Re: [Asterisk-Users] Some questions of heavy * deployment and stability.
quote who=Anton Tinchev How stable are these channel drivers? I haven't run into any real stability issues, then again I just have 4 grandstreams sitting on a desk. :) Oh, and a CAC AB1 channel bank connected to a Digium T100P. Is there any commercial support for faster bugcleaning, fixing ... (anything will be in the GPL field)? http://www.digium.com/index.php?menu=software_support Look at the second support option... Is there any way for more stability of SIP channel drivers? Will be some support (including bugfixes/stability issues) if we buy a lot of digium cards (let say 20+ TE410 for 6 months). I'm asking this, becouse i have on my horizons deployment of telephone system for a small city (2000-5000 users for 2 years) shortly after New Year. That sounds really cool. I am sure the whole list would like to hear how it goes. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch statement taking over my local dialplan
It should always check your local plan first. Are you sure it's looking in default? Does it work if you do not include = provider? Mark On Fri, 17 Oct 2003, jerk face wrote: I have two Asterisk servers, one of which uses a switch statement (Server 2). On Server 2, the dialplan is as follows: [provider] switch... [default] include=provider exten=451,1,Dial,Zap/1 ... (No extensions defined for Server 2 are can_match (eg. exten=_9XX...)) The problem is that when I pick up a phone and dial 451, it searches Server 1 before using the extension defined in the default context. Is there a way to set Asterisk up to search the local dialplan before checking the switched server? - Thank you for your time. __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH and VAD
No, you can't easily figure out the remote clock. Look at the timerfd stuff if you want to see how to do it with zap timing anyway. Mark On 17 Oct 2003, Juan J. Sierralta P. wrote: On Fri, 2003-10-17 at 08:06, Mark Spencer wrote: But my second questions arises, why VAD affects MOH since VAD is on the ATA but MOH on Asterisk should transmit anyway. In order to keep the timing clean, Asterisk transmits precisely the same number of bytes of audio when running a sound generator (MOH, GenTones, etc), thus synchronizing the production of sound to the clock on the remote system. If we generate the audio ourselves independently of the far end (e.g. off a zap timer) then there will be slips relative to the far end. It would certainly be possible to add such a feature, but I'm not sure it's a good idea. Thanks Mark ! Where is code that control that ? I´ll like to play with it a bit, I saw the warnings about not supporting RFC3389 it is posible to use CN or its RTP timestamp to get the remote clock ? TIA -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch statement taking over my local dialplan
Now it *is* notworthy that even if he finds your 451 it may still check the other server to see if there is anything else beginning with 451 since that could allow a matchmore. Mark On Fri, 17 Oct 2003, jerk face wrote: I have tried all of that. I have found that the order of includes don't matter at all. Regardless of where they are placed, I have to wait for Asterisk to check the other server for the extension before dialing the local one. --- Florian Overkamp [EMAIL PROTECTED] wrote: Hi, At 06:50 17-10-2003 -0700, you wrote: [provider] switch... [default] include=provider exten=451,1,Dial,Zap/1 ... The problem is that when I pick up a phone and dial 451, it searches Server 1 before using the extension defined in the default context. Is there a way to set Asterisk up to search the local dialplan before checking the switched server? Have you tried [default] exten=451,1,Dial,Zap/1 include=provider ?? Or maybe even: [extensions] exten=451,1,Dial,Zap/1 [default] include=extensions include=provider With the includes order seems to matter :) Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback request: AGI GET DATA change termination digits
Hi, this is my 1.st response to this list, i hope this will work. I tend to agree with Steven since just allowing other termination digits probaly wont solve your upcoming the issues anyway. I use a wrapper around the 'get digit' which allows me to specify that the * digit repeats the menu but maxium 3 times and if the * star digit is used twice in sequence (without other digits inbetween ) then it means 'go to the menulevel above current level'. This was todays 'noise' from me. Freddi Subject: Re: [Asterisk-Users] Feedback request: AGI GET DATA change - termination digits From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Sat, 18 Oct 2003 19:14:14 -0500 Reply-To: [EMAIL PROTECTED] While that change is fine, you could also just write the same functionality with get digit and deal with it inside the AGI app. On Sat, 2003-10-18 at 16:50, Paul Crick wrote: ** REPOST: A week later and no feedback - am I the only one ** who'd find this functionality useful? No other AGI stuff ** out there needing something similar? I'd like some feedback on potentially submitting a request (and probably a patch too) to change the way the AGI command GET DATA works. Right now, # terminates the entry, which is then returned with the # stripped off the end. What I'd like is to allow user configurable termination digits, which are not stripped off the end. Reasoning: Some entries you'd like to terminate with #. Right now it's fine, you can tell if # was pressed or not by looking for the lack of a (timeout) entry in the returned result. You may want to allow * to cancel an entry. This is not possible right now. Systems I've coded previously allow # to terminate and complete a digit entry, * to correct an incorrect entry (playing the prompt again and restarting digit collection). Pressing * with no prior digit entry cancels the step and returns to the previous menu. I guess there's a compatibility issue with stuff that's out there already but if it was an optional 4th parameter this would be backwards compatible. Proposed new syntax: GET DATA filename timeout maxdigits terminator If terminator is specified (and it may be multicharacter, like *# to give me the functionality above), return the digit string collected so far, including the terminating digit. The calling app can strip the trailing character if needed. Thoughts? Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Start
Thanks all for the replies. I now * starting when the machine reboots without any user intervention required. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: 19 October 2003 03:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto Start I now have only ./safe_asterisk in my rc.local file. Another way to start/monitor * is to put this in /etc/inittab: # Run asterisk in runlevels 2-5 A1:2345:respawn:/usr/sbin/safe_asterisk If * dies for any reason, its automatically restarted within about five minutes. (It also causes messages such as: -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected to be reported on the CLI and log files, cluttering both. Doesn't seem to impact actual connections, etc, though.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Project Completed [Files Attached]
Thnx for the interest in the ivr sample [btw: I am not an expert in PERL/AGI :) ]comments are welocome. You can download the demo files and sounds from http://www.consulttech.com.pk/asterisk/IVR.rar There is a flowchart in the excel format thats shows how it works. you will need to place the sound files in the /backup/en directory, or u can change the code as well. I will be putting another application as soon as it will be prepared. Azher Do you Yahoo!? The New Yahoo! Shopping - with improved product search
[Asterisk-Users] X100P and Call Waiting Caller ID on the PSTN line
Hi, I have the following basic configuration: - one X100P card - two IP Phones (connected to an ATA186) When an incomming call on X100P, both phones connected to an ATA186 rings (configured like that in extensions.conf). When a call is established between the two ATA phones and a call occurs from the PSTN line, the callerid is correctly displayed on both phones (as internal callwaiting callerid). I have the callwaiting functionality activated on the PSTN line. Let's say now that a call is established between PSTN and one of the phones connected to the ATA. When another incomming call on the same PSTN line (Callwaiting) the number of the new call is not displayed on the phone display (callwaiting callerid). It can be done something to solve this issue? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 5 safe to use?
I am using 5.03 image on 7940 and 7960 and it is ok - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 1:03 PM Subject: RE: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 5 safe to use? On 18/10/2003 at 18:22 Juan J. Sierralta P. wrote: On Sat, 2003-10-18 at 17:25, Paul Mahler wrote: Howdy, Does anyone know if there are any problems running Asterisk when using later 7960 SIP versions like 04.04 or 05.03? I have 4.4 running without problems. I have 05.03 running with no problems... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr
[Asterisk-Users] The Start extension
I have my sip phones going into the context [from-sip] and would like to play an introduction message and then have the caller enter the extension. The message (dir-info was picked just to have something) doesn't play. Maybe I misunderstood the s extension. According to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The following is in extensions.com: [from-sip] exten= s,1,Answer exten= s,2,Background,dir-intro exten= s,3,DigitTimeout,3 exten= s,4,ResponseTimeout,10 exten = 2000,1,Dial(SIP/2000,20) exten = 2001,1,Dial(SIP/2000,21) Any ideas are appreciated. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk/Freebsd network connections
For those of you running * on FreeBSD: I compiled everything and can start. Sockstat -l shows that Asterisk listens on the correct interfaces and ports. Sniffing, I see registrations coming in to SIP debug, but nothing seems so reach Asterisk except IAX registration from a peer. Can't dial anywhere, can't reach the Asterisk with asterisk -r. Anyone recognizing this problem? * Recent Asterisk CVS with FreeBSD patch from bugs.digium.com * FreeBSD 4.8. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA Call Waiting
I have a CO line hooked to an X100P. I also have call waiting on the POTS line. How do you answer a call waiting call on a Cisco ATA? I get the Caller ID, when I hookflash I get dial tone, how do I hookflash to get CO line to connect to the caller?
[Asterisk-Users] Flastman 0.0.1-pre-alpha
Hi. My first 'snapshot' of flastman is out. Flastman stands for FLash ASTerisk MANager. written in flash, this first version is just a proof of concept, ie doesn't nothing except for logging in/out displaying manager events while logged in. But is realtime in any flash-enabled browser. Not very useful yet, but I'm going to improve it. For the hardcore testers, grab it from here: http://asterisk.espia-net.net/flastman-0.0.1-pre-alpha.tar.gz Please send me feedback P.S. also an alternative manager.c is shipped, since flash expect a message terminated by '\0' . this modified version of manager.c does that. updated to today cvs ;) . Look into INSTALL for more details -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA Call Waiting
Kevin wrote: I have a CO line hooked to an X100P. I also have call waiting on the POTS line. How do you answer a call waiting call on a Cisco ATA? I get the Caller ID, when I hookflash I get dial tone, how do I hookflash to get CO line to connect to the caller? if you have a phone connected directly to the line how do you change from call 1 to call 2?? If you do somthing like press #, then that same thing should work to change calls on the X100P.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Project Completed [Files Attached]
I have not found the type of license you are using for this demo. Can you please confirm which one you plan to use for this. On Sun, Oct 19, 2003 at 05:16:29AM -0700, Azher Amin wrote: Thnx for the interest in the ivr sample [btw: I am not an expert in PERL/AGI :) ] comments are welocome. You can download the demo files and sounds from http://www.consulttech.com.pk/asterisk/IVR.rar There is a flowchart in the excel format thats shows how it works. you will need to place the sound files in the /backup/en directory, or u can change the code as well. I will be putting another application as soon as it will be prepared. Azher - Do you Yahoo!? The New Yahoo! Shopping - with improved product search ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA Call Waiting
Hi, - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 8:55 PM Subject: Re: [Asterisk-Users] Cisco ATA Call Waiting Kevin wrote: I have a CO line hooked to an X100P. I also have call waiting on the POTS line. How do you answer a call waiting call on a Cisco ATA? I get the Caller ID, when I hookflash I get dial tone, how do I hookflash to get CO line to connect to the caller? if you have a phone connected directly to the line how do you change from call 1 to call 2?? If you do somthing like press #, then that same thing should work to change calls on the X100P.. But '#' is used for transfer with * then. what to do? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA Call Waiting
Dan wrote: Hi, - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 8:55 PM Subject: Re: [Asterisk-Users] Cisco ATA Call Waiting Kevin wrote: I have a CO line hooked to an X100P. I also have call waiting on the POTS line. How do you answer a call waiting call on a Cisco ATA? I get the Caller ID, when I hookflash I get dial tone, how do I hookflash to get CO line to connect to the caller? if you have a phone connected directly to the line how do you change from call 1 to call 2?? If you do somthing like press #, then that same thing should work to change calls on the X100P.. But '#' is used for transfer with * then. what to do? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Remove the t and/or T options from your dial string.. If your phone has a flash button you should be able to transfer with that (I think).. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Patch testers needed
Hello - As I've mentioned before, we have a large number of patches starting to build up in the bugtracker interface waiting for addition to CVS. Many of these patches are ready to go and have been fully tested but have not been added due to time schedules. However, almost all of the bugnotes that have the term [patch] in the title have code in them that has been submitted but does not have any verifiable testing done on it other than what the author has done. WE NEED TESTERS TO CONFIRM PATCH VALIDITY. The folks at Digium (mark, specifically) don't have time to do exhaustive testing on these patches all by themselves. Having other people apply these patches and try them out saves huge amounts of time, and often finds bugs that can be repaired by the authors before review by Digium staff (which saves even more time.) Please see if you can find some time to apply one or two patches to your current-CVS system, and then note your results in the ticket. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Start extension
At 6:23 PM +0200 10/19/03, rnc Info Lists wrote: I have my sip phones going into the context [from-sip] and would like to play an introduction message and then have the caller enter the extension. The message (dir-info was picked just to have something) doesn't play. Maybe I misunderstood the s extension. According to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The following is in extensions.com: [from-sip] exten= s,1,Answer exten= s,2,Background,dir-intro exten= s,3,DigitTimeout,3 exten= s,4,ResponseTimeout,10 exten = 2000,1,Dial(SIP/2000,20) exten = 2001,1,Dial(SIP/2000,21) Any ideas are appreciated. Robert The s extension is used when there is no known called number. In other words, if you are dialing 2000, the dialplan will always prefer the priority list for 2000 instead of going to 's', so that is why your current system doesn't work. I assume that you are ignoring the actual number that people are dialing, since you are forcing them to re-enter an extension after hearing a recording. This seems a little odd, but that's what you describe, so I'll give you an example with that method. I use the _. modifier to grab all dialed sequences and then simply re-map that to a new context and extension for ease of processing. If you're dealing with wildcards for all numbers, you should get rid of them as quickly as possible and turn them into something that can be handled without using wildcard matching. I could have easily specified a string instead of a number (i.e.: allnumbers instead of ) as that eliminates the ability for the normal DTMF user to input your magic secret number. However, that is up to the programmer as to what method they choose. [from-sip] exten = _.,1,Goto(from-sip2,,1) exten = h,1,Hangup [from-sip2] exten = ,1,Answer exten = ,2,DigitTimeout(3) exten = ,3,ResponseTimeout(10) exten = ,4,Background(dir-intro) exten = h,1,Hangup exten = t,1,Goto(,4) exten = i,1,Playback(invalid) exten = i,2,Goto(,4) exten = 2001,1,Dial(SIP/2000,20) exten = 2000,1,Dial(SIP/2000,20) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk/Freebsd network connections
Tilghman Lesher wrote: On Sunday 19 October 2003 11:45, Olle E. Johansson wrote: For those of you running * on FreeBSD: I compiled everything and can start. Sockstat -l shows that Asterisk listens on the correct interfaces and ports. Sniffing, I see registrations coming in to SIP debug, but nothing seems so reach Asterisk except IAX registration from a peer. Can't dial anywhere, can't reach the Asterisk with asterisk -r. The code to read the current routing table has not yet made its way into the FreeBSD port. You'll need to set a specific IP address as the bindaddr in sip.conf. Already there, but still doesn't work. Any use of loopback interface? 127.0.0.1 ? And yes, sockstat confirms binding to IP address. Yes, Virginia, Linux does have a better way of doing this. :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Start extension
The s extension is used when there is no known called number. In other words, if you are dialing 2000, the dialplan will always prefer the priority list for 2000 instead of going to 's', so that is why your current system doesn't work. John, Thanks for the details. Actually what I want to do is to play an announcement and then pass the person along to the extension that they dialed. Use of Background was probably not the correct command. (sb. Playback). YOur details clear up the order of processing. Think I can get it from here. Thanks again. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco ATA Call Waiting
To receive the call on the CO line, a hookflash is required on the CO line. If I hookflash on the ATA line it sends the hookflash to asterisk no to the co line. -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Sunday, October 19, 2003 1:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco ATA Call Waiting Kevin wrote: I have a CO line hooked to an X100P. I also have call waiting on the POTS line. How do you answer a call waiting call on a Cisco ATA? I get the Caller ID, when I hookflash I get dial tone, how do I hookflash to get CO line to connect to the caller? if you have a phone connected directly to the line how do you change from call 1 to call 2?? If you do somthing like press #, then that same thing should work to change calls on the X100P.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 5 safe to use?
You mean 5.3. I'm currently running 5.3 on my 7960. bkw On Sun, 19 Oct 2003, Tomica Crnek wrote: I am using 5.03 image on 7940 and 7960 and it is ok - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 1:03 PM Subject: RE: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 5 safe to use? On 18/10/2003 at 18:22 Juan J. Sierralta P. wrote: On Sat, 2003-10-18 at 17:25, Paul Mahler wrote: Howdy, Does anyone know if there are any problems running Asterisk when using later 7960 SIP versions like 04.04 or 05.03? I have 4.4 running without problems. I have 05.03 running with no problems... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Start extension
No! s is executed when Asterisk has no destination extension. For example when a call comes in from the PSTN Asterisk doesn't know what extension to send the call to, so it sends it to the s extension. On Sun, 2003-10-19 at 11:23, rnc Info Lists wrote: I have my sip phones going into the context [from-sip] and would like to play an introduction message and then have the caller enter the extension. The message (dir-info was picked just to have something) doesn't play. Maybe I misunderstood the s extension. According to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The following is in extensions.com: [from-sip] exten= s,1,Answer exten= s,2,Background,dir-intro exten= s,3,DigitTimeout,3 exten= s,4,ResponseTimeout,10 exten = 2000,1,Dial(SIP/2000,20) exten = 2001,1,Dial(SIP/2000,21) Any ideas are appreciated. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback request: AGI GET DATA change - termination digits
Why use this rather than STREAM FILE? On Sat, 2003-10-18 at 16:50, Paul Crick wrote: ** REPOST: A week later and no feedback - am I the only one ** who'd find this functionality useful? No other AGI stuff ** out there needing something similar? I'd like some feedback on potentially submitting a request (and probably a patch too) to change the way the AGI command GET DATA works. Right now, # terminates the entry, which is then returned with the # stripped off the end. What I'd like is to allow user configurable termination digits, which are not stripped off the end. Reasoning: Some entries you'd like to terminate with #. Right now it's fine, you can tell if # was pressed or not by looking for the lack of a (timeout) entry in the returned result. You may want to allow * to cancel an entry. This is not possible right now. Systems I've coded previously allow # to terminate and complete a digit entry, * to correct an incorrect entry (playing the prompt again and restarting digit collection). Pressing * with no prior digit entry cancels the step and returns to the previous menu. I guess there's a compatibility issue with stuff that's out there already but if it was an optional 4th parameter this would be backwards compatible. Proposed new syntax: GET DATA filename timeout maxdigits terminator If terminator is specified (and it may be multicharacter, like *# to give me the functionality above), return the digit string collected so far, including the terminating digit. The calling app can strip the trailing character if needed. Thoughts? Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Success story
Hi Marcel, Good to hear that everything is working well for you. Just one question, how do your users transfer calls to each other? I.e. is it announced or blind? Regards, Aaron. - Original Message - From: Marcel Prisi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 15, 2003 12:02 AM Subject: [Asterisk-Users] Success story Hi all, Just a little note for the records and archives. We see many small glitches / troubles in the mailing-list but rarely success stories ... Here's one : Asterisk is running perfectly fine in our setup : Debian 3.0 stable / Athlon 1.8, 256 MB Ram / Digium E-100P / Swisscom PRI isdn We have 6 companies (more to come) sharing the system in the building with a total of about 20 Grandstream 101's, a Cisco 7940 and a few SJPhones / XTen. We have many different setups (vhosts-like) for every company and it is a real joy ! Had some troubles at first with the PRI configuration (first time) but now everything is running perfectly well, and everyone is really impressed here ! Thanks to all developers for such an impressive piece of software ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
Also, what do the different 'format' numbers mean? Is there a table
somewhere showing which format is which number?
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 7:37 AM
Subject: [Asterisk-Users] use of SIP SHOW CHANNELS question
I am trying to figure out the correct syntax for the CLI command SIP SHOW
CHANNELS. I have tried
SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is
connected such as:
-- Zap/15-1 is ringing
-- Zap/15-1 answered SIP/206-4299
asterisk*CLI sip show channel SIP/206-4299
No such SIP Call ID 'SIP/206-4299'
I always get the No such SIP Call ID ...
Thanks, Walker
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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Re: [Asterisk-Users] Grandstream phone :(
I have also found the build quality to be very poor, i.e. I bought 3 phones a few weeks ago, and when they arrived one of them had orange glue spots on it, another had a hook that was sticky due to glue on it, and the other didnt have any screws holding the motherboard in place! - Original Message - From: Bartosz Jozwiak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 17, 2003 11:28 AM Subject: [Asterisk-Users] Grandstream phone :( Hello, I have just bought two Grandstream BudgeTone phones. One is working ok - no problems with using so far (3 days) Another one just hangs. And the second thing, microphone in handset somehow is not where it suppouse to be, so you cannot almost hear what the person is talking about :( Does somebody has problems like me with Grandstream BT101? The software of phones is the newest one. Regards, -- Bart - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 5 safe to use?
P0S3-05-3-00.bin - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 11:03 PM Subject: Re: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 5 safe to use? You mean 5.3. I'm currently running 5.3 on my 7960. bkw On Sun, 19 Oct 2003, Tomica Crnek wrote: I am using 5.03 image on 7940 and 7960 and it is ok - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 1:03 PM Subject: RE: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 5 safe to use? On 18/10/2003 at 18:22 Juan J. Sierralta P. wrote: On Sat, 2003-10-18 at 17:25, Paul Mahler wrote: Howdy, Does anyone know if there are any problems running Asterisk when using later 7960 SIP versions like 04.04 or 05.03? I have 4.4 running without problems. I have 05.03 running with no problems... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr
[Asterisk-Users] Music on hold...
Hi, I need a sound card and mpg123 for music on hold??? When I call Digium the guys toll me is not necessary to have a sound card. My music on hold doesn't work :(( Best regards, Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Thursday, October 16, 2003 8:24 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SER vs STUND with Asterisk.. Olle E. Johansson wrote: WipeOut wrote: Olle E. Johansson wrote: WipeOut wrote: Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to address NAT is to use STUN or uPNP.. STUN is helpful, but as I understand it analyzes the situation and reports the configuration of a NAT. It doesn't help you keeping the NAT session open, as SER module nathelper or the FWD/Jasomi solution. Check here http://www.voip-info.org/wiki-SER+module+nathelper It's ugly, but what it does is sending UDP packets from the outside to the NAT to keep the ports open for incoming calls. NAT is an ugly thing, so it propably needs ugly solutions... ;-) Looking at that page you mentioned it still seems to me that the nathelper module for SER and adding nat=yes to the sip.conf essentially do the same thing apart from the NAT pings you mentioned below.. Right. There's also more commands so that you can tweak SER into doing different kinds of SIP message mangling than the - still rather undocumented - nat=yes. My guess is that nat=yes changes the Contact to the actual IP used to contact Asterisk, not the IP given in the SIP headers. Right? Not sure about the intimate details of what nat=yes does exactly but it defiantely works, also have just found out (thanks to John Todd) the if you add qualify=500 to your UA configuration in the sip.conf then it essentially uses keep alives in the form of a OPTIONS request every 60 seconds.. So by having nat= and qualify= removes the need to have SER and the nathelper module.. (No doubt there is more that SER can do and if you really need those features then go for it..) As I understand it, it works like this: * Client on the inside of a NAT registers to an outside SIP Proxy * THe outside SIP Proxy keeps sending UDP packets (NAT PINGS) to the client to keep the UDP session open in the NAT * When someone calls, the session is open and the client (UAC/S) may answer... * In addition to the solution for handling SIP this way, there's a need for an RTP media server to handle the RTP stream. I guess that if you use SER or STUN and Asterisk the RTP is still going to be an issue if the call is needing to go between two SIP UA's that are both behind NAT (UA---NAT--Internet--NAT--UA) so the RTP streams are going to have to go via the central server (aka canreinvite=no in Asterisk).. So if NAT is in the picture you have no choice but to load the server with all the traffic.. Right. That's where the PortaOne RTP proxy - or Asterisk - come in. The RTP proxy in combination with SERs nathelper changes the SDP to point to the RTP proxy in this case and informs the RTP proxy of the session through a Unix pipe. Personally I think I would stick with Asterisk to handle all the RTP traffic, just by adding canreinvite=no to the sip.conf will cause all traffice between the endpoints to go via Asterisk.. The fewer systems that need to be tied together the better IMO.. If it can all be done with one then there is less to go wrong.. :) So my question is would it not be better to couple STUND (Vovida.org) with Asterisk and then use nat=yes in the sip.conf for UA's that do not support STUN, instead of using SER which would be like learning Asterisk all over again and would require you to learn how to use the SER config language to manage your NAT transtaltions.. Integrating a STUN server into ASterisk... I don't see the point. But if you're talking about asterisk as a SIP client (registrering to other SIP servers) supporting STUN to find out if it's behind a NAT and how the NAT works, yes, that's a good idea. I wasn't talking about intergrating STUN into asterisk, I was thinking more along the lines of using STUND in conjunction with Asterisk instead of SER and Asterisk.. :) Sorry, my misunderstanding. Are you thinking the way I did, with Asterisk as a SIP client or are you thinking of supporting Asterisk's SIP clients, the phones, with a STUND? I was thinking of the supporting the SIP clients (phones).. I think that it is the resposibility of the server to handle as much complexity as possible making it easier for the UA's to be configured.. So if you are trying to connect Asterisk(as a client) to a third party to route your calls I would say that it is their responsibility to handle NAT issues.. Thats not to say that Asterisk can be made to help out as well.. We need to form a strategy of what can be done with Asterisk's
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
On Sunday 19 October 2003 18:01, Aaron Martin wrote: Also, what do the different 'format' numbers mean? Is there a table somewhere showing which format is which number? *CLI show codec 4 4 (1 2) G.711 u-law *CLI show audio codecs -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!)
On the analog front, since we're talking about paging or intercom: it has been mentioned that the Sayson ADSI phones (Aastra?) are integrated with an Altigen PBX system, that the phones can support paging. See the link below. http://www.sayson.com/product/Altigen.htm If this could be combined with a Cisco ATA-186 or similar product, would it not be possible to have paging via SIP delivery? I have not put anything other than the most basic thought into this, and I don't know exactly how they support ADSI-based paging, but has anyone worked on this? I mentioned it via IRC and someone said they took a half-hearted stab at it, but it didn't sound like it had been explored in depth. That is a major feature issue that most business customers are looking for, and it would be great to have some combination of devices that offered paging or intercom. JT I have just put in some analog phones that offer intercom/paging and work rather well so far. They are Smartalk phones. (NOT ADSI) The guys there have been real good about helping me get buttons programmed exactly the way I want them etc. These phones also allow for staion monitoring and have an attendant console. These phones use 4-pair wiring so 3 lines + power pair. The power pair is also responsible for the intercom/paging/station monitoring. With the right wiring it might work with an ata device. Just a thought. http://smartalk.ca/nrgover.htm Andy Looks useful, but requires essentially a second line to work as a pager or intercom. Not necessarily a bad thing, but as an an example it would require a whole ATA-186 to just get one line and the paging feature working, plus perhaps some additional wiring to work that all into the power pair. Also, it is unclear if that will work at all, since there is no documentation on their website about how, exactly, the pager or intercom features work. It's completely undiscussed at the nuts-and-bolts level (though they tell you what buttons to push.) I'm still waiting for a _good_ implementation of paging and intercom for use with SIP. Let's define good for an intercom/pager (i/p): Mandatory: - i/p can be activated while user is off hook (speakerphone or handset) - deskset has local option for refusal of i/p while off-hook - deskset has local option for refusal of i/p completely - deskset has separate volume controls for i/p messages - deskset plays announcement beep before intercom auto-answer - password authenticated SIP messages for auto-answer i/p (i.e.: no spam calls get through that aren't authenticated by a chosen upstream proxy server) optional: - pager announcement audio via multicast (very, very optional) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream phone :(
If anyone buys GS phones from us (Chagres Technoloiges) and runs into such problems, please let us know. We will do what needs to be done to make it right. I'll make sure that this feed back gets to Grandstream's president.. John Brown, CEO Chagres Technologies, Inc On Mon, Oct 20, 2003 at 12:08:49PM +1300, Aaron Martin wrote: I have also found the build quality to be very poor, i.e. I bought 3 phones a few weeks ago, and when they arrived one of them had orange glue spots on it, another had a hook that was sticky due to glue on it, and the other didnt have any screws holding the motherboard in place! - Original Message - From: Bartosz Jozwiak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 17, 2003 11:28 AM Subject: [Asterisk-Users] Grandstream phone :( Hello, I have just bought two Grandstream BudgeTone phones. One is working ok - no problems with using so far (3 days) Another one just hangs. And the second thing, microphone in handset somehow is not where it suppouse to be, so you cannot almost hear what the person is talking about :( Does somebody has problems like me with Grandstream BT101? The software of phones is the newest one. Regards, -- Bart - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on hold...
No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus - Original Message - From: Chris Hariga [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... Hi, I need a sound card and mpg123 for music on hold??? When I call Digium the guys toll me is not necessary to have a sound card. My music on hold doesn't work :(( Best regards, Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Thursday, October 16, 2003 8:24 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SER vs STUND with Asterisk.. Olle E. Johansson wrote: WipeOut wrote: Olle E. Johansson wrote: WipeOut wrote: Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to address NAT is to use STUN or uPNP.. STUN is helpful, but as I understand it analyzes the situation and reports the configuration of a NAT. It doesn't help you keeping the NAT session open, as SER module nathelper or the FWD/Jasomi solution. Check here http://www.voip-info.org/wiki-SER+module+nathelper It's ugly, but what it does is sending UDP packets from the outside to the NAT to keep the ports open for incoming calls. NAT is an ugly thing, so it propably needs ugly solutions... ;-) Looking at that page you mentioned it still seems to me that the nathelper module for SER and adding nat=yes to the sip.conf essentially do the same thing apart from the NAT pings you mentioned below.. Right. There's also more commands so that you can tweak SER into doing different kinds of SIP message mangling than the - still rather undocumented - nat=yes. My guess is that nat=yes changes the Contact to the actual IP used to contact Asterisk, not the IP given in the SIP headers. Right? Not sure about the intimate details of what nat=yes does exactly but it defiantely works, also have just found out (thanks to John Todd) the if you add qualify=500 to your UA configuration in the sip.conf then it essentially uses keep alives in the form of a OPTIONS request every 60 seconds.. So by having nat= and qualify= removes the need to have SER and the nathelper module.. (No doubt there is more that SER can do and if you really need those features then go for it..) As I understand it, it works like this: * Client on the inside of a NAT registers to an outside SIP Proxy * THe outside SIP Proxy keeps sending UDP packets (NAT PINGS) to the client to keep the UDP session open in the NAT * When someone calls, the session is open and the client (UAC/S) may answer... * In addition to the solution for handling SIP this way, there's a need for an RTP media server to handle the RTP stream. I guess that if you use SER or STUN and Asterisk the RTP is still going to be an issue if the call is needing to go between two SIP UA's that are both behind NAT (UA---NAT--Internet--NAT--UA) so the RTP streams are going to have to go via the central server (aka canreinvite=no in Asterisk).. So if NAT is in the picture you have no choice but to load the server with all the traffic.. Right. That's where the PortaOne RTP proxy - or Asterisk - come in. The RTP proxy in combination with SERs nathelper changes the SDP to point to the RTP proxy in this case and informs the RTP proxy of the session through a Unix pipe. Personally I think I would stick with Asterisk to handle all the RTP traffic, just by adding canreinvite=no to the sip.conf will cause all traffice between the endpoints to go via Asterisk.. The fewer systems that need to be tied together the better IMO.. If it can all be done with one then there is less to go wrong.. :) So my question is would it not be better to couple STUND (Vovida.org) with Asterisk and then use nat=yes in the sip.conf for UA's that do not support STUN, instead of using SER which would be like learning Asterisk all over again and would require you to learn how to use the SER config language to manage your NAT transtaltions.. Integrating a STUN server into ASterisk... I don't see the point. But if you're talking about asterisk as a SIP client (registrering to other SIP servers) supporting STUN to find out if it's behind a NAT and how the NAT works, yes, that's a good idea. I wasn't talking about intergrating STUN into asterisk, I was thinking more along the lines of using STUND in conjunction with Asterisk instead of SER and Asterisk.. :) Sorry, my misunderstanding. Are you thinking the way I did, with Asterisk as a SIP client or are you thinking of supporting Asterisk's SIP clients, the phones, with a STUND? I was thinking of the supporting the SIP clients (phones).. I think that it is the
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
Please do:
sip show channels
See Channel ID number and repear the 'sip show channel ' where is
Channel ID number from 'sip show channels'.
Regards,
Gus
- Original Message -
From: Aaron Martin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 19, 2003 8:01 PM
Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
Also, what do the different 'format' numbers mean? Is there a table
somewhere showing which format is which number?
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 7:37 AM
Subject: [Asterisk-Users] use of SIP SHOW CHANNELS question
I am trying to figure out the correct syntax for the CLI command SIP
SHOW
CHANNELS. I have tried
SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is
connected such as:
-- Zap/15-1 is ringing
-- Zap/15-1 answered SIP/206-4299
asterisk*CLI sip show channel SIP/206-4299
No such SIP Call ID 'SIP/206-4299'
I always get the No such SIP Call ID ...
Thanks, Walker
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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Re: [Asterisk-Users] Music on hold...
On Sunday 19 October 2003 18:19, Chris Hariga wrote: Hi, I need a sound card and mpg123 for music on hold??? When I call Digium the guys toll me is not necessary to have a sound card. My music on hold doesn't work :(( Sound card is not necessary, but mpg123 is. Please make sure that you really have mpg123, not RedHat's mpg321 symlinked to mpg123. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newb - want to create a Dialpad like system
Hi all, i am planning to create Dialpad like system for fun. i want to build itin such a way that one can use either web based app or GnoPhone / MsnMessenger to connect to my server and then dial a land line.i did a search on the archives but couldnt find any good pointers. i would appreciate if someone could let me know whether this possible to create this app using asteriskand send me some pointers. i hv the following RH 7.3 Dialogic D41 thanks a lot, -Balaji Do you Yahoo!? The New Yahoo! Shopping - with improved product search
[Asterisk-Users] SIP soft phone
Hi, I am new in VOIP area, so any help is really appreciated. I setup asterisk at home and I am trying softphone. I download SJphone from SJlabs and I can place calls. Question is, how can I make a call to that softphone What would be config in asterisk and in softphone. I am trying to use SIP. Also, can I make call outside trough modem on Linux? and how. Thank you for any help Drazen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
I dont think that is it: *CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 12:40 PM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question On Sunday 19 October 2003 18:01, Aaron Martin wrote: Also, what do the different 'format' numbers mean? Is there a table somewhere showing which format is which number? *CLI show codec 4 4 (1 2) G.711 u-law *CLI show audio codecs -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!)
On Sun, 2003-10-19 at 18:40, John Todd wrote: Looks useful, but requires essentially a second line to work as a pager or intercom. Not necessarily a bad thing, but as an an example it would require a whole ATA-186 to just get one line and the paging feature working, plus perhaps some additional wiring to work that all into the power pair. Also, it is unclear if that will work at all, since there is no documentation on their website about how, exactly, the pager or intercom features work. It's completely undiscussed at the nuts-and-bolts level (though they tell you what buttons to push.) Whats important to note, I think these do out of band negotiation on a common wire pair. These phones usually are setup on the end of a set of analog phones lines all wired the same. Basically instead of having a intelligent switch front the phones, each phone communicates to each other what it is doing. Since they need common wiring, a ATA186 wouldn't help you here. I'm still waiting for a _good_ implementation of paging and intercom for use with SIP. Let's define good for an intercom/pager (i/p): Mandatory: - i/p can be activated while user is off hook (speakerphone or handset) - deskset has local option for refusal of i/p while off-hook - deskset has local option for refusal of i/p completely - deskset has separate volume controls for i/p messages - deskset plays announcement beep before intercom auto-answer - password authenticated SIP messages for auto-answer i/p (i.e.: no spam calls get through that aren't authenticated by a chosen upstream proxy server) optional: - pager announcement audio via multicast (very, very optional) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold...
On Sun, 2003-10-19 at 21:39, CW_ASN wrote: No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? AFAIK, MOH does no nees a zaptel device or zaptel dummy driver, just MeetMe needs it. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
*CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) Rebuild your system. I had that happen and my Makefile was screwed up and didn't actually build any codecs. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
On Sunday 19 October 2003 20:56, Aaron Martin wrote: I dont think that is it: *CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) You're using an old version of Asterisk. Please checkout the latest CVS and recompile. The command 'show codec n' is built into the core of Asterisk. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Sunday, October 19, 2003 6:40 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!) On the analog front, since we're talking about paging or intercom: it has been mentioned that the Sayson ADSI phones (Aastra?) are integrated with an Altigen PBX system, that the phones can support paging. See the link below. http://www.sayson.com/product/Altigen.htm If this could be combined with a Cisco ATA-186 or similar product, would it not be possible to have paging via SIP delivery? I have not put anything other than the most basic thought into this, and I don't know exactly how they support ADSI-based paging, but has anyone worked on this? I mentioned it via IRC and someone said they took a half-hearted stab at it, but it didn't sound like it had been explored in depth. That is a major feature issue that most business customers are looking for, and it would be great to have some combination of devices that offered paging or intercom. JT I have just put in some analog phones that offer intercom/paging and work rather well so far. They are Smartalk phones. (NOT ADSI) The guys there have been real good about helping me get buttons programmed exactly the way I want them etc. These phones also allow for staion monitoring and have an attendant console. These phones use 4-pair wiring so 3 lines + power pair. The power pair is also responsible for the intercom/paging/station monitoring. With the right wiring it might work with an ata device. Just a thought. http://smartalk.ca/nrgover.htm Andy Looks useful, but requires essentially a second line to work as a pager or intercom. Not necessarily a bad thing, but as an an example it would require a whole ATA-186 to just get one line and the paging feature working, plus perhaps some additional wiring to work that all into the power pair. Also, it is unclear if that will work at all, since there is no documentation on their website about how, exactly, the pager or intercom features work. It's completely undiscussed at the nuts-and-bolts level (though they tell you what buttons to push.) Snip JT It runs on Cat5. The paging/intercom/station monitoring all occurs over the power pair. (pin18)The power pair wouldn't go to * at all. It works seemlessly so far on a channel bank. As for the ATA-186, I am not familiar with the wiring of the ATA, but it seems that you could use a surface mount jack with 1 or 2 lines as needed. You'd just have to get pair 18 hooked up to the central power supply. It may be more trouble than its worth to you, but I mentioned it in response to your earlier post just because people always seem to be asking for these features regarless of the type of phone. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
On Sunday 19 October 2003 21:30, Andrew Kohlsmith wrote: *CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) Rebuild your system. I had that happen and my Makefile was screwed up and didn't actually build any codecs. Stop already. The command serves only as a translation table. It does NOT, repeat NOT, repeat NOT, suggest anything about which codecs are actually loaded. If you actually run this now, you should get the following message on your console (put there SPECIFICALLY because people were confusing what the command actually does): Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
If you had a look under the help as the prompt said and entered help show - you would have found that it is show codecs Paul - Original Message - From: Aaron Martin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 11:56 AM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question I dont think that is it: *CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 12:40 PM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question On Sunday 19 October 2003 18:01, Aaron Martin wrote: Also, what do the different 'format' numbers mean? Is there a table somewhere showing which format is which number? *CLI show codec 4 4 (1 2) G.711 u-law *CLI show audio codecs -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold...
Yes, right... sorry. - Original Message - From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 11:27 PM Subject: Re: [Asterisk-Users] Music on hold... On Sun, 2003-10-19 at 21:39, CW_ASN wrote: No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? AFAIK, MOH does no nees a zaptel device or zaptel dummy driver, just MeetMe needs it. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modem
Can I use modem on Linux box for making outgoing calls? And receiving to? Drazen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
How about 'show translations' command? This machine doesn't have G.729 codec licenses... AFAIK, this command calculates the cost for each translation... noc2pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - GSM - - 2 2 2 1 13 - - 38 ULAW - 6 - 1 2 1 13 - - 38 ALAW - 6 1 - 2 1 13 - - 38 MP3 - 15 11 11 11 10 22 - - 47 ADPCM - 6 2 2 - 1 13 - - 38 SLINR - 5 1 1 1 - 12 - - 37 LPC10 - 9 5 5 5 4 - - - 41 9A - - - - - - - - - - EX - - - - - - - - - - ILBC - 11 7 7 7 6 8 - - - noc2pbx*CLI And this machine has G.729 loaded: noc2pbx2*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 62 56 56 56 55 64106 - 106 GSM240 - 3 3 3 2 11 53 - 53 ULAW239 8 - 1 2 1 10 52 - 52 ALAW239 8 1 - 2 1 10 52 - 52 MP3258 27 21 21 21 20 29 71 - 71 ADPCM239 8 2 2 - 1 10 52 - 52 SLINR238 7 1 1 1 - 9 51 - 51 LPC10245 14 8 8 8 7 - 58 - 58 G729A 100237 16 10 10 10 9 18 - - 100050 EX - - - - - - - - - - ILBC247 16 10 10 10 9 18 0 - - noc2pbx2*CLI Regards, Gus - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 11:44 PM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question On Sunday 19 October 2003 21:30, Andrew Kohlsmith wrote: *CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) Rebuild your system. I had that happen and my Makefile was screwed up and didn't actually build any codecs. Stop already. The command serves only as a translation table. It does NOT, repeat NOT, repeat NOT, suggest anything about which codecs are actually loaded. If you actually run this now, you should get the following message on your console (put there SPECIFICALLY because people were confusing what the command actually does): Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modem
On Sunday 19 October 2003 22:39, Drazen Vidakovic wrote: Can I use modem on Linux box for making outgoing calls? And receiving to? http://asstricks.org/faq.html See question 7. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] use of SIP SHOW CHANNELS question
Why is mine different? localhost*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 45 41 41 41 40 46 - - 73 GSM713 - 2 2 2 1 7 - - 34 ULAW713 6 - 1 2 1 7 - - 34 ALAW713 6 1 - 2 1 7 - - 34 MP3723 16 12 12 12 11 17 - - 44 ADPCM713 6 2 2 - 1 7 - - 34 SLINR712 5 1 1 1 - 6 - - 33 LPC10717 10 6 6 6 5 - - - 38 G729A - - - - - - - - - - SPEEX - - - - - - - - - - ILBC718 11 7 7 7 6 12 - - - Notice the G723 almost 4x higher. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of CW_ASN Sent: Sunday, October 19, 2003 11:43 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question pamAssassin 2.55 (1.174.2.19-2003-05-19-exp) How about 'show translations' command? This machine doesn't have G.729 codec licenses... AFAIK, this command calculates the cost for each translation... noc2pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - GSM - - 2 2 2 1 13 - - 38 ULAW - 6 - 1 2 1 13 - - 38 ALAW - 6 1 - 2 1 13 - - 38 MP3 - 15 11 11 11 10 22 - - 47 ADPCM - 6 2 2 - 1 13 - - 38 SLINR - 5 1 1 1 - 12 - - 37 LPC10 - 9 5 5 5 4 - - - 41 9A - - - - - - - - - - EX - - - - - - - - - - ILBC - 11 7 7 7 6 8 - - - noc2pbx*CLI And this machine has G.729 loaded: noc2pbx2*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 62 56 56 56 55 64106 - 106 GSM240 - 3 3 3 2 11 53 - 53 ULAW239 8 - 1 2 1 10 52 - 52 ALAW239 8 1 - 2 1 10 52 - 52 MP3258 27 21 21 21 20 29 71 - 71 ADPCM239 8 2 2 - 1 10 52 - 52 SLINR238 7 1 1 1 - 9 51 - 51 LPC10245 14 8 8 8 7 - 58 - 58 G729A 100237 16 10 10 10 9 18 - - 100050 EX - - - - - - - - - - ILBC247 16 10 10 10 9 18 0 - - noc2pbx2*CLI Regards, Gus - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 11:44 PM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question On Sunday 19 October 2003 21:30, Andrew Kohlsmith wrote: *CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) Rebuild your system. I had that happen and my Makefile was screwed up and didn't actually build any codecs. Stop already. The command serves only as a translation table. It does NOT, repeat NOT, repeat NOT, suggest anything about which codecs are actually loaded. If you actually run this now, you should get the following message on your console (put there SPECIFICALLY because people were confusing what the command actually does): Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
I think the system calculates the cost in CPU milliseconds to translate each codec with another one. In different machines (slower or faster), the costs vary. The machine that have G.729 codec loaded is a Pentium III 733 MHz, 128MB. I assume this is a way to know which codecs was loaded, because if I unload G.729 codec disappears from 'show translation' printout. But, who knows... Regards, Gus - Original Message - From: Andrew Joakimsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 12:59 AM Subject: RE: [Asterisk-Users] use of SIP SHOW CHANNELS question Why is mine different? localhost*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 45 41 41 41 40 46 - - 73 GSM713 - 2 2 2 1 7 - - 34 ULAW713 6 - 1 2 1 7 - - 34 ALAW713 6 1 - 2 1 7 - - 34 MP3723 16 12 12 12 11 17 - - 44 ADPCM713 6 2 2 - 1 7 - - 34 SLINR712 5 1 1 1 - 6 - - 33 LPC10717 10 6 6 6 5 - - - 38 G729A - - - - - - - - - - SPEEX - - - - - - - - - - ILBC718 11 7 7 7 6 12 - - - Notice the G723 almost 4x higher. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of CW_ASN Sent: Sunday, October 19, 2003 11:43 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question pamAssassin 2.55 (1.174.2.19-2003-05-19-exp) How about 'show translations' command? This machine doesn't have G.729 codec licenses... AFAIK, this command calculates the cost for each translation... noc2pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - GSM - - 2 2 2 1 13 - - 38 ULAW - 6 - 1 2 1 13 - - 38 ALAW - 6 1 - 2 1 13 - - 38 MP3 - 15 11 11 11 10 22 - - 47 ADPCM - 6 2 2 - 1 13 - - 38 SLINR - 5 1 1 1 - 12 - - 37 LPC10 - 9 5 5 5 4 - - - 41 9A - - - - - - - - - - EX - - - - - - - - - - ILBC - 11 7 7 7 6 8 - - - noc2pbx*CLI And this machine has G.729 loaded: noc2pbx2*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 62 56 56 56 55 64106 - 106 GSM240 - 3 3 3 2 11 53 - 53 ULAW239 8 - 1 2 1 10 52 - 52 ALAW239 8 1 - 2 1 10 52 - 52 MP3258 27 21 21 21 20 29 71 - 71 ADPCM239 8 2 2 - 1 10 52 - 52 SLINR238 7 1 1 1 - 9 51 - 51 LPC10245 14 8 8 8 7 - 58 - 58 G729A 100237 16 10 10 10 9 18 - - 100050 EX - - - - - - - - - - ILBC247 16 10 10 10 9 18 0 - - noc2pbx2*CLI Regards, Gus - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 11:44 PM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question On Sunday 19 October 2003 21:30, Andrew Kohlsmith wrote: *CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) Rebuild your system. I had that happen and my Makefile was screwed up and didn't actually build any codecs.
RE: [Asterisk-Users] use of SIP SHOW CHANNELS question
Interesting because what I posted was from an Athlon 1600 or so machine. And I thought I had removed the G723 codec from it... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of CW_ASN Sent: Monday, October 20, 2003 12:12 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question I think the system calculates the cost in CPU milliseconds to translate each codec with another one. In different machines (slower or faster), the costs vary. The machine that have G.729 codec loaded is a Pentium III 733 MHz, 128MB. I assume this is a way to know which codecs was loaded, because if I unload G.729 codec disappears from 'show translation' printout. But, who knows... Regards, Gus - Original Message - From: Andrew Joakimsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 12:59 AM Subject: RE: [Asterisk-Users] use of SIP SHOW CHANNELS question Why is mine different? localhost*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 45 41 41 41 40 46 - - 73 GSM713 - 2 2 2 1 7 - - 34 ULAW713 6 - 1 2 1 7 - - 34 ALAW713 6 1 - 2 1 7 - - 34 MP3723 16 12 12 12 11 17 - - 44 ADPCM713 6 2 2 - 1 7 - - 34 SLINR712 5 1 1 1 - 6 - - 33 LPC10717 10 6 6 6 5 - - - 38 G729A - - - - - - - - - - SPEEX - - - - - - - - - - ILBC718 11 7 7 7 6 12 - - - Notice the G723 almost 4x higher. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of CW_ASN Sent: Sunday, October 19, 2003 11:43 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question pamAssassin 2.55 (1.174.2.19-2003-05-19-exp) How about 'show translations' command? This machine doesn't have G.729 codec licenses... AFAIK, this command calculates the cost for each translation... noc2pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - GSM - - 2 2 2 1 13 - - 38 ULAW - 6 - 1 2 1 13 - - 38 ALAW - 6 1 - 2 1 13 - - 38 MP3 - 15 11 11 11 10 22 - - 47 ADPCM - 6 2 2 - 1 13 - - 38 SLINR - 5 1 1 1 - 12 - - 37 LPC10 - 9 5 5 5 4 - - - 41 9A - - - - - - - - - - EX - - - - - - - - - - ILBC - 11 7 7 7 6 8 - - - noc2pbx*CLI And this machine has G.729 loaded: noc2pbx2*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 62 56 56 56 55 64106 - 106 GSM240 - 3 3 3 2 11 53 - 53 ULAW239 8 - 1 2 1 10 52 - 52 ALAW239 8 1 - 2 1 10 52 - 52 MP3258 27 21 21 21 20 29 71 - 71 ADPCM239 8 2 2 - 1 10 52 - 52 SLINR238 7 1 1 1 - 9 51 - 51 LPC10245 14 8 8 8 7 - 58 - 58 G729A 100237 16 10 10 10 9 18 - - 100050 EX - - - - - - - - - - ILBC247 16 10 10 10 9 18 0 - - noc2pbx2*CLI Regards, Gus - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To:
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
Yep, asterisk times how long each codec takes to translate a second worth of voice, obviously only on codecs you have installed I think the system calculates the cost in CPU milliseconds to translate each codec with another one. In different machines (slower or faster), the costs vary. The machine that have G.729 codec loaded is a Pentium III 733 MHz, 128MB. I assume this is a way to know which codecs was loaded, because if I unload G.729 codec disappears from 'show translation' printout. But, who knows... Regards, Gus - Original Message - From: Andrew Joakimsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 12:59 AM Subject: RE: [Asterisk-Users] use of SIP SHOW CHANNELS question Why is mine different? localhost*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 45 41 41 41 40 46 - - 73 GSM713 - 2 2 2 1 7 - - 34 ULAW713 6 - 1 2 1 7 - - 34 ALAW713 6 1 - 2 1 7 - - 34 MP3723 16 12 12 12 11 17 - - 44 ADPCM713 6 2 2 - 1 7 - - 34 SLINR712 5 1 1 1 - 6 - - 33 LPC10717 10 6 6 6 5 - - - 38 G729A - - - - - - - - - - SPEEX - - - - - - - - - - ILBC718 11 7 7 7 6 12 - - - Notice the G723 almost 4x higher. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of CW_ASN Sent: Sunday, October 19, 2003 11:43 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question pamAssassin 2.55 (1.174.2.19-2003-05-19-exp) How about 'show translations' command? This machine doesn't have G.729 codec licenses... AFAIK, this command calculates the cost for each translation... noc2pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - GSM - - 2 2 2 1 13 - - 38 ULAW - 6 - 1 2 1 13 - - 38 ALAW - 6 1 - 2 1 13 - - 38 MP3 - 15 11 11 11 10 22 - - 47 ADPCM - 6 2 2 - 1 13 - - 38 SLINR - 5 1 1 1 - 12 - - 37 LPC10 - 9 5 5 5 4 - - - 41 9A - - - - - - - - - - EX - - - - - - - - - - ILBC - 11 7 7 7 6 8 - - - noc2pbx*CLI And this machine has G.729 loaded: noc2pbx2*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723GSM ULAW ALAW ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 62 56 56 56 55 64106 - 106 GSM240 - 3 3 3 2 11 53 - 53 ULAW239 8 - 1 2 1 10 52 - 52 ALAW239 8 1 - 2 1 10 52 - 52 MP3258 27 21 21 21 20 29 71 - 71 ADPCM239 8 2 2 - 1 10 52 - 52 SLINR238 7 1 1 1 - 9 51 - 51 LPC10245 14 8 8 8 7 - 58 - 58 G729A 100237 16 10 10 10 9 18 - - 100050 EX - - - - - - - - - - ILBC247 16 10 10 10 9 18 0 - - noc2pbx2*CLI Regards, Gus - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 19, 2003 11:44 PM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question On Sunday 19 October 2003 21:30, Andrew Kohlsmith wrote:
Re: [Asterisk-Users] Project Completed [Files Attached]
Ohh, I forgot mentioning it. You can use it , modify it at your own or reproduce something more interesting. Just like an open source GNU project. Regards Azher PJ Welsh [EMAIL PROTECTED] wrote: I have not found the type of license you are using for this demo. Can you please confirm which one you plan to use for this.On Sun, Oct 19, 2003 at 05:16:29AM -0700, Azher Amin wrote: Thnx for the interest in the ivr sample [btw: I am not an expert in PERL/AGI :) ] comments are welocome. You can download the demo files and sounds from http://www.consulttech.com.pk/asterisk/IVR.rar There is a flowchart in the excel format thats shows how it works. you will need to place the sound files in the /backup/en directory, or u can change the code as well. I will be putting another application as soon as it will be prepared. Azher - Do you Yahoo!? The New Yahoo! Shopping - with improved p roduct search___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? The New Yahoo! Shopping - with improved product search
[Asterisk-Users] Your comment on the following test bed setup ?
Hello All, I would like to seek your comment on the architecture of Test Bed we plan to have for Asterisk in our Lab. --- HP Procurve|--- IP Phone 1 24 Ports Ethernet | IP Phone 2 Switch | IP Phone 3 --- | | | Wildcard T100P -- Asterisk Server--- Nortel Switch SL-100 - | |Wildcard TDM400P | - | | | | | | | | | | | | Analog Phones (1-4) Please let me know if I am missing anything in the above setup. Best regards, Tarun
