Re: [asterisk-users] Re: Moh stops immediately
Mac OS X, Asterisk 1.4 beta --- Martin Joseph [EMAIL PROTECTED] wrote: On 2006-11-12 23:08:05 -0800, zen Perry [EMAIL PROTECTED] said: I'm trying to set up the Music on Hold feature. However, when I place a call the moh starts and stops immediately and as a result I dont hear the audio. -- Started music on hold, class 'default', on channel 'SIP/XXX' -- Stopped music on hold on SIP/XXX NOTICE[380]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! My extensions.conf reads: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) I've also tried: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) exten = 2000,3,WaitMusicOnHold(20) exten = 2000,4,Hangup Which OS? Which asterisk? Mine does this also. Marty ___ --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 Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.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] Re: IAX2 one way audio
On 2006-11-12 14:48:13 -0800, joe a. ([EMAIL PROTECTED]) [EMAIL PROTECTED] said: Experiencing one way audio using IAX2. I did see some other posts on this, and see there may be some internal issues with asterisk and one way audio. Can this be a widespread problem? So many seem to be using IAX, I find it puzzling. Some information points to this being a problem on asymmetrical connections . This is a decidedly asymmetrical connection, with 1.5 Mbs download and 256 kbs, upload. A satellite link, to boot. So, maybe this is a meltdown right from the start? Event the vendor of the IAX service was not too keen. There is an open bug reports on IAX2 audio going one way after a spike in lag. This seemed to be connected to the newer jitter buffer, but it isn't clear yet really. Oddly, my first few connections worked fine (unexpectedly good audio, both ways). Being all happy and stuff, made a call to a client, to show off.Yep. could not hear me. Since then all calls have connected quickly, but are receive only. I've tried rebooting the asterisk box, changing jitter related stuff, no joy. It is behind a firewall, but I can see no packets dropped, related to the IP's involved. Anyway, if there are experiences to relate, please do. Don't know really, but it does sound similar to the open bug report. Marty ___ --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] Can i have two asterisk vcersions running on same PC??
On Mon, Nov 13, 2006 at 12:45:40PM +0530, Sri Keerthy wrote: Can two versions of asterisk run on same PC?? Basically, yes: use a custom asterisk.conf (or custom compiled defaults) to use different pathes for just about anything. Note, however, that only one program can listen on the same port. So if you want both to e.g., listen on IAX, one, at least, has to listen on a custom port. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] operator console
Could this be considered spam ? I believe this is second threas realted to that pbx .On 13/11/06, Jordi Nelissen [EMAIL PROTECTED] wrote: Check out the ESCAUX net.PBX operator console. In use in variouscompanies with 200+ extensions. Powerfull and convenient. http://www.escaux.com//index.php?option=com_contenttask=viewid=61Itemid=350Best Regards,Jordi--www.escaux.comBusiness IP Telephony Forrest Beck wrote: Talk to the folks at Asteria.The have a product called Reign.It looks just like your old interface, runs off .NET as a client on the machine. http://www.asteriasgi.com/pbx/reign On 11/7/06, Stephen Wingfield [EMAIL PROTECTED] wrote: Andres, The Bicom Systems Operator Panel is probably what you are looking for. OPCOM http://www.bicomsystems.com/docs/opcom/1.0/html/ This is included with every copy of PBXware and is fully supported. If you care to register you may order a trial of PBXware with our SOHO. Regards Steve steve 'at' bicomsystems 'dot' com - Original Message - From: Andres Paglayan To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 5:27 PM Subject: [asterisk-users] operator console Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and extension,and voila. I'll be switching the company's phone system to Asterisk. I know * is way much more flexible and rich featured than the box we currently have, ...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? I don't mind using a commercial product, if the only part we have to pay for is the gui, besides, we will buying the enterprise * version Thanks a bunch, Andres ___ --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--Jordi NelissenE S C A U XBusiness IP Telephony www.escaux.com--Email from people at escaux.com does not usually represent officialpolicy of ESCAUX. See http://www.escaux.com/disclaimer for details.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] IAX2 one way audio
I am not sure if it will help but try to put notansfer=yes in ur iax2 extension (just experiment a bit ;) ).On 12 Nov 2006 17:48:13 -0500, joe a. ( [EMAIL PROTECTED]) [EMAIL PROTECTED] wrote: Experiencing one way audio using IAX2.I did see some other posts on this, and see there may be some internal issues with asterisk and one way audio.Can this be a widespread problem?So many seem to be using IAX, I find it puzzling. Some information points to this being a problem on asymmetrical connections.This is a decidedly asymmetrical connection, with 1.5 Mbs download and 256 kbs, upload. A satellite link, to boot.So, maybe this is a meltdown right from the start?Event the vendor of the IAX service was not too keen. Oddly, my first few connections worked fine (unexpectedly good audio, both ways).Being all happy and stuff, made a call to a client, to show off. Yep.could not hear me.Since then all calls have connected quickly, but are receive only.I've tried rebooting the asterisk box, changing jitter related stuff, no joy. It is behind a firewall, but I can see no packets dropped, related to the IP's involved.Anyway, if there are experiences to relate, please do.joe a.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Sending '#' with Dial
Hi! I have a working asterisk-setup with four sip-clients. Everything works great but when the users call someone the phonenumber shows up on the receiving ends callerid-display. To correct this my provider told me to send #31# before the phonenumber, tried this with: Dial(SIP/[EMAIL PROTECTED]) but my asterisk tells me that it isn't a valid extension. The INVITE looks fine, '#31#phonenumber@provider' but my provider then sends SIP/2.0 404 Not Found back to me. Any thoughts? /e -- http://hostname.nu/~emil ___ --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] several behind NAT
Also, where can I get information on provisioning? These phones will be out of my hands soon and I'd like to be able to update the configs. I saw a few utilities for generating the configs, but I'd like more specific info - I don't mind editing files by hand but want to know how it works. Does anyone have some resources? Check the Grandstream website for a java-based provisioning tool for Linux Cameron ___ --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: Moh stops immediately
On 2006-11-12 23:08:05 -0800, zen Perry [EMAIL PROTECTED] said: I'm trying to set up the Music on Hold feature. However, when I place a call the moh starts and stops immediately and as a result I dont hear the audio. -- Started music on hold, class 'default', on channel 'SIP/XXX' -- Stopped music on hold on SIP/XXX NOTICE[380]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! My extensions.conf reads: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) I've also tried: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) exten = 2000,3,WaitMusicOnHold(20) exten = 2000,4,Hangup Which OS? Which asterisk? Mine does this also. Marty ___ --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] Can i have two asterisk vcersions running on same PC??
Tzafrir Cohen wrote: Note, however, that only one program can listen on the same port. So if you want both to e.g., listen on IAX, one, at least, has to listen on a custom port. ...or you need to run two IP addresses on the machine, and configure each Asterisk installation to use the different IP addresses. ___ --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] bindport
Is there any way to let asterisk listen to two different ports 5060 and 5061 for example , or this can be done from iptables firewall Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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: Moh stops immediately
What is the length of music on old mp3 file ? Maybe file is very short .On 13/11/06, zen Perry [EMAIL PROTECTED] wrote: Mac OS X, Asterisk 1.4 beta--- Martin Joseph [EMAIL PROTECTED] wrote: On 2006-11-12 23:08:05 -0800, zen Perry [EMAIL PROTECTED] said: I'm trying to set up the Music on Hold feature. However, when I place a call the moh starts and stops immediately and as a result I dont hear the audio. -- Started music on hold, class 'default', on channel 'SIP/XXX' -- Stopped music on hold on SIP/XXX NOTICE[380]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! My extensions.conf reads: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) I've also tried: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) exten = 2000,3,WaitMusicOnHold(20) exten = 2000,4,Hangup Which OS? Which asterisk? Mine does this also. Marty ___ --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 Cheap talk?Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Slow to get dialtone when going off hook - bigproblem for me :(
You can also use waitexten = X,1,Wait(3)(for3secs) On 13/11/06, Jim Archer [EMAIL PROTECTED] wrote: --On Sunday, November 12, 2006 11:53 PM -0500 John Novack[EMAIL PROTECTED] wrote: Dovid B wrote: snip How hard would it be to have asterisk detect a dial tone ? I really can't say. I am not a C programmer, so I wouldn't even know where to start. Given that cheap dial up modems have, for the past ??20?? years, have been able to do just that, I would think it should have been an early consideration For those 1% of users, the last time I tried, the insertion of a whad no effect for pulse dialing either.Well thanks to everyone who responded, and thanks to multiple w's I am back in operation.I went off hook a bunch of times and the worst case seemedto be 3 seconds to get a dial tone (which is pretty bad).It's hard togoogle one letter, but I eventually found that each w is .5 seconds, so 7 w's were inserted to be safe.I also called Cox and griped but I doubtthat will do me any good.I am a C programmer, but I don't know anything about the inards ofAsterisk.However, I would expect that dial tone detection would be a function of the hardware, not the Asterisk software.The cheap modems dothis on board and export a simple command set.But I also don't knowanything about Digium's hardware either.Thanks again!I really appreciate the help! Jim___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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 and realtime : the emailbody option ...
Dear all, I've just a little question ... I've configured asterisk to run with voicemail realtime in the extconfig.conf like this : voicemail = mysql,database,voicemail I just want to have a row, in the voicemail table, like emailbody, which is capable to give me a body to the mail. Therefor the body is set user by user, and it could be very fun. Is it possible ? I've made some test of course, without success ... Thanks ! -- Jean-Baptiste Bellet Ingénieur Développpement Lucyde SAS Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex +33 (0)5 34 31 86 36 http://www.lucyde.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] Dial : Executing context/priority after bridge?
Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file when the user picks the phone. I use Gizmo service for SIP and I'm able to call through it. However, when asterisk dials a number, Gizmo first answers then tries bridging 2 channels. Right after answer Asterisk starts playing the reminder.It obviously results in hearing silence when the call is bridged (you've picked the phone). I use Asterisk cmd Dial like this : exten = s,1,Dial(SIP/NUMBER,30,rA(announce)) which should play file announce to the called party once they answer. I also tried exten = s,1,Dial(SIP/NUMBER,30,rG(default^play^1)) which separates caller and callee,for the same purpose. Here's the asterisk console: -- Executing SetCallerID(SIP/sipphone-cbfb, NAME NUMBER) in new stack -- Executing NoOp(SIP/sipphone-cbfb, Dialing 011 to deliver file /usr/vt/result/200611135/test) in new stack -- Executing SetVar(SIP/sipphone-cbfb, __MSG=/usr/vt/result/200611135/98_011380673805838) in new stack -- Executing Dial(SIP/sipphone-cbfb, SIP/[EMAIL PROTECTED]|45|rA(/usr/vt/result/200611135/test)) in new stack -- Called [EMAIL PROTECTED] -- SIP/sipphone-ebf3 answered SIP/sipphone-cbfb -- Playing '/usr/vt/result/200611135/98_011380673805838' (language 'en') -- Attempting native bridge of SIP/sipphone-cbfb and SIP/sipphone-ebf3 It looks like Asterisk is starting to execute context/play announcement after the dialed channel answers. The problem is that Gizmo SIP first answers and then tries bridging (that's where the actual call is taking place), so my announcement is played before the call and when I pick up I just hear the silence. Is there a workaround or a way to make Asterisk play the message when the call is bridged?I use Asterisk CVS-HEAD built on 28 Oct 2006.Any advice is highly appreciated.Yuri PS. I tried this on my local server with a local SIP account, and the bridge step was absent. So it worked. Is it then a Gizmo issue or a standard SIP way? ___ --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] Dial : Executing context/priority after bridge?
Put canreinvite=no in asterisk sip user extension.Someprovidersdonotsupportreinvitesandhenceyougetsilenceiguess. On 13/11/06, Yuri Veremeyenko [EMAIL PROTECTED] wrote: Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file when the user picks the phone. I use Gizmo service for SIP and I'm able to call through it. However, when asterisk dials a number, Gizmo first answers then tries bridging 2 channels. Right after answer Asterisk starts playing the reminder.It obviously results in hearing silence when the call is bridged (you've picked the phone). I use Asterisk cmd Dial like this : exten = s,1,Dial(SIP/NUMBER,30,rA(announce)) which should play file announce to the called party once they answer. I also tried exten = s,1,Dial(SIP/NUMBER,30,rG(default^play^1)) which separates caller and callee,for the same purpose. Here's the asterisk console: -- Executing SetCallerID(SIP/sipphone-cbfb, NAME NUMBER) in new stack -- Executing NoOp(SIP/sipphone-cbfb, Dialing 011 to deliver file /usr/vt/result/200611135/test) in new stack-- Executing SetVar(SIP/sipphone-cbfb, __MSG=/usr/vt/result/200611135/98_011380673805838) in new stack -- Executing Dial(SIP/sipphone-cbfb, SIP/[EMAIL PROTECTED]|45|rA(/usr/vt/result/200611135/test)) in new stack-- Called [EMAIL PROTECTED]-- SIP/sipphone-ebf3 answered SIP/sipphone-cbfb -- Playing '/usr/vt/result/200611135/98_011380673805838' (language 'en')-- Attempting native bridge of SIP/sipphone-cbfb and SIP/sipphone-ebf3 It looks like Asterisk is starting to execute context/play announcement after the dialed channel answers. The problem is that Gizmo SIP first answers and then tries bridging (that's where the actual call is taking place), so my announcement is played before the call and when I pick up I just hear the silence. Is there a workaround or a way to make Asterisk play the message when the call is bridged? I use Asterisk CVS-HEAD built on 28 Oct 2006.Any advice is highly appreciated.YuriPS. I tried this on my local server with a local SIP account, and the bridge step was absent. So it worked. Is it then a Gizmo issue or a standard SIP way? ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Desktop integration
Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --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 IVR functionality
Hi i have an application developed with bayonne. Recentely i'm experiencing some problems and i am planning to migrate to asterisk. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) - TTS - record exploration (for example, check if some resources are available in the database, and list them to the user (via TTS)) Are these things possible? ___ --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] Can i have two asterisk vcersions running on same PC??
Yes, it is possible also if you use chroots under linux ( http://en.wikipedia.org/wiki/Chroot ). But you'll need to change the listening ports. Regards, Pavel Siderov Tzafrir Cohen wrote: On Mon, Nov 13, 2006 at 12:45:40PM +0530, Sri Keerthy wrote: Can two versions of asterisk run on same PC?? Basically, yes: use a custom asterisk.conf (or custom compiled defaults) to use different pathes for just about anything. Note, however, that only one program can listen on the same port. So if you want both to e.g., listen on IAX, one, at least, has to listen on a custom port. ___ --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] Desktop integration
Basically here is what the application you need has to do:1. take the number you paste2. make a call file 3. drop it in the outgoing spool directory of asteriskThis could be easily done in php - just one page. Donno if any app already exists (have heard of many, but not sure if they come alone or as part of other apps). But writing one should not take much time. -VijOn 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote: Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Where there is a willI want my name in 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] Desktop integration
Ondrej Valousek napisał(a): Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. I have done this by simple CGI program that creates call file. I put asterisk and HTTP server (apache) on the same machine. In CGI program I also do some number normalization, so if user sends me +48 91 123 45 67 I remove spaces and other special chars and add Zap/g1/ prefix. Be aware of file permissions and do not create call file on place, but rather create it in tmp directory and than move it to final spool directory. Regards, Michał Niklas ___ --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 IVR functionality
Yes, you can but you need to use external software for TTS. nik600 wrote: Hi i have an application developed with bayonne. Recentely i'm experiencing some problems and i am planning to migrate to asterisk. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) - TTS - record exploration (for example, check if some resources are available in the database, and list them to the user (via TTS)) Are these things possible? ___ --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
Fwd: [asterisk-users] Asterisk IVR functionality
On 11/13/06, nik600 [EMAIL PROTECTED] wrote: i have an application developed with bayonne. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) Yes, you have to write AGI scripts to do this. - TTS No idea. - record exploration (for example, check if some resources are available in the database, and list them to the user (via TTS)) Should be possible via AGI. raj ___ --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 IVR functionality
On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote: Hi i have an application developed with bayonne. Recentely i'm experiencing some problems and i am planning to migrate to asterisk. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) Asterisk uses a system called AGI to provide IVR. For more information see http://www.voip-info.org/wiki-Asterisk+AGI. As such, IVR applications are just scripts. You can use whatever the underlying platform supports. - TTS Text to Speech in Asterisk is supported by Festival (http://www.voip-info.org/wiki/view/Festival). - record exploration (for example, check if some resources are available in the database, and list them to the user (via TTS)) Again, this just done by your script. If you can do it in the underlying langugae, you can do it in your IVR app. Are these things possible? ___ --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] operator console
Vicky, my other post related to a Web GUI for asterisk. This post is related to an Operator Console. I am simply answering the user's question, so I don't see why you would consider this to be spam, and I never read you can not send two mails to the list on the same day. Jordi Vicky wrote: Could this be considered spam ? I believe this is second threas realted to that pbx . On 13/11/06, *Jordi Nelissen* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Check out the ESCAUX net.PBX operator console. In use in various companies with 200+ extensions. Powerfull and convenient. http://www.escaux.com//index.php?option=com_contenttask=viewid=61Itemid=350 http://www.escaux.com//index.php?option=com_contenttask=viewid=61Itemid=350 Best Regards, Jordi -- www.escaux.com http://www.escaux.com Business IP Telephony Forrest Beck wrote: Talk to the folks at Asteria. The have a product called Reign. It looks just like your old interface, runs off .NET as a client on the machine. http://www.asteriasgi.com/pbx/reign On 11/7/06, Stephen Wingfield [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Andres, The Bicom Systems Operator Panel is probably what you are looking for. OPCOM http://www.bicomsystems.com/docs/opcom/1.0/html/ This is included with every copy of PBXware and is fully supported. If you care to register you may order a trial of PBXware with our SOHO. Regards Steve steve 'at' bicomsystems 'dot' com - Original Message - From: Andres Paglayan To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 5:27 PM Subject: [asterisk-users] operator console Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and extension, and voila. I'll be switching the company's phone system to Asterisk. I know * is way much more flexible and rich featured than the box we currently have, ...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? I don't mind using a commercial product, if the only part we have to pay for is the gui, besides, we will buying the enterprise * version Thanks a bunch, Andres ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jordi Nelissen E S C A U X Business IP Telephony www.escaux.com http://www.escaux.com -- Email from people at escaux.com http://escaux.com does not usually represent official policy of ESCAUX. See http://www.escaux.com/disclaimer http://www.escaux.com/disclaimer for details. ___ --Bandwidth and Colocation provided by Easynews.com http://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:
Re: [asterisk-users] Asterisk IVR functionality
On 11/13/06, Pavel Siderov [EMAIL PROTECTED] wrote: Yes, you can but you need to use external software for TTS. ok, thanks can you suggest me some example for the database interaction? for example, how can i connect to the database? how can i make a query and list the records found? 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] Asterisk IVR functionality
On 11/13/06, Brian Rogan [EMAIL PROTECTED] wrote: On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote: Hi i have an application developed with bayonne. Recentely i'm experiencing some problems and i am planning to migrate to asterisk. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) Asterisk uses a system called AGI to provide IVR. For more information see http://www.voip-info.org/wiki-Asterisk+AGI. As such, IVR applications are just scripts. You can use whatever the underlying platform supports. 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
[asterisk-users] Question about the GUI for 1.4
Hi, I havent tested this yet, but I am just wondering what are the advantages of using this GUI? Does it help you with creating extensions or what? Many thanks, Christian ___ --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] Desktop integration
Hello Michal, Thank you for the hint! Can I ask you for your script so I have some idea how it works? I have apache already running on my * box. Thank you, Ondrej Michał Niklas wrote: Ondrej Valousek napisał(a): Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. I have done this by simple CGI program that creates call file. I put asterisk and HTTP server (apache) on the same machine. In CGI program I also do some number normalization, so if user sends me "+48 91 123 45 67" I remove spaces and other special chars and add "Zap/g1/" prefix. Be aware of file permissions and do not create call file "on place", but rather create it in tmp directory and than move it to final spool directory. Regards, Michał Niklas ___ --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] sip forward behind a nat
On 11/12/06, nik600 [EMAIL PROTECTED] wrote: On 11/12/06, Vicky [EMAIL PROTECTED] wrote: Yep make the server with dynamic ip register to server with static ip ( sip or iax both will do but in sip keep nat=yes while making extension ) the problem is that the server with dynamic ip can't register on the other server! This is the situation: Server with SIP application (public_address) | | - - - Internet | | Firewall (NAT) | | Server Asterisk (private ip:192.168.100.249/public ip:public_address_2) | Analogic Board | Telecom I want to make a call from Server Asterisk to the server with SIP Application. The SIP Application can't register to Server Asterisk (because the application can't do it, i know, it isn't a good thingbut this is the application) When The SIP Application receives a SIP call it responds (because a dummy SIP user is autoregistered on hisself) So i only have to make a call to SIP/[EMAIL PROTECTED] I've also tried to setup an asterisk server on my laptop, and make a call to SIP/[EMAIL PROTECTED] from the public_address network. It works! I only have to setup the Asterisk server in production to make a SIP call throw the NAT but without any SIP user registered on it. Can i do that? Many thanks to all maybe you need some other information? ___ --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] Presence-awareness in Asterisk
Hi,How would you monitor screensaver activity ?Cheers ___ --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: FW: [asterisk-users] Desktop integration
Hi Dean, I will check that site - thanks for the hint. The biggest problem I see with authentication and I do not think mexuar could help me here (and I am definitely going to pay $2000 for it :-) But it is another story... Thank you! Ondrej Dean Collins wrote: Ondrej, You could do it using Mexuar Corraleta but this is a commercial application for Asterisk (US$2,000) http://www.mexuar.com/products_sdk.shtml http://www.mexuar.com/downloads/Press/CorraletaLaunchPR-1-branded.pdf However it has a whole heap more functionality than what you are looking for. If you just want to do 2 legged outbound calls check out ‘call files’ on www.voip-info.org Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Ondrej Valousek Sent: Monday, 13 November 2006 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Desktop integration Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --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 to get CDR to show answered calls only
Why is it awful ?Regards ___ --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] Problem with internet down
Hello peoples, I have a grave problem. In my work i have an asterisk functioning perfect. However whenever the link of internet falls the even for of function. For that everything come back to the normal necessary one remove the trunk sip. Someone knows say me as contour that situation? Even without internet obtain utililzar the trunck PSTN and the internal extensions without be necessary remove the trunck sip?? I thank to all of the help Andre Luiz Martins [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
Re: [asterisk-users] Problem with internet down
What are you using for your Internet connection? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Andre Luiz Martins wrote: Hello peoples, I have a grave problem. In my work i have an asterisk functioning perfect. However whenever the link of internet falls the even for of function. For that everything come back to the normal necessary one remove the trunk sip. Someone knows say me as contour that situation? Even without internet obtain utililzar the trunck PSTN and the internal extensions without be necessary remove the trunck sip?? I thank to all of the help Andre Luiz Martins [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 Inbound (clean). Database: 0648-0, 11/12/2006 - 11/13/2006 8:46:33 AM ___ --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] Problem with internet down
We have a link dedicated of radio. But that presents problems the times! Al Bochter escreveu: What are you using for your Internet connection? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Andre Luiz Martins wrote: Hello peoples, I have a grave problem. In my work i have an asterisk functioning perfect. However whenever the link of internet falls the even for of function. For that everything come back to the normal necessary one remove the trunk sip. Someone knows say me as contour that situation? Even without internet obtain utililzar the trunck PSTN and the internal extensions without be necessary remove the trunck sip?? I thank to all of the help Andre Luiz Martins [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 Inbound (clean). Database: 0648-0, 11/12/2006 - 11/13/2006 8:46:33 AM ___ --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] Problem with internet down
Well if you want to use VOIP you will have to get a better Internet connection. You can't do anything to the PBX Server to fix this. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Andre Luiz Martins wrote: We have a link dedicated of radio. But that presents problems the times! Al Bochter escreveu: What are you using for your Internet connection? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Andre Luiz Martins wrote: Hello peoples, I have a grave problem. In my work i have an asterisk functioning perfect. However whenever the link of internet falls the even for of function. For that everything come back to the normal necessary one remove the trunk sip. Someone knows say me as contour that situation? Even without internet obtain utililzar the trunck PSTN and the internal extensions without be necessary remove the trunck sip?? I thank to all of the help Andre Luiz Martins [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 Inbound (clean). Database: 0648-0, 11/12/2006 - 11/13/2006 8:46:33 AM ___ --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 Inbound (clean). Database: 0648-0, 11/12/2006 - 11/13/2006 8:55:33 AM ___ --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] Problem with internet down
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Bochter Sent: 13 November 2006 14:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with internet down Well if you want to use VOIP you will have to get a better Internet connection. You can't do anything to the PBX Server to fix this. Best regards, Al Bochter Bochter Services I don't think that's necessarily true on a couple of counts: 1) You can use VoIP in-house and something else out to the rest of the world. 2) The original poster's problems *might* be to do with DNS lookups failing, which *is* something you can solve on the server running Asterisk. ___ --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: FW: [asterisk-users] Desktop integration
Snap will do this for you. (Check my signature) On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote: Hi Dean, I will check that site - thanks for the hint. The biggest problem I see with authentication and I do not think mexuar could help me here (and I am definitely going to pay $2000 for it :-) But it is another story... Thank you! Ondrej Dean Collins wrote: Ondrej, You could do it using Mexuar Corraleta but this is a commercial application for Asterisk (US$2,000) http://www.mexuar.com/products_sdk.shtml http://www.mexuar.com/downloads/Press/CorraletaLaunchPR-1-branded.pdf However it has a whole heap more functionality than what you are looking for. If you just want to do 2 legged outbound calls check out 'call files' on www.voip-info.org Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ondrej Valousek Sent: Monday, 13 November 2006 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Desktop integration Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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: FW: [asterisk-users] Desktop integration
asteriskextras.com has a FREE click to call script that is very popular. On 11/13/06, mitcheloc [EMAIL PROTECTED] wrote: Snap will do this for you. (Check my signature) On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote: Hi Dean, I will check that site - thanks for the hint. The biggest problem I see with authentication and I do not think mexuar could help me here (and I am definitely going to pay $2000 for it :-) But it is another story... Thank you! Ondrej Dean Collins wrote: Ondrej, You could do it using Mexuar Corraleta but this is a commercial application for Asterisk (US$2,000) http://www.mexuar.com/products_sdk.shtml http://www.mexuar.com/downloads/Press/CorraletaLaunchPR-1-branded.pdf However it has a whole heap more functionality than what you are looking for. If you just want to do 2 legged outbound calls check out 'call files' on www.voip-info.org Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ondrej Valousek Sent: Monday, 13 November 2006 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Desktop integration Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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 -- Tom Vile ___ --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 and norstar
Thanks for the response!I've been reading and trying things and I cannot find a way to do a supervised transfer using this topology:pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk Because if I do a flash() and a SendDTMF() to transfer the extension I have to Hungup(), otherwise it never reaches the called extension. So if I do a Hangup() I cannot know of the result of the call.I think that the only way is going to be like this: pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk (fxo zap/2) - ATA - (ext 321) norstarWith that topology I'll be able to do a DIAL() on the other zap (zap/2) and with that know the state of the call. The problem with this topology is that for 5 lines is gonna be expensive and difficult to find 10 ATAs!If you have any suggestions and configurations they will be very appreciated.Thanks!-- Gustavo BermanSysadminDepto. InformaticaUniversidad Nacional del ComahueCentro Regional Universitario Bariloche ___ --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: FW: [asterisk-users] Desktop integration
Hi Mitchel, this looked Very good to me at the first glimpse - then I realized the client is Windows only :-( We have Linux desktops here... Thanks anyway... Ondrej mitcheloc wrote: Snap will do this for you. (Check my signature) On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote: Hi Dean, I will check that site - thanks for the hint. The biggest problem I see with authentication and I do not think mexuar could help me here (and I am definitely going to pay $2000 for it :-) But it is another story... Thank you! Ondrej Dean Collins wrote: Ondrej, You could do it using Mexuar Corraleta but this is a commercial application for Asterisk (US$2,000) http://www.mexuar.com/products_sdk.shtml http://www.mexuar.com/downloads/Press/CorraletaLaunchPR-1-branded.pdf However it has a whole heap more functionality than what you are looking for. If you just want to do 2 legged outbound calls check out 'call files' on www.voip-info.org Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Ondrej Valousek Sent: Monday, 13 November 2006 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Desktop integration Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --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] How to get CDR to show answered calls only
Its pretty easy . If you have mysql records enabled via a patch just do sql queryuse asteriskcdrdb;select * from `cdr` where billsec 0 ( if answered then billsec always greater than 0 or you cna also use disposition = 'ANSWERED' ) On 13/11/06, Olivier [EMAIL PROTECTED] wrote: Why is it awful ?Regards___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Defunct / zombie AGI after some execution time
Hello, We are running Asterisk-1.0.12 in a CentOS 4-4.2 system, kernel 2.6.9-42.0.3.ELsmp. We have some custom AGI, and when we launch Asterisk the system works fine. But **after some time**, each AGI execution generates a zombie defunct process. We believe that it's not a problem in the AGI code, because Asterisk+AGI is working fine in the first n minutes/hours. This is a pstree sample: init-+-asterisk---asterisk---48*[asterisk] But after some execution time, this is the pstree output: init-+-asterisk---asterisk-+-28*[asterisk] | |-asterisk-+-21*[x.agi] | | `-40*[x.agi] | |-5*[asterisk-+-y.agi] | | |-z.agi] (...) Each agi is a defunct process. It dies when the call (parent) finishes. When the first zombie appears, then ALL next AGI launched from Asterisk generates a zombie. We have tested some improvements to solve the problem, with no success: - Upgrade from RedHat 8 to Centos 3.x - Upgrade from Centos 3.x to Centos 4.x - LD_ASSUME_KERNEL=2.4.1 - ulimit -n 65535 - Upgrade from asterisk 1.0.7 to 1.0.12 Currenly we can not easily migrate from asterisk-1.0.x to 1.2.x Any ideas?. Could be Debian a solution? Thank you. ___ --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] Problem with internet down
I believe that the problem really is fault of DNS lookups, but as I should proceed for resolve that?? Steve Langstaff escreveu: -Original Message- I don't think that's necessarily true on a couple of counts: 1) You can use VoIP in-house and something else out to the rest of the world. 2) The original poster's problems *might* be to do with DNS lookups failing, which *is* something you can solve on the server running Asterisk. ___ --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: FW: [asterisk-users] Desktop integration
We use Asterisk Desktop Manager http://adm.hamnett.org/ very successfully with both debian and windows desktops. Bails Ondrej Valousek wrote: Hi Mitchel, this looked Very good to me at the first glimpse - then I realized the client is Windows only :-( We have Linux desktops here... Thanks anyway... Ondrej mitcheloc wrote: Snap will do this for you. (Check my signature) On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote: Hi Dean, I will check that site - thanks for the hint. The biggest problem I see with authentication and I do not think mexuar could help me here (and I am definitely going to pay $2000 for it :-) But it is another story... Thank you! Ondrej Dean Collins wrote: Ondrej, You could do it using Mexuar Corraleta but this is a commercial application for Asterisk (US$2,000) http://www.mexuar.com/products_sdk.shtml http://www.mexuar.com/downloads/Press/CorraletaLaunchPR-1-branded.pdf However it has a whole heap more functionality than what you are looking for. If you just want to do 2 legged outbound calls check out 'call files' on www.voip-info.org Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ondrej Valousek Sent: Monday, 13 November 2006 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Desktop integration Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --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] Custom voicemail extension greeting
Making custom voicemail greetings seems fairly straight forward, and I've done it. However, I'm looking for a way to make the actual extension answer with You've reached my Jim Dandy voice mailbox, go take a flying . . .. (OK, so maybe not), instead of The person at extension , is unavailable Possible? Easy? Under my nose? joe a. ___ --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] operator console
oops sorry i didnt saw quoted text of other user and it showed as first post in gmail draft so i thought u made a topic for that pbx ( so considered spam :P ) . Sorry again :)On 13/11/06, Jordi Nelissen [EMAIL PROTECTED] wrote: Vicky,my other post related to a Web GUI for asterisk. This post is related toan Operator Console. I am simply answering the user's question, so Idon't see why you would consider this to be spam, and I never read you can not send two mails to the list on the same day. ___ --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: Desktop integration
I have been using http://www.snapanumber.com/ 's Windows tray utility, and it works great. -- -- Steven http://www.glimasoutheast.org Ondrej Valousek [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --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: FW: [asterisk-users] Desktop integration
On 15:34, Mon 13 Nov 06, bails wrote: We use Asterisk Desktop Manager http://adm.hamnett.org/ very successfully with both debian and windows desktops. Firefox can't find the server at adm.hamnett.org. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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: Monitor, MixMonitor and volume levels
On 11/10/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Steve Davies wrote: *bump* No suggestions at-all? Does anyone use this facility in a similar way and NOT have problems? Check the gain on your ISDN interface. The monitor command doesn't modify the volume by default. Have you tested calls via IAX to your cell? Leo Yes, it is strange - The gains are fine - users can hear the calls perfectly at both ends, fax works fine etc etc - Only the recording is odd. I have tested this with IAX, and three different ISDN interfaces (ISDN, Quad ISDN and Sangoma PRI) - All of them have the same symptom. SIP to SIP is the only case that seems to work, almost as if the SIP code is re-levelling the non-RTP stream. I think I need to try a Zap to Zap forwarded call to see how that works... Cheers, Steve ___ --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] Problem with internet down
I believe that the problem really is fault of DNS lookups, but as I should proceed for resolve that?? see the first point at http://www.voip-info.org/wiki/view/Asterisk+administration The best solution for now is probably to have a caching dns server on your Asterisk box or in your LAN ___ --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] problem with redirects
Dear all, My architecture is having some problems with redirects. In the following diagram is shown a simple erroneous test. When someone dials from the PSTN, signalling of the incoming call is passed to Asterisk which routes to SIP Express Route (Ser), and then Ser routes to the phone. The user has configured the phone to forward all calls to another PSTN number, and then, a 302 Moved Temporarily reply goes back to Ser which forwards back to Asterisk. Because Asterisk is configured with promiscredir=yes, it sends a reINVITE to the number announced in the 302 message as expected, and then that new INVITE goes back to Ser. Ser looks at the called number in that INVITE and because it is a PSTN number, sends the call back to Asterisk so this gateway can route it to PSTN. Because Asterisk receives the last INVITE with the same Call-ID that he passed to Ser in the anterior INVITE, he thinks it's a loop, and ends the communication with a 482 Loop Detected message. How can I configure Asterisk so that he can route the last INVITE to PSTN without giving me that error? PSTN Asterisk Ser UAC |INVITE| | | | -- | | | | 100 Trying | | | | --- | | | - | | INVITE| INVITE| | | -- | --- | | | 100 trying | 100 trying | | | --- | | | | 302 Moved Temporarily | 302 Moved Temporarily | | | -- | --- | | | ACK | ACK | | | --- | --- | - | | INVITE| | | | --- | | | | 100 trying | | | | --- | | | | INVITE| | | | --- | | | | 482 Loop Detected | | | | --- | | | | ACK | | | | --- | | Thanks in advance, Ricardo. ___ --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] Problem with internet down
A search of google should turn up some recommendations about running a local cacheing DNS proxy, or similar. I've never done it myself (the cacheing proxy, not the searching on google) so I don't know the specifics. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Luiz Martins Sent: 13 November 2006 15:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with internet down I believe that the problem really is fault of DNS lookups, but as I should proceed for resolve 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] Random 'no audio' problem
I've figured out the problem, alaredy, and posted in another thread, but no one seems to have an answer yet. It is a problem in the IAX trunk. If I turn the jitterbuffer on, I get one-way-audio when I put someone on hold. If I turn the jitterbuffer off... I still have two way audio. THis is running 1.2.x Anyone have any bright ideas? On 11/12/06, Jordi Nelissen [EMAIL PROTECTED] wrote: Matt, as a start, what I can advise you is to take a tethereal trace and try to reproduce the problem. nohup tethereal host a.b.c.d -s2000 -w /tmp/yourtrace.cap Where a.b.c.d is the IP address of your IP phone. You can then analyse the trace and at least see if the asterisk box is sending AND receiving RTP traffic to and from the phone. We have seen some issues in the past with 'no audio' or 'unidirectional audio' due to wrong firmware versions in SIP phones or due to ethernet switch instability, even on a cisco 3560 switches. Hope this helps, Jordi Matt wrote: I have no idea.. that sounds like your Internet connection is going down and leaving you for a bit and then coming back. My issue is a local network connection, no public Internet... or you can even call in from outside on the PSTN and the audio, both ways, will just stop. On 11/3/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I have the same problem with IAX trunk and SIP extensions. Now I think its the IAX. I never had this problem om SIP trunk. Am I right? ___ --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 -- Jordi Nelissen E S C A U X Business IP Telephony www.escaux.com -- Email from people at escaux.com does not usually represent official policy of ESCAUX. See http://www.escaux.com/disclaimer for details. ___ --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] Asterisk IVR functionality
On Mon, Nov 13, 2006 at 07:10:12AM -0500, Brian Rogan wrote: On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote: Hi i have an application developed with bayonne. Recentely i'm experiencing some problems and i am planning to migrate to asterisk. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) Asterisk uses a system called AGI to provide IVR. Not exactly true: IVR may be implemented using standard dialplan (extensions.conf). AGI is a way to let a totally external program control the flow of a call through Asterisk. It generally adds overhead and/or complexity to the system. It can be used for various operations, and not inherently related to IVR. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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: FW: [asterisk-users] Desktop integration
On Mon, 2006-11-13 at 17:00 +0100, Michiel van Baak wrote: On 15:34, Mon 13 Nov 06, bails wrote: We use Asterisk Desktop Manager http://adm.hamnett.org/ very successfully with both debian and windows desktops. Firefox can't find the server at adm.hamnett.org. Just downloaded it with Firefox, try again. -- Dave Cotton [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
Re: [asterisk-users] Re: Desktop integration
On 11/13/06, Steven [EMAIL PROTECTED] wrote: I have been using http://www.snapanumber.com/ 's Windows tray utility, and it works great. -- -- Steven http://www.glimasoutheast.org Ondrej Valousek [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --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 when you use snap, does the call go to your iax hardphone connected to asterisk or do you need a softphone on your PC? ___ --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] Recording outbound analog calls with X100P
List members, Is it possible to record outbound analog calls using an X100P? I was asked if I knew how to record all calls for a shop with 4 analog phones transparently to the end users. I thought Asterisk was a good fit for this and I envisioned using either Digium TDM400Ps or Sangoma A200s with 4 FXO and 4 FXS modules. The FXO modules would be connected to the existing PBX and the FXS modules to the existing analog phones. Then with a simple dialplan, all inbound and outbound calls could be recorded by Monitor. I wanted to mock this up using some X100Ps that I had laying around, but found that I could only record inbound calls. I believe that I need an FXS interface to record outbound analog calls but my past experience is with T1 interfaces, so I could be mistaken. If anyone could suggest any improvements to my recording scheme, they would also be appreciated. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] ChanSpy problems in Unicall to SIP calls
Hi: I'm having audio dropouts in ChanSpy when the call is originated in a Unicall (E1/MFCR2) channel and the destination is an Agent using a SIP phone. If the agent is using a traditional phone (going from the PBX to asterisk via another Unicall line) no dropouts are present. The dropouts seems to occur only in the audio coming from the Unicall line, and mostly when there is no audio coming from the SIP end. If both are talking, or only the SIP end, audio seems fine. I saw the problem in all asterisk 1.2 versions, but after the late ChanSpy rewriting, it seemed the problem was solved in an unloaded test system, but in the production one, the dropouts are still present. Any idea? Guillermo Freige _ De todo para la Mujer Latina http://latino.msn.com/mujer/ ___ --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] Custom voicemail extension greeting
Am Montag, den 13.11.2006, 10:37 -0500 schrieb joe a.:
Making custom voicemail greetings seems fairly straight forward, and I've
done it.
However, I'm looking for a way to make the actual extension answer with
You've reached my Jim Dandy voice mailbox, go take a flying . . .. (OK, so
maybe not), instead of The person at extension , is unavailable
Possible? Easy? Under my nose?
If you want to replace the default announcement, look in
the /usr/share/asterisk/sounds/ directory for files like vm-intro.gsm.
Alternatively you can record a file and place it as unavailable
greeting for all your voiceboxes instead of recording announcements for
each single voicebox by dropping a file unavail.wav
in /var/spool/asterisk/voicemail/${VOICEBOXCONTEXT}/${VOICEBOXNUMBER}/
for all possible combinations of VOICEBOXCONTENT and VOICEBOXNUMBER
BR
Anselm
___
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[asterisk-users] Slow playback of sound prompts
I have this very weird situation where some callers hear the playback of sound prompts on half speed. It only lasts a few second but it can happen at any time during playback. My server is a 3.4 Ghz Xeon with 1 GB RAM and 80 GB SATA disk. I run Asterisk 1.2.13 on FreeBSD 6.1 Anyone who has a clue to what can be the cause of this? Med vänliga hälsningar/Kind regards Roger Lewau Serverhallen i Norden ABBox 20087, 200 74 Malmö, SwedenTel: +46-40-6905000Fax: +46-40-6905001 Web: www.serverhallen.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] Custom voicemail extension greeting
joe a.[EMAIL PROTECTED] Wrote on: 11/13/2006 10:37 AM: Making custom voicemail greetings seems fairly straight forward, and I've done it. However, I'm looking for a way to make the actual extension answer with You've reached my Jim Dandy voice mailbox, go take a flying . . .. (OK, so maybe not), instead of The person at extension , is unavailable Possible? Yes Easy? Yes. Under my nose? Almost. joe a. joe a. ___ --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] 2 * servers Host=ip - doesn't work Host=dynamic with register is OK, why?
Hi all,I've 2 * servers with static IP, and i notice that if i set both sip peers with host=server_ip and qualify=yes it presents UNREACHABLE on asterisk CLI.When i changed the host parameter to host=dynamic and set the register string in the [general] of sip.conf on both servers, the connection has been reestablished.I must say the 2 servers are in the same LAN with static IP.What could be the problem?-- 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: Desktop integration
On Monday 13 November 2006 11:49, Yu Safin wrote: when you use snap, does the call go to your iax hardphone connected to asterisk or do you need a softphone on your PC? Please trim your posts, you don't need to keep the headers and signature lines of the entire thread to ask one sentence, now do you? I've been using SNAP for quite some time now, and I need to get it registered on our 30 office machines. To answer your question: Yes. SNAP can be configured any way you like. If you use a softphone, you can have it connect the call to your softphone. Or your hardphone. Or your cell phone. However you like. It's all in the configuration and how you tell it to do things. -A. ___ --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 Call Statistics
Hi Moises, Coul you give more details about how to use Cacti for CDR analysis, there is some special pluggin, additional conf? Your help will be appreciated. Rgds. On 10/31/06, Moises Silva [EMAIL PROTECTED] wrote: of course you can always use http://cacti.net/download_cacti.php On 10/31/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: Check out voip-info.org, there are quite a few GUIS some even generate nice graphs! On 10/31/06, omar parihuana [EMAIL PROTECTED] wrote: Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --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 -- 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 -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --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] Modprobe Zaptel
Hi Eric, Your answer solved my problem. I did a uname -r which= 1.6.12-27mdksmp, the Makefile had -27mdkblahblahblah. Once I switchedthe makefileeverything worked. Thanks for your help. Julian Date: Fri, 10 Nov 2006 12:05:37 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Modprobe Zaptel Doug Lytle wrote: Tzafrir Cohen wrote: On Thu, Nov 09, 2006 at 05:25:02PM -0600, Eric ManxPower Wieling wrote: Is such horror normally needed with Mandrake? Doesn't Mandrake provide working kernel headers linked from /lib/modules/`uname -r`/build ? I have no issues with Mandriva 2006 or 2007 on compiling Zaptel, nor have I ever had to modify the Makefile. 2006 and 2007 MAY have changed that. I needed it on 8.1 and 9.2 ___ --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 multiple languages
Hello, I try to configure Asterisk to send voicemail in the language of the user's mailbox. But the only way I see is to modify the app_voicemail.c anybody has an alternative idea for me ? Lot of 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: FW: [asterisk-users] Desktop integration
On 17:27, Mon 13 Nov 06, Dave Cotton wrote: On Mon, 2006-11-13 at 17:00 +0100, Michiel van Baak wrote: On 15:34, Mon 13 Nov 06, bails wrote: We use Asterisk Desktop Manager http://adm.hamnett.org/ very successfully with both debian and windows desktops. Firefox can't find the server at adm.hamnett.org. Just downloaded it with Firefox, try again. At home it's working. Must be something in our work dns setup. Thnx. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] newbie question
Hello,I know little to nothing about asterisk. I am currently reading info online but was looking for other links as to good places to start. I basically just need to use asterisk to create a prototype. I need to demo a system that will allow the user to call a number, enter in some data, the backend then queries our db and returns results to the user via the phone. I've been told this can be done with asterisk so need to get started. This may never turn into an actual project so just need the minimal amount of work now to get it working. Any way to use a softphone or whatever to call, have the PBX prompt for info, receive it and then query the db and read the results to the user is what I want to do. The back end code will probably be PHP but can be in something else if needed. I am currently looking here;http://www.voip-info.org/tiki-index.php?page=Asterisk+AGIAre there other places to start? Is there a place to get an asterisk box/number set up for testing? 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] problem with redirects
OK, to simplify the reading I'll resume my problem... Is there a way to make Asterisk send a call to Ser witch reroutes it back to the same asterisk server ,without resulting in a loop detected error in Asterisk? Thanks, Ricardo. ___ --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] Fast Busy with autodial using a call file
Setup overview:
We have an asterisk server serving a small number of SIP phones. The
asterisk server is connected to an old phone system via a T1. The
asterisk server is also connected to a second T1 used for
inbound/outbound calls.
Scenario:
We are using a call file to do auto-dialing. A call is made to a phone
on the old phone system. This call always works without a problem.
When that call is picked up asterisk then calls a second number (either
a SIP phone, another number on the old phone system, or an external
number) and connects the 2 calls. The second call is always dialed with
a 91NXXNXX so that it goes out over the T1, even if it is destined
for an internal SIP extension or an extension on the old phone system.
If it is destined for 1 of those 2 locations it is wrapped back through
the T1 and comes in as an incoming call.
If the second call is to either an extension on the old phone system or
an external number then it works fine. If it is destined for one of the
SIP extensions it fails with a fast busy about 95% of the time. The 5%
of the time when it does work seems to be when I have not used a call
file to call that extension in a long time (say 10 or 15 minutes). So
it seems that something may not be getting closed correctly. Calling
manually to the same number works 100% of the time.
Any ideas why we might be getting a fast busy when using a call file?
Related configs:
The call file looks like this (with the extensions altered):
Channel: Zap/g2/6070
RetryTime: 60
WaitTime: 10
Context: internal
Extension: 913125551212
Priority: 1
extensions.conf:
TRUNK=Zap/g3; Trunk interface
...
[internal]
...
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:1})
exten = _91NXXNXX,n,Congestion()
zapata.conf:
context=internal
switchtype=national
signalling=pri_cpe
group=3
channel= 6-27
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[asterisk-users] Music on hold question
Hi all, Using the latest 1.4 of Asterisk. I have noticed that the music on hold files are in wav, isn't mp3 supported anymore? Many thanks, Christian ___ --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] DSl and more then 1 call
Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls going at the same time the ping times go up to 2500 and above and the call quality is horrible. If I add a third call the system becomes unusable. But if you hang up all calls except 1 (it doesn't matter which one) it works fine again. Any help you could provide would be greatly appreciated. Kelly ___ --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] Mysql 6 second rounding
This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud like 30 seconds ) ..Suppose if billsec field is 3 seconds then it should round to 6 seconds , if its 13 second then it should round up to 18 seconds ( for 6sec pulse counting ) . What would be mysql function to do this ? ___ --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 as a Media Gateway
I am planning to use asterisk with Digium TDM2404E card as a media gateway to terminate traffic to Cell phones. Anyone got this working before with no problmes, specially with Answer/Disconnect supervision? 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] Problem with internet down
Why not directly use ip address in host= lineinextensions instead of dynamic address like sip.voipprovider.com .. temporary fix but it may work . On 13/11/06, Steve Langstaff [EMAIL PROTECTED] wrote: A search of google should turn up some recommendations about running alocal cacheing DNS proxy, or similar.I've never done it myself (the cacheing proxy, not the searching ongoogle) so I don't know the specifics. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Andre Luiz Martins Sent: 13 November 2006 15:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with internet down I believe that the problem really is fault of DNS lookups, but as I should proceed for resolve that??___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Can AGI do this?
Please pardon the absolute noob questions. Someone has asked me to interface with Asterisk and have it dial 4 numbers in succession to have it track down an on-call person. My initial reaction was to write an AGI program and return all 4 numbers and have Asterisk hunt them - can Asterisk do this? If not, is it possible to write an AGI program that gets all 4 numbers, then somehow hands them one-by-one to Asterisk? If so, how does Asterisk manage the communication of failed to complete the call with the AGI app? Does the AGI just monitor stdin looking for status messages and returns the next number? If Asterisk/AGI can do both, is the first method better than the second? It certainly seems easier. Thanks in advance! Rgds, Bret ___ --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 forward behind a nat
IF your asterisk server is behind NAT and no port forwarding is done then how can that static ip user/device reach it .Youwillhavetokeepasteriskserverinstaticipordoportforwardingtoacceptconnectionsfromoutside. ORmaybeididntunderstandsenarioproperlyhere.Isitlikeyour Server with SIP application (public_address) responds to sip calls made by any program ( like sjphone pc-pc sip ) . If thats case then asterisk should be able to call it like any other program or maybe theres nat scenario playing bad here :-/ . Can you port forwardfromfirewalltoasteriskserver ?? On 13/11/06, nik600 [EMAIL PROTECTED] wrote: On 11/12/06, nik600 [EMAIL PROTECTED] wrote: On 11/12/06, Vicky [EMAIL PROTECTED] wrote: Yep make the server with dynamic ip register to server with static ip ( sip or iax both will do but in sip keep nat=yes while making extension ) the problem is that the server with dynamic ip can't register on the other server! This is the situation: Server with SIP application (public_address) | | - - - Internet | | Firewall (NAT) | | Server Asterisk (private ip: 192.168.100.249/public ip:public_address_2) | Analogic Board | Telecom I want to make a call from Server Asterisk to the server with SIP Application. The SIP Application can't register to Server Asterisk (because the application can't do it, i know, it isn't a good thingbut this is the application) When The SIP Application receives a SIP call it responds (because a dummy SIP user is autoregistered on hisself) So i only have to make a call to SIP/[EMAIL PROTECTED] I've also tried to setup an asterisk server on my laptop, and make a call to SIP/[EMAIL PROTECTED] from the public_address network. It works! I only have to setup the Asterisk server in production to make a SIP call throw the NAT but without any SIP user registered on it. Can i do that? Many thanks to all maybe you need some other information?___--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] DSl and more then 1 call
Does it happen when you make more than one call from you main voip server alone ? Or it happens when there are more than 1 call on your branch server ? Pin the problem is in which server first , If main server can handle 2-3 calls with no lag then its probably problem in branch server . On 13/11/06, Kelly Opal [EMAIL PROTECTED] wrote: Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls going at the same time the ping times go up to 2500 and above and the call quality is horrible. If I add a third call the system becomes unusable. But if you hang up all calls except 1 (it doesn't matter which one) it works fine again. Any help you could provide would be greatly appreciated.Kelly___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Mysql 6 second rounding
sum(duration+(6-mod(duration,6) for summary of seconds divisible by 6, /60 for minutes On Tue, 2006-11-14 at 00:07 +0530, Vicky wrote: This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud like 30 seconds ) ..Suppose if billsec field is 3 seconds then it should round to 6 seconds , if its 13 second then it should round up to 18 seconds ( for 6sec pulse counting ) . What would be mysql function to do this ? ___ --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] Modprobe Zaptel
Sorry meant 2.6.12-27.. From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSubject: RE: [asterisk-users] Modprobe ZaptelDate: Mon, 13 Nov 2006 17:31:24 + Hi Eric,Your answer solved my problem. I did a uname -r which= 1.6.12-27mdksmp, the Makefile had -27mdkblahblahblah. Once I switchedthe makefileeverything worked. Thanks for your help.Julian Date: Fri, 10 Nov 2006 12:05:37 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Modprobe Zaptel Doug Lytle wrote: Tzafrir Cohen wrote: On Thu, Nov 09, 2006 at 05:25:02PM -0600, Eric ManxPower Wieling wrote: Is such horror normally needed with Mandrake? Doesn't Mandrake provide working kernel headers linked from /lib/modules/`uname -r`/build ? I have no issues with Mandriva 2006 or 2007 on compiling Zaptel, nor have I ever had to modify the Makefile. 2006 and 2007 MAY have changed that. I needed it on 8.1 and 9.2 ___ --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 with ss7 and sip-t
Hello All, as good?It would like to know if somebody has experience in asterisk with ss7 protocol for isdn and asterisk with support to the protocol sip-t. Best RegardsJosué ___ --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] FAX using T38
Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in Asterisk. Although I can't send FAXes to PSTN using T38. The problem, as far as I know, might be assigned with the Content-Length shown in the message header of every SIP message negotiating T38 parameters. I've observed that after leaving Asterisk, the Content-Length of every message carrying T38 parameters gets shorter than truly is, and contrarily to my ATAs that seem to don't care about this, my Telco analyses the packet length written in this messages and truncates them, aborting the call. Does anyone experienced this too? Any ideas besides editing the chan_sip.c code to fix this problem? Thanks, Ricardo. ___ --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 argument size limit
I'm trying to implement a voicemail distribution list using asterisk and I've hit a bump. I've got an agi script that parses voicemail.conf and generates a list of voicemail boxes to use as an argument to the voicemail() function. The problem is that the argument exceeds 256 characters (100 mailboxes * 5 characters per mailbox [ + 4 digit mailbox] = 500+ characters) and asterisk seems to trunk there resulting in only the first 50 people getting the voicemail. I've searched for a solution on the lists and through google to no avail. Can this be fixed with a simple buffer size change in a couple of places or am I going to run into dependencies everywhere? Going through the code it looks like char[256]'s in a number of places that could be causing the problem but I haven't had the time to go through the code in any depth. This is asterisk 1.2.12.1 although I am going to be upgrading to the latest version shortly. If there is a limit then is this documented anywhere? Should a function that gets an argument of 500 characters simply truncate or should it simply return an error that the input is too long so as to avoid operating on partial data? I've already considered a number of workarounds including copying the messages right from the file system but the only way to get things like email notification is to use the function. Is there a copy function that I could use in a loop that would also notify anyone with an email notifcation set up? Any help or guidance would be appreciated. Thanks in advance, -Don ___ --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] Mysql 6 second rounding
Most usage charges are stored in various billing databases as per MINUTE of use, not per 6 seconds of use. 6 second billing simply means that you bill in decimal fractions of a minute, 66 seconds becomes 1.1 minutes. 1. Divide your billsec value by 60 and round to 1 decimal place. Add 0.5 to the result and then round if you want to round UP, that way 61 seconds is still 1.1 minutes. 2. Multiply the result times your per minute of use charge. The reason 6 second billing is often used is because it translates easily into decimal minutes J Billminutes=round((billsec/60)+0.5),1) Charge = round(billminutes*minutecharge,2) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vicky Sent: Monday, November 13, 2006 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mysql 6 second rounding This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud like 30 seconds ) ..Suppose if billsec field is 3 seconds then it should round to 6 seconds , if its 13 second then it should round up to 18 seconds ( for 6sec pulse counting ) . What would be mysql function to do this ? ___ --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] Survey: In what ways do you use Asterisk at your house?
All, I'm starting to tinker with Asterisk for use in my home. Here's my current setup: Cox broadband telephone -- spa3k-fxo analog phones + answering machine (all on one line) -- spa3k-fxs I can pick up a phone in my house, dial a certain extension, and the spa3k will connect me to Asterisk, which currently plays a message and hangs up (not particularly useful). If I dial any other number, the spa3k dials that number out on the fxo. The main thing I'm trying to do right now is replace my answering machine with *-based voicemail. I want to retain the ability to screen calls (listen on a speaker while a person is leaving a message), but I'm not sure of the best way to go about this. Recommendations are welcome. Note that my * box and answering machine are on two different floors in my house, so running a speaker in the kitchen (answering machine location) from the sound card on the * box is doable, but not desirable. Also of note: I only have basic no-frills phone service (no caller id, no call waiting, etc), though I am open to adding options if there's a good reason. The main reason for this e-mail is to see what other people are doing. - What service provider/technology do you use for origination/termination? - What hardware/software do you use and how does it all tie together? - What tasks do you use * to accomplish? - Any other pertinent info. I'm trying to find practical uses for *, but I'd like to throw in some fun/pointless stuff as well. Thanks for your time. Earle ___ --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] Mysql 6 second rounding
Thx and what would the sql query be ?.Iplantoputadditionalfieldas6second.Howcanimakebillsecofvaluesofwholetablegetroundedandfilledinfield6second Sorry i am a noob with mysql :D On 14/11/06, James Coberly [EMAIL PROTECTED] wrote: sum(duration+(6-mod(duration,6) for summary of seconds divisible by 6, /60 for minutes On Tue, 2006-11-14 at 00:07 +0530, Vicky wrote: This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud like 30 seconds ) ..Suppose if billsec field is 3 seconds then it should round to 6 seconds , if its 13 second then it should round up to 18 seconds ( for 6sec pulse counting ) . What would be mysql function to do this ? ___ --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 listTo 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] DSl and more then 1 call
Hi It's defiantly the branch server. My main server handles 30 to 40 calls at a time on a regular basis. It is only happening on the branch server and it acts like it is using up all the bandwidth of the DSL. It is a 1.5 meg down and 512 up DSL line. I would think it could handle 2 simultaneous calls. I have tried using g729, ulaw, alaw and gsm. There is no difference in the behavior. Could it possible be a routing issue on the LAN side of server 2. Kelly - Original Message - From: Vicky To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 13, 2006 1:59 PM Subject: Re: [asterisk-users] DSl and more then 1 call Does it happen when you make more than one call from you main voip server alone ? Or it happens when there are more than 1 call on your branch server ? Pin the problem is in which server first , If main server can handle 2-3 calls with no lag then its probably problem in branch server . On 13/11/06, Kelly Opal [EMAIL PROTECTED] wrote: Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls going at the same time the ping times go up to 2500 and above and the call quality is horrible. If I add a third call the system becomes unusable. But if you hang up all calls except 1 (it doesn't matter which one) it works fine again. Any help you could provide would be greatly appreciated. Kelly___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Native TDM Bridge
I have a two port TE205P Digium card. I have set everything up to
create a native zap bridge between the two spans. Everything works
perfectly except one thing. Our telco has a password that has to be
entered as soon as a long distance call is made. So if I dial a long
distance call from my meridian system, asterisk bridges the call
between two channels, my telco picks up and gives me a tone, I enter
my 3 digit password, and the call is supposed to be completed.
Instead I get a busy signal. The call is already bridged by the time
I have to punch in the telco password. This works fine If I plug my
norstar system directly into the PRI telco. Another strange issue is,
If I make a slip of the finger and I dial 1-349-555-1 with a
trailing digit all works fine.
Anyone have an idea I tried relaxing the DTMF on both PRI's as
well. Debug does't show anything either.
SPAN1 is connected to my Telco's PRI and SPAN2 is connected to a
Norstar Meridian.
Here is my zapata.conf:
[channels]
#PRI to TimeWarner defined as group 2
switchtype = national
signalling = pri_cpe
context=pstn
callerid=asreceived
resetinterval = never
group = 2
channel = 1-23
#PRI to Norstar Meridian defined as group 3
switchtype = national
signalling = pri_net
context=meridian
callerid=asreceived
resetinterval = never
group = 3
channel = 25-47
Here is my zaptel.conf:
#PRI to TimeWarner
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
#PRI to Norstar Meridian
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
loadzone=us
defaultzone=us
Here is my extensions.conf:
;
; Context for meridian incoming calls
; Check incoming call to see if callerid is already set. If not
; then set it to the Main number, and forward it out to the
; highest available channel on the TW PRI.
[meridian]
exten = _X.,1,GoToIf($[${CALLERIDNUM} = ]?2:3)
exten = _X.,2,Set(CALLERID(num)=EXCLUDEDFORLIST)
exten = _X.,3,NoOp(${CALLERIDNUM})
exten = _X.,4,Dial(${PSTNOUT}/${EXTEN})
exten = _X.,5,Hangup()
exten = i,1,Answer()
exten = i,n,Wait(1)
exten = i,n,Playback(cannot-complete-as-dialed)
exten = i,n,Playback(please-contact-tech-supt)
exten = i,n,Hangup()
;
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Re: [asterisk-users] Voicemail argument size limit
Donald Stahl wrote: I'm trying to implement a voicemail distribution list using asterisk and I've hit a bump. I've got an agi script that parses voicemail.conf and generates a list of voicemail boxes to use as an argument to the voicemail() function. Generate in groups of 50 and loop it until you have them all? 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] Mysql 6 second rounding
Supposing you have an extra column called 6second: UPDATE cdr SET 6second=billsec+(6-mod(billsec,6) where 6second=0 if you want a decimal minutes column called billmin UPDATE cdr SET billmin=round((billsec/60)+0.5),1) where billmin=0 Vicky wrote: Thx and what would the sql query be ? . I plan to put additional field as 6second . How can i make billsec of values of whole table get rounded and filled in field "6second" Sorry i am a noob with mysql :D On 14/11/06, James Coberly [EMAIL PROTECTED] wrote: sum(duration+(6-mod(duration,6) for summary of seconds divisible by 6, /60 for minutes ___ --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] MWI not working in 1.4
Before I open a bug I'll ask again if anyone else is having trouble with receiving MWI on SIP devices in 1.4. My configuration was working fine in 1.2 but as soon as I change to any build of 1.4 I don't get notification on any of several SIP devices. I can post my configuration but since it was working I can only assume it would break if something in voicemail.conf has changed or sip.conf but current examples appear to concur with my setup. From my peer definition: [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED] subscribemwi=yes voicemail.conf [mainmenu] 100 = 1234,User 1,[EMAIL PROTECTED],,saycid=no|envelope=no|review=yes|tz=eastern 200 = 1234,User 2,[EMAIL PROTECTED],,saycid=yes|envelope=yes|review=yes|tz=eastern 300 = 1234,User 3,[EMAIL PROTECTED],,saycid=yes|envelope=yes|review=yes|tz=eastern ___ --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] Dial/Continue/Announce
I initiate a call with a callfile, specifying the From phone# as the channel Dial(), and the To phone# as the Extension Dial(). I announce the To phone# to the From listener with the A() option to the Dial() command. It seems that the A() app plays audio while blocking return from the From Dial(), so the From person has to wait more time while listening to the announcement before the To Dial() can even begin, which then must start ringing. How can I connect to the From person, then immediately connect to the To person, while playing an announcement to the From person? -- (C) Matthew Rubenstein ___ --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 argument size limit
I'm trying to implement a voicemail distribution list using asterisk and I've hit a bump. I've got an agi script that parses voicemail.conf and generates a list of voicemail boxes to use as an argument to the voicemail() function. Generate in groups of 50 and loop it until you have them all? Define Generate. Right now I Answer and then send them into the voicemail function with the list of mailboxes. How would I go about first recording the message and _then_ sending it to voicemail in a loop, 50 at a time? If there is a function I missed or am unaware of please let me know. Recording the message multiple times is not an option :) Thanks, -Don ___ --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] SIP Ports (1000 to 2000 works)
I was reading the posts and someone said about the default 1000 to 2000 I see in the .conf the default is 1 to 2 I found a service that gives inbound DID's in the firewall 5060 and 1 - 2 is setup no workie on the DID But when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000 Now the DID works fine. So you me what the standard is -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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] Mysql 6 second rounding
Hey thanx for that Marnus . Thatsworking just exactly how i wanted :).Damoniactuallycameupwithsamerow/60+0.5thenrounduptrickwheniwasdoingsomethingsameinexcelsheets:)anditsusefulforbillingin1minuteroundup(60secpulse)butifailedtogetitworkingfor6secondpulse. Marnus'ssqlqueryisperfect..nowsupposechargeis1.5cent/minthenicanused6secondroundup'dvalueandmultipleby0.15togetcallbillincentand it can also be used for 30 second pulse or any other valuewithsmallmodification ..Thx. On 14/11/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: Supposing you have an extra column called 6second: UPDATE cdr SET 6second=billsec+(6-mod(billsec,6) where 6second=0 if you want a decimal minutes column called billmin UPDATE cdr SET billmin=round((billsec/60)+0.5),1) where billmin=0 Vicky wrote: Thx and what would the sql query be ?.Iplantoputadditionalfieldas6second. Howcanimakebillsecofvaluesofwholetablegetroundedandfilledinfield6second Sorry i am a noob with mysql :D On 14/11/06, James Coberly [EMAIL PROTECTED] wrote: sum(duration+(6-mod(duration,6) for summary of seconds divisible by 6, /60 for minutes___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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 argument size limit
Donald Stahl wrote: Define Generate. Right now I Answer and then send them into the voicemail function with the list of mailboxes. How would I go about first recording the message and _then_ sending it to voicemail in a loop, 50 at a time? If there is a function I missed or am unaware of please let me know. I was not clear on how your AGI functioned. 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] SIP Ports (1000 to 2000 works)
actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: I was reading the posts and someone said about the default 1000 to 2000I see in the .conf the default is 1 to 2I found a service that gives inbound DID's in the firewall 5060 and1 - 2 is setup no workie on the DIDBut when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000Now the DID works fine.So you me what the standard is--Best regards,Al BochterBochter Services http://www.BochterServices.com/?t=EmailAre you outside of the US?Do you need to call US Toll Free Numbers?We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED](Cellular) 1-712-432-5401(Voip PBX) Free World DialUp: 780-217 EXT: 250WebSite: http://www.freeworlddialup.com/BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=emailFor new and used security items http://www.bochterservices.com/?j=storet=email_securityGOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email___--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 argument size limit
Right now I Answer and then send them into the voicemail function with the list of mailboxes. How would I go about first recording the message and _then_ sending it to voicemail in a loop, 50 at a time? If there is a function I missed or am unaware of please let me know. I was not clear on how your AGI functioned. I'm happy to alter my AGI file in any way that will allow this to function. If you can suggest a means of getting this to work please let me know. Alternatively I may ask or in -dev if they know what would need to be changed for a recompile. Thanks, -Don ___ --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 argument size limit
Donald Stahl wrote: Right now I Answer and then send them into the voicemail function with the list of mailboxes. How would I go about first recording the message and _then_ sending it to voicemail in a loop, 50 at a time? If there is a function I missed or am unaware of please let me know. I was not clear on how your AGI functioned. I'm happy to alter my AGI file in any way that will allow this to function. If you can suggest a means of getting this to work please let me know. I would suggest that you follow your original thought of using the file system. You can turn on the phone's MWI by touching the msg.txt file 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
