisk events in a MySQL database, that should not be a problem. earl On Tuesday 10 October 2006 10:44, Michael Sampson wrote: I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like accounts 1 - 100 are in group 1, 101 - 200 are in group two. Some agents can get calls from group 1, some from group 2 and some from both groups. Most solutions I have found are meant for inbound call centers that handle only a few types of calls and have little need to make large distinctions between different DIDs. I have played around with QueueMetrics and it is a good piece of software, but does not handle the DIDs the way I need. Really any recommendations for software to go with asterisk that inbound call centers are using and find useful would be great. ___ --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] Voicemail Press '0'
Douglas Garstang wrote: Dang it. Thanks. Blindly trusting the voip-wiki is bad When using the zero '0' and star '*' it's important to note that the context you placed the application voicemail in is irrelvant... I've made note on the wiki. He did his testing via a macro under 1.07 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] T1 Passthrough
We have a solution like this working just fine for almost a year. We are
using qurad card for that. It is a good idea to have both PRI on one card.
CLID shouldl remain the same.
Vlad
- Original Message -
From: Forrest Beck [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Monday, October 09, 2006 8:18 PM
Subject: [asterisk-users] T1 Passthrough
I want to setup a asterisk server with two T1 spans (two TE110P
cards). The server will have one card connected to the PRI and the
other will connect to our Norstar Meridian ICS system. I want to have
a very simple dial plan for the context that the PRI card will be
assigned to something like this. Note that our telecom provider sends
final three digits of the phone number:
SPAN 1
Channels 1-23
g1
context: pri_incoming
SPAN 2
Channels 25-48
g2
context: norstar_ics
[pri_incoming]
exten = _XXX,1,Dial,ZAP/g2/${EXTEN}
My questions are:
Will I need to set the callerid before routing to the next span, or
will the three digits remain intact.?
and
Has anyone tried this? and if so do you forsee any problems i will run
into?
This is all theroey in my head right now, since I am awaiting the
second cards arrival.
Thanks.
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Re: [asterisk-users] Voicemail Press '0'
Douglas Garstang wrote: Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention Whatever that means... My dialplan has: [foo] exten = 556,1,Answer exten = 556,n,Voicemail([EMAIL PROTECTED]) [default] exten = o,n,Playback(tt-monkeys) My voicemail.conf has: [general] operator=yes [default] 3254101 = 1234,Foo When I press '0' nothing happens. Nothing is displayed on the console to indicate any attempt to dial 'o'. Put exten = o in the same context as Voicemail. I don't know if you can include = the context it is in or not. ___ --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] Connection question...
I want to try something with my asterisk but I have something that I need to know. The thing is that I am behind a NAT (I have to phones in a lan connected to the internet with a router), my server is directly conected to the internet on a different connection (in another place). I make a call from one phone to the other, but will they connect directly inside my lan? will I need an important Internet connection (I mean fast)? what info will be transfered from the server to the phones and from the phones to the server? If someone know something about this, I will appreciate any info, thanks to all, Danko ___ --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] Mitel 5224/SIP no MWI
Does anybody know if this is supposed to work and if so, what, if any, workaround is needed? I have other phones (Snom, Polycom) MWI working with this system fine. 6.0.0.19 (latest) Mitel SIP firmware is loaded. Thanks for your time, - Jesse -- Jesse Peterson [EMAIL PROTECTED] ___ --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] sequential Dial() commands
Hi,How do I cause the dial plan to dial a different extension if the first either never picks up or presses ignore or what have you?For example, something like this:exten = context,1,Dial( SIP/[EMAIL PROTECTED])exten = context,2,Dial(SIP/[EMAIL PROTECTED])Currently, if the first number doesn't answer, the session is closed.ThanksMark ___ --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] Cisco CCM - Asterisk
Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages everytime I send a call. Had play a lot with the way SIP messages are sent to the Cisco, but always been unseccessful. I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to "enable" SIP messageing/reception in the Cisco. Regards, Alyed ___ --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] Inbound Callcenter with multiple DIDs
If you wanted to everything manually it could be done. I would use asterisk real time. Never worked with any specific programs that are designed for this so I can't reccoemnd one. - Original Message - From: Michael Sampson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 4:44 PM Subject: [asterisk-users] Inbound Callcenter with multiple DIDs I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like accounts 1 - 100 are in group 1, 101 - 200 are in group two. Some agents can get calls from group 1, some from group 2 and some from both groups. Most solutions I have found are meant for inbound call centers that handle only a few types of calls and have little need to make large distinctions between different DIDs. I have played around with QueueMetrics and it is a good piece of software, but does not handle the DIDs the way I need. Really any recommendations for software to go with asterisk that inbound call centers are using and find useful would be great. -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --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] Echo Cancel Cards
I second that. I had card from Sangoma with echo can. When ever the echo can. was enabled ZAP/2 would work with one way audio. Sangoma had a tech ssh into my box for a few hours ar no charge. It was the first time they saw such a problem with the card the supplier sent me a new one the next day. - Original Message - From: John Novack [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 3:38 PM Subject: Re: [asterisk-users] Echo Cancel Cards The Sangoma single port T1 card also works well, and , along with the 5 year warranty, works with just about any MB that it will fit, makes it a no brainer choice over the Digium products. Sangoma just doesn't say try another motherboard. If their product doesn't work, they find out why and fix it! John Novack Dovid B wrote: I have never used T1 cards but as far as POTS line cards I would say that I like sangoma better. It is a little bit harder to set up but works wonders. - Original Message - From: Thomas Kenyon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 12:31 PM Subject: Re: [asterisk-users] Echo Cancel Cards Joseph wrote: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. I've got an SPA-3000 at home that is constantly crashing, echoey and is almost unusable. (The CS4660-based ATA and PA1688-based handsets have otherwise been fine, as were the the Cisco 468 and Linysys PAP2 when they were in use). ___ --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 ___ --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] Inbound Callcenter with multiple DIDs
Hi Michael, do you want to do the reporting or to configure the dialplan? QueueMetrics will do the reporting for no matter how many ACD queues, and will automatically sync to the underlying * config files, so there should be no problem with reporting. You can also configure it in self-service mode, so that each owner of the 1000 DIDs can log in individually and pull stats or real-time reports for their own DID. Hope this helps l. In data Tue, 10 Oct 2006 16:44:04 +0200, Michael Sampson [EMAIL PROTECTED] ha scritto: I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like accounts 1 - 100 are in group 1, 101 - 200 are in group two. Some agents can get calls from group 1, some from group 2 and some from both groups. Most solutions I have found are meant for inbound call centers that handle only a few types of calls and have little need to make large distinctions between different DIDs. I have played around with QueueMetrics and it is a good piece of software, but does not handle the DIDs the way I need. Really any recommendations for software to go with asterisk that inbound call centers are using and find useful would be great. -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --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] Voicemail Press '0'
-Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Press '0' Douglas Garstang wrote: Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention Whatever that means... My dialplan has: [foo] exten = 556,1,Answer exten = 556,n,Voicemail([EMAIL PROTECTED]) [default] exten = o,n,Playback(tt-monkeys) My voicemail.conf has: [general] operator=yes [default] 3254101 = 1234,Foo When I press '0' nothing happens. Nothing is displayed on the console to indicate any attempt to dial 'o'. Put exten = o in the same context as Voicemail. I don't know if you can include = the context it is in or not. G. It's a fussy bastard. I took what I finally got working in a simple test scenario and tried to apply that to production and it ain't working. Doug. ___ --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] sequential Dial() commands
Mark Price wrote: Hi, How do I cause the dial plan to dial a different extension if the first either never picks up or presses ignore or what have you? For example, something like this: exten = context,1,Dial(SIP/[EMAIL PROTECTED]) exten = context,2,Dial(SIP/[EMAIL PROTECTED]) Currently, if the first number doesn't answer, the session is closed. Thanks Mark You need to specify a timeout on at least the first dial command. From 'show application dial': Unless there is a timeout specified, the Dial application will wait indefinitely until one of the called channels answers, the user hangs up, or if all of the called channels are busy or unavailable. Dialplan executing will continue if no requested channels can be called, or if the timeout expires. 'show application dial' on the command line will tell you how and where to put the necessary options. -Dave ___ --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] Cisco CCM - Asterisk
Looks like the CallManager is unable to find the endpoint in its database. Make sure asterisk trunk on the Call manager is in the same calling Search Space as the phones are in, or make sure there is access between the calling search spaces -Eric -- Original message -- From: Alyed Tzompa [EMAIL PROTECTED] Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integrat ion but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages everytime I send a call. Had play a lot with the way SIP messages are sent to the Cisco, but always been unseccessful. I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to enable SIP messageing/reception in the Cisco. Regards, Alyed ---BeginMessage--- Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages everytime I send a call. Had play a lot with the way SIP messages are sent to the Cisco, but always been unseccessful. I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to "enable" SIP messageing/reception in the Cisco. Regards, Alyed ___ --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 ---End Message--- ___ --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] Voicemail Press '0'
Douglas Garstang wrote: -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Press '0' Douglas Garstang wrote: Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention Whatever that means... My dialplan has: [foo] exten = 556,1,Answer exten = 556,n,Voicemail([EMAIL PROTECTED]) [default] exten = o,n,Playback(tt-monkeys) My voicemail.conf has: [general] operator=yes [default] 3254101 = 1234,Foo When I press '0' nothing happens. Nothing is displayed on the console to indicate any attempt to dial 'o'. Put exten = o in the same context as Voicemail. I don't know if you can include = the context it is in or not. G. It's a fussy bastard. I took what I finally got working in a simple test scenario and tried to apply that to production and it ain't working. In Asterisk you are NEVER supposed to be able to access a different context without an explicit Goto, include =. There are a few applications that let you access another context, but you must still specify it somewhere. ___ --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] sequential Dial() commands
exten = context,1,Dial( SIP/[EMAIL PROTECTED]) exten = context,2,Dial(SIP/[EMAIL PROTECTED]) Currently, if the first number doesn't answer, the session is closed. Specify a time out. Without it * will not continue to priority 2 if [EMAIL PROTECTED] is reachable but does not answer. exten = context,1,Dial(SIP/[EMAIL PROTECTED],20) exten = context,2,Dial(SIP/[EMAIL PROTECTED],20) ___ --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] Cisco CCM - Asterisk
I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to enable SIP messageing/reception in the Cisco. Yeah, pretty sure to enable it you're going to have to upgrade to CCM 5. I could be wrong, that has been known to happen, once. ___ --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] Re: Voicemail Press '0'
Alternativly you can use the exitcontext parameter in the voicemail.conf to define a separate context in your extensions.conf where the o or a extensions are handled. Douglas Garstang [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Press '0' Douglas Garstang wrote: Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention Whatever that means... My dialplan has: [foo] exten = 556,1,Answer exten = 556,n,Voicemail([EMAIL PROTECTED]) [default] exten = o,n,Playback(tt-monkeys) My voicemail.conf has: [general] operator=yes [default] 3254101 = 1234,Foo When I press '0' nothing happens. Nothing is displayed on the console to indicate any attempt to dial 'o'. Put exten = o in the same context as Voicemail. I don't know if you can include = the context it is in or not. G. It's a fussy bastard. I took what I finally got working in a simple test scenario and tried to apply that to production and it ain't working. Doug. ___ --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] 1.4 and slow sound playback
I am testing 1.4 and am having trouble with the sound files. The gsm files are much larger than they used to be. Sox (12.18.2) plays them back really sllo. Apparently it thinks the sampling rate is 8000. When I specify -r 48000 it play back properly. I mention the sox behavior because Asterisk plays them back the same way sox does, very slowly. I am using the ulaw codec and I installed the ulaw sound files. Asterisk still plays the sound very slowly. I don't know if it is using the ulaw files or gsm files. How do I tell Asterisk to use the right sampling rate? Bill ___ --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] transfer from VM to Cell Phone
Hi Folks, I'm not sure if this is possible, but I'd like to give users the option of transfering to an employee's cell phone when they get to their greeting. This is a feature that is common on Nortel KSUs. Is there an easy way to do this on a per employee basis? I can see how it can be done globally. TIA Brian, ___ --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] 1.4 and slow sound playback
I have seen this if you do not include -c1 for stereo audio files. --Brian On Tue, Oct 10, 2006 at 02:31:59PM -0400, Bill Merriam wrote: I am testing 1.4 and am having trouble with the sound files. The gsm files are much larger than they used to be. Sox (12.18.2) plays them back really sllo. Apparently it thinks the sampling rate is 8000. When I specify -r 48000 it play back properly. I mention the sox behavior because Asterisk plays them back the same way sox does, very slowly. I am using the ulaw codec and I installed the ulaw sound files. Asterisk still plays the sound very slowly. I don't know if it is using the ulaw files or gsm files. How do I tell Asterisk to use the right sampling rate? Bill ___ --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] Increase VoiceMail Messages Recording Gain - Audio Calls are Ok
Hi allI'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions.On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:)The problem is:Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls.I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience?-- Best regardsMarco Mouta ___ --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] Re: Voicemail Press '0'
Is that only available in 1.4? 'exitcontext' does not exist anywhere in my default 1.2.x voicemail.conf file. -Original Message- From: LJ [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 12:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Voicemail Press '0' Alternativly you can use the exitcontext parameter in the voicemail.conf to define a separate context in your extensions.conf where the o or a extensions are handled. Douglas Garstang [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Press '0' Douglas Garstang wrote: Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention Whatever that means... My dialplan has: [foo] exten = 556,1,Answer exten = 556,n,Voicemail([EMAIL PROTECTED]) [default] exten = o,n,Playback(tt-monkeys) My voicemail.conf has: [general] operator=yes [default] 3254101 = 1234,Foo When I press '0' nothing happens. Nothing is displayed on the console to indicate any attempt to dial 'o'. Put exten = o in the same context as Voicemail. I don't know if you can include = the context it is in or not. G. It's a fussy bastard. I took what I finally got working in a simple test scenario and tried to apply that to production and it ain't working. Doug. ___ --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 ___ --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] Re: Voicemail Press '0'
I was able to get pressing '0' while in voicemail to work in a simple test
case, but was unable in a more complicated scenario. Here's a stripped down,
sanitized version of that complex scenario...
[start] ; phones start here
include = some_contexts
include = some_more_contexts
include = route
[route]
; When findme/follome is finished, the script dials
Local/${EXTEN}global_vmdeposit for vm deposit
exten = _[*0123456789].,n,AGI(ipt/originator.py)
[global_vmdeposit]
exten = _[sub].,1,Answer
exten = _[sub].,2,Wait,1
exten = _[sub].,3,Voicemail([EMAIL PROTECTED])
I tried putting the 'o' extension in all three contexts there and it worked in
none of them. I don't know if dialling Local is screwing it up, or if because
we dialled Local from an AGI script, or maybe because of the includes or what...
Doug
-Original Message-
From: LJ [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 12:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Voicemail Press '0'
Alternativly you can use the exitcontext parameter in the
voicemail.conf
to define a separate context in your extensions.conf where
the o or a
extensions are handled.
Douglas Garstang [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 10, 2006 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Press '0'
Douglas Garstang wrote:
Crikey. I can't get this to work!
Allegedly, you can press 0 while in the voicemail greeting
and be dropped to the 'o' extension. For some reason, I can't
get it to work. The 'docs' aren't clear about what context
the o extension should be in. The voip wiki says
the context for the voicemail box that we're looking for
in the dialplan for the jump to the 'a' or 'o' extention
Whatever that means...
My dialplan has:
[foo]
exten = 556,1,Answer
exten = 556,n,Voicemail([EMAIL PROTECTED])
[default]
exten = o,n,Playback(tt-monkeys)
My voicemail.conf has:
[general]
operator=yes
[default]
3254101 = 1234,Foo
When I press '0' nothing happens. Nothing is displayed on
the console to indicate any attempt to dial 'o'.
Put exten = o in the same context as Voicemail. I don't
know if you
can include = the context it is in or not.
G. It's a fussy bastard. I took what I finally got
working in a simple
test scenario and tried to apply that to production and it
ain't working.
Doug.
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[asterisk-users] Hangup or busy when the peer answer outgoing calls
Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: - loadzone = es defaultzone=es fxsks=1 zapata.conf -- [channels] signalling=fxs_ks busydetect=yes answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=yes progzone=es context = contexto group = 1 channel = 1 And this is the asterisk log: -- Executing Dial(SIP/200-4803, ZAP/1/966736800|90) in new stack -- Called 1/966736800 -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing PlayTones(SIP/200-4803, busy) in new stack -- Executing Wait(SIP/200-4803, 10) in new stack == Spawn extension (indeos, 0966736800, 103) exited non-zero on 'SIP/200-4803' Thanks all Eloy. -- Indeos Consultoria ___ --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] RE: Welcome to the asterisk-users mailing list
Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine on the same box. Any ideas? ___ --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] How big is *your* dialplan??
Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be interesting... For example: Digium's dialplan is roughly 50 contexts, 304 total extensions, 870 total priorities. My home system has 100 contexts, 400 total extensions, 935 total priorities. My biggest extension has 129 priorities... no inflation or useless cruft there, either... mostly. These would seem small compared to some dialplans out there, I'll bet. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ --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] sequential Dial() commands
Simple
Exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],90) ; This will ring thier phone
for 90 seconds
Exten =1234,2,Noop("If user dosent pick
up do something here")
Exten = 1234,102,Dial(SIP/[EMAIL PROTECTED],90) ; WIll ring user B if User is
Busy or hits the reject button
Exten = 1234,103,Noop("If user dosent pick up
do something here")
Exten = 1234,203,Noop("If user B is busy or
rejects call do something here")
- Original Message -
From:
Mark Price
To: Asterisk Users
Sent: Tuesday, October 10, 2006 7:28
PM
Subject: [asterisk-users] sequential
Dial() commands
Hi,How do I cause the dial plan to dial a different
extension if the first either never picks up or presses ignore or what have
you?For example, something like this:exten = context,1,Dial( SIP/[EMAIL PROTECTED])exten = context,2,Dial(SIP/[EMAIL PROTECTED])Currently, if the first
number doesn't answer, the session is closed.ThanksMark
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Re: [asterisk-users] transfer from VM to Cell Phone
You can create a macro that tells the caller that the user is unavailable. It then asks them if they want to go to the usersVM or be transfred to thier cell phone. I also created a macro where users can dial an extension and set thier mobile number. Let me know if you want it. - Original Message - From: Mr. Jones [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 8:45 PM Subject: [asterisk-users] transfer from VM to Cell Phone Hi Folks, I'm not sure if this is possible, but I'd like to give users the option of transfering to an employee's cell phone when they get to their greeting. This is a feature that is common on Nortel KSUs. Is there an easy way to do this on a per employee basis? I can see how it can be done globally. TIA Brian, ___ --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] How big is *your* dialplan??
Steve Murphy wrote: Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? Sounds interesting. Small facility of 60 users: -= 161 extensions (597 priorities) in 59 contexts. =- -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] How big is *your* dialplan??
Do you want single server stats, or cluster stats? Single server: -= 1004 extensions (1403 priorities) in 45 contexts. =- Aaron On Tue, 2006-10-10 at 14:16 -0600, Steve Murphy wrote: Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be interesting... For example: Digium's dialplan is roughly 50 contexts, 304 total extensions, 870 total priorities. My home system has 100 contexts, 400 total extensions, 935 total priorities. My biggest extension has 129 priorities... no inflation or useless cruft there, either... mostly. These would seem small compared to some dialplans out there, I'll bet. murf ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] How big is *your* dialplan??
Last machine that I set up is roughly 30 contexts 400 priorotys and 20 extensions. Did it on a dual core 3.0 with 2 gigs of ram and raid 1 sata. System is a bit of an over kill but client wanted it. Works like a charm. I know it's not a match for what you have but I figured I would throw it out there. - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 10:16 PM Subject: [asterisk-users] How big is *your* dialplan?? ___ --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] How big is *your* dialplan??
Steve is their a CLI command you can make from the console that will tell you the answer? LOL or are we expected to count? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Tuesday, 10 October 2006 4:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How big is *your* dialplan?? Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be interesting... For example: Digium's dialplan is roughly 50 contexts, 304 total extensions, 870 total priorities. My home system has 100 contexts, 400 total extensions, 935 total priorities. My biggest extension has 129 priorities... no inflation or useless cruft there, either... mostly. These would seem small compared to some dialplans out there, I'll bet. murf -- Steve Murphy Software Developer Digium ___ --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
