RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Joakimsen Sent: Saturday, October 18, 2003 12:55 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Paging/Intercom (was: OT - SIP Auto-Answer for Cisco 7940/7960!!) There is no rule, it is just my way of thinking. Everything related to multiple lines should be handled by the Asterisk server / PBX. Is there any specific advantage that you have seen to using multiline phones? For the most part * does handle it all. However, having only 1 line makes it difficult to field calls (as in a receptionist). If you are on the phone someone else calls, the Flash functionality no longer works. With 2 lines, a person could be on line 2, field a call on line 1 do a consultative transfer without any problem. This is the main advantage. Also, you could make 3 3-way calls on this phone, assuming you had all 3 lines configured live, and then bridge all 3 together to have a 6 person conference call from your phone. Most of my phones have only 1 line configured, but all of the secretaries have 2 lines. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System layout
Hi, I'm a bit new to phone systems technology, so sorry if this question may sound uninformed. I want to put together a system of about 20 stations. What I'm invisioning is a system where about 16 users have a inexpensive handset hooked up to their computer via some sort of modem and the computer would run their usual Windows apps with a client that serves as a more complex interface to voice mail and extended dialing features. I would like the handset to behave as a normal phone as long as the computer is turned on. The other 4 stations would be stand alone handsets so that a receptionist could answer and forward calls to either the user or his/her voicemail. Is something like this possible using asterisk software/digicom hardware? Would it be reliable for 20 stations/16 users, 6 phone lines? Also I'd like to have worldwide users appear to be on the local phone system through voice over ip. Is that possible? Our local network is 100 mb/s and our internet connection is 768kb/s in both directions. Would that be enough? Thanks, Jake Yes, Asterisk can do everything you've listed. - www.xten.com for Windows SIP voice clients - ? for USB handsets - xten supports some, I think - look on their website for vendors - www.cisco.com for 7960 nice operator stations - 768kbps is fine for 6 simultaneous calls (preferably something other than the g.711 codec, which is the highest quality sound by a slim margin but at least double the bandwidth of the nearest comparable codec) as long as it's not saturated with other traffic already. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System layout
Also I'd like to have worldwide users appear to be on the local phone system through voice over ip. Is that possible? Our local network is 100 mb/s and our internet connection is 768kb/s in both directions. Would that be enough? SIP users need to be on the same side of the firewall as the Asterisk server. So if you want to protect or inside users with a firewall your worldwide users will need another Asterisk server that is outside of your firewall on the public Internet. These two servers will have to be connected via iax2 which can pass through firewalls. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ 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] x-lite
Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr
Re: [Asterisk-Users] chan_skinny XML Files for 7920
Skinny is not well supported by Asterisk. This was the answer when I asked the same question few days ago. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 18, 2003 4:42 AM Subject: [Asterisk-Users] chan_skinny XML Files for 7920 Hi, I have a Cisco 7920 that I'm trying to get working with my * box. When the phone boots it requests XMLDefault.cnf.xml and SEPMACADDRESSHERE.cnf. I assume I set the line number, etc in the latter of the two. However I cannot find any reference to how this file is structured. Anyone know? I assume this is why I'm getting the errors below: Oct 17 19:47:24 WARNING[1357974832]: File chan_skinny.c, Line 1807 (handle_message): Client sent message #0 without first registering. Oct 17 19:47:24 ERROR[1357974832]: File chan_skinny.c, Line 1823 (handle_message): Rejecting Device SEP000D282E95F0: Device not found Oct 17 19:47:24 WARNING[1357974832]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 17 19:47:24 NOTICE[1357974832]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Connection reset by peer Thanks, Justin ___ 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
Re: [Asterisk-Users] x-lite
Tomica Crnek wrote: Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr Yes X-Lite works fine with Asterisk using G.711 or GSM codecs.. Under System SettingsSip Proxy [first proxy] 1- Enable: yes 2- Username: The name or number in your SIP.CONF [brackets] 3- Authorized User: Leave blank (and remark out in SIP.CONF if you have it in there.) 4- Password: Set the password. 5- Domain/Realm: The Name or IP of your Asterisk box. 6- Sip Proxy: The Name or IP of your Asterisk box. Thats should get it working.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x-lite
Hi! Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. 1. Read the FAQ on the xten site, the have good documentation over there. Also read their PDF manual. 2. If X-Lite displays the settings menu when you launch it then your settings are not correct/ not complete 3. There is a bug in the currently available build 1079 (version 1) of X-Lite where no matter which RTP ports you specify X-Lite will analyse your NAT/firewall and select the ports by itself. In other words: X-Lite doesn't care what range you enter as RTP ports. Press F9 to find out wich ports are in use. 4. By the way: You only enter the starting port, by default 8000. X-Lite will then use 8000 and 8001 for channel 1, 8002 and 8003 for channel 2, and 8004 and 8005 for channel 3. Greetings, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x-lite
Philipp von Klitzing wrote: 3. There is a bug in the currently available build 1079 (version 1) of X-Lite where no matter which RTP ports you specify X-Lite will analyse your NAT/firewall and select the ports by itself. In other words: X-Lite doesn't care what range you enter as RTP ports. Press F9 to find out wich ports are in use. Biuld 1082 is out on the xten site.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto Start
Hi all, Is there any way to get * to start when linux boots? I am running Red Hat 8.0, but a remote site I am testing IAX with has power problems and the server there keeps re-booting, would be nice if everything started up again automatically. I noticed this in the list the other day, I suggest people download and install dameontools http://cr.yp.to/daemontools.html and have asterisk as a supervised service. If it fails, supervise will restart it after 5 seconds. Regards...Martin Would this do the trick, or is this just when the system has already been run. Thanks in advance. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Start
David J Carter wrote: Hi all, Is there any way to get * to start when linux boots? I am running Red Hat 8.0, but a remote site I am testing IAX with has power problems and the server there keeps re-booting, would be nice if everything started up again automatically. I noticed this in the list the other day, I suggest people download and install dameontools http://cr.yp.to/daemontools.html and have asterisk as a supervised service. If it fails, supervise will restart it after 5 seconds. Regards...Martin Would this do the trick, or is this just when the system has already been run. Thanks in advance. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users For RedHat there is an init script provided for the zaptel drivers, and then you can just add safe_asterisk to your startup script.. This setup has worked well for me.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Start
Cheers, Do I add the safe_asterisk to the rc.local file? You may tell I am new to Linux. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: 18 October 2003 10:40 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto Start David J Carter wrote: Hi all, Is there any way to get * to start when linux boots? I am running Red Hat 8.0, but a remote site I am testing IAX with has power problems and the server there keeps re-booting, would be nice if everything started up again automatically. I noticed this in the list the other day, I suggest people download and install dameontools http://cr.yp.to/daemontools.html and have asterisk as a supervised service. If it fails, supervise will restart it after 5 seconds. Regards...Martin Would this do the trick, or is this just when the system has already been run. Thanks in advance. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users For RedHat there is an init script provided for the zaptel drivers, and then you can just add safe_asterisk to your startup script.. This setup has worked well for me.. Later.. ___ 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] AGI script question
Hi, I am trying to write an AGI script that executes a shell command in C ie. ls. I tried VERBOSE AGI command or to send the !command ls to stderr but the command does not execute it just displays on asterisk console. Does anybody has any idea about it? Thanks Dimitris __ 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
Re: [Asterisk-Users] Auto Start
David J Carter wrote: Cheers, Do I add the safe_asterisk to the rc.local file? You may tell I am new to Linux. Dave Yes, rc.local should do it for you.. Don't worry about being new to Linux, I have been using it for a few years and I am still learning, I still think I am a newbie.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x-lite
I did this, but ok, I'll try again - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 18, 2003 11:01 AM Subject: Re: [Asterisk-Users] x-lite Tomica Crnek wrote: Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr Yes X-Lite works fine with Asterisk using G.711 or GSM codecs.. Under System SettingsSip Proxy [first proxy] 1- Enable: yes 2- Username: The name or number in your SIP.CONF [brackets] 3- Authorized User: Leave blank (and remark out in SIP.CONF if you have it in there.) 4- Password: Set the password. 5- Domain/Realm: The Name or IP of your Asterisk box. 6- Sip Proxy: The Name or IP of your Asterisk box. Thats should get it working.. Later.. ___ 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
RE: [Asterisk-Users] Auto Start
I have put ./var/sbin/safe_asterisk in the rc.local file but it still doesn't start. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: 18 October 2003 11:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto Start David J Carter wrote: Cheers, Do I add the safe_asterisk to the rc.local file? You may tell I am new to Linux. Dave Yes, rc.local should do it for you.. Don't worry about being new to Linux, I have been using it for a few years and I am still learning, I still think I am a newbie.. :) ___ 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] Prob with Ringing multiple Channels
one more point, infact we tried with 'callprogress=yes', then both of the extensions starts ringing, but the callee can not hear the ringback... any suggestions...??? - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 17, 2003 12:45 PM Subject: [Asterisk-Users] Prob with Ringing multiple Channels hi, The prob is when we ring 2 channels simultaneously, only 1 channel is actually ringing. In our configuration, the Asterisk box is connected to an E1 channel bank, where 15 analog extensions are conencted to channelbank inturn. We tried following, Dial,Zap/g4/444Zap/g4/448|20|t Heres the output, -- Executing Dial([EMAIL PROTECTED]/1, Zap/g4/444Zap/g4/448|20|t) in new stack -- Called g4/444 -- Called g4/446 -- Zap/1-1 answered [EMAIL PROTECTED]/1 -- Hungup 'Zap/2-1' the above, Zap/1-1 answered [EMAIL PROTECTED]/1 line comes as soon as that Zap/1-1 line starts ringing, while the Zap/2-1 is hungup. in our zapata conf, 'callprogress=yes' is commented out. any idea why is this happening? Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Start
David J Carter wrote: I have put ./var/sbin/safe_asterisk in the rc.local file but it still doesn't start. Have you got the zaptel drivers loading at startup? This can either be done by using modprobe commands in the rc.local or by using the init script that comes with the zaptel source.. just run make config from /usr/src/zaptel (AFAIK only for redhat).. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Start
Yup, I did modprobe when I loaded everything. If I issue the reboot command then I see the zaptel being unloaded. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: 18 October 2003 12:11 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto Start David J Carter wrote: I have put ./var/sbin/safe_asterisk in the rc.local file but it still doesn't start. Have you got the zaptel drivers loading at startup? This can either be done by using modprobe commands in the rc.local or by using the init script that comes with the zaptel source.. just run make config from /usr/src/zaptel (AFAIK only for redhat).. Later.. ___ 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] IAX Clients not connecting
On Wed, 15 Oct 2003 13:49:23 -0500 (CDT), Dave Weis wrote
On Wed, 15 Oct 2003, M.A. Ali wrote:
I am kind of new to asterisk. Here is a little prolem that I am
facing.
Here is my problem and questions: I am just adding two gnophone
users to
my dialplan, all three systems are within lan.
1. in iax.conf:
[mako]
type=friend
auth=pliantext
Was this copied and pasted or mistyped?
secret=myown
context=default
host=dynamic
permit=0.0.0.0/0.0.0.0
dave
on a side note, have you tried with auth=md5?
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Re: [Asterisk-Users] x-lite
I did this, but ok, I'll try again If your box is behind NAT you need to tell xlite NOT to detect it -- I had this problem where the * box and the xlite box were behind NAT and NAT was not needed for xlite to talk to *, but xlite decided that it was so was sending the wrong IP. Everything else that WipeOut wrote is 100% correct; that is all that is necessary for xlite to talk to *. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] my asterisk experience (long)
I thought I'd post my experiences for the benefit of anyone else who may be at the point I was when I first started with asterisk. I have 2 incoming analog lines (north eastern U.S., Verizon) where one is set to ring if the first is busy. I bought a bare-bones system from abs-pc with the following components: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) I also bought 2 X100P's and 1 TDM400P from Digium, and installed them in the above system. I installed RedHat 9 onto the PC. During the RH install, I selected the server install, and tried to weed out most of the packages that I didn't need. I'm no Linux expert, but I didn't want a lot of stuff running on my server. IMO simple is better (and more secure). Along these same lines, I ran the RH command 'setup' and turned off all of the services that I didn't need. I would do the same with the kernel, but I'm not that Linux savvy yet. Setting up Linux, installing Asterisk, and writing some basic conf files took about 2 weeks in my spare time. Most of that time was spent learning about asterisk, and what I needed to include in my conf files. My initial conf files were mostly adaptations of others that I found around on the net. I bought two radioshack single line phones (one was cordless), plugged them into the TDM400P. After getting the drivers loaded, and asterisk running, I ran into my first problem. I've covered this problem extensively in earlier posts (subject: TDM400P??), so I will just briefly mention it here. The Pro-SLIC modules were resetting on hook transitions. Its like they were not getting enough power. After much debugging, and work with Digium, the problem could not be solved. I sent the card back to Digium, and they sent me a new one. The new card behaved the same way. Mark edited the driver on my machine to prevent the module reset from crashing the wcfxo driver, but the problem was not solved. Eventually I came to accept that the card simply did not work with my motherboard, an ASUS A7n8X-Deluxe. Digium refunded my money for the card, and I returned to the drawing board. I bought a Grandstream 101, then I bought 2 more. I also got a Cisco ATA186. I had looked into using the ATA186 with asterisk, and it looked like I could get it to work. When I got it, I realized that It didn't have the same firmware as I thought it would. In fact, as it was, I couldn't get it to work with asterisk at all. I tried to get a firmware update from the Cisco website. Their website is ridiculously complex and annoying. In the end, though the web site didn't tell me this explicitly, I found that they would not let me download a firmware upgrade. Luckily I was able successfully navigate their huge and annoying phone system to reach an engineer who was nice enough to email me the SIP firmware upgrade as a courtesy. After I loaded that firmware the Cisco ATA186 has worked good. The motherboard I am using has 2 Ethernet ports, but RH9 only recognizes one. I downloaded a Linux driver from NVIDIA, and had to manually edit the /etc/sysconfig files; redhat's config menus can't handle 2 Ethernet ports apparently. I set a DHCP server to run on the second Ethernet port, and also set up a NTP server for the grandstream phones' time display. I did not set up a route between the two ports. This gives me a separate isolated network for my BT-101's and the ATA186. I recorded audio using a regular PC mic, and Goldwave. Goldwave is nice as it lets you edit wav files, equalizing volumes, and applying filters. I converted the files from wav to gsm using Sox. After I got all of this set up I began testing after-hours in the office. The echo problem immediately became obvious. Everything else seemed to work good. I set the grandstream phones to use SIP-info for signaling, and spent some time massaging my conf files. After activating the Aggressive Suppressor option in the zaptel makefile, and recompiling the zaptel driver, the echo problem was greatly reduced on all but one grandstream phone. I noticed that one phone had older firmware. I set up a tftp server, and updated the BT-101's firmware. The firmware upgrade seemed to fix the remaining echo on that one phone. The echo is still audible as the occasional chirp or crackle, but it is now at a tolerable level. There is the additional problem of regular speech audio occasionally getting suppressed when both parties start talking at the same time. That is not a bad problem as it doesn't happen often, and quickly fixes itself. The current system is working good, except for the above mentioned problems with audio. The Grandstream phones' function buttons integrate nicely with asterisk. All of them seem to work. I loaded some nice (and hopefully legal) tunes
[Asterisk-Users] We have added an Asterisk Forums to our existing web site.
Title: Message We have added an Asterisk Forums to our existing web site. It will make things easier to search for related problems, etc. http://www.pbxtech.info/forumdisplay.php?f=113
Re: [Asterisk-Users] AGI script question
On Sat, 2003-10-18 at 05:23, Jim Paraschou wrote: Hi, I am trying to write an AGI script that executes a shell command in C ie. ls. I tried VERBOSE AGI command or to send the !command ls to stderr but the command does not execute it just displays on asterisk console. Does anybody has any idea about it? Well using the proper command is important. *CLI show Agi verbose Usage: VERBOSE message level Sends message to the console via verbose message system. level is the the verbose level (1-4) Always returns 1 What about that made you think a command would be run? *CLI show agi exec Usage: EXEC application options Executes application with given options. Returns whatever the application returns, or -2 on failure to find application From this you might exec a system ls command, but I doubt this will do what you want. So in the interest of getting to what you want to do. Why do you need to do a ls, and do you plan on doing anything with the output of the command? If you plan to do anything with the output of the command, then you need to ***NOT*** make asterisk do the work via agi. You would probably need to do this as a local to you application function. Perl has opendir, readdir, and closedir. I bet there is similar things in C. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing call to IVR not being answered
I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the End Call soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't picked up yet. Here are a couple examples of IVRs that elicit this behavior: 800 327 2177 800 433 7300 Thanks, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk
Hi, We would be interested in this project also. Paulo Mannheimer wrote: Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ 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] RE: Outgoing call to IVR not being answered
Ok after find this post http://www.mail-archive.com/[EMAIL PROTECTED]/msg07004.htm l. I have figured out my issue and I have refined my question. Does anyone how to make asterisk tell the calling sip-ua that the remote party answered the phone as soon as it sees rtp coming in even though the remote IVR didn't connect the call. I found a couple of remote IVRs that when I call them they don't connect the call until the caller chooses an option via DTMF... but my cisco phone won't send DTMF until asterisk tells it the call has been answered Its american express (800 327 2177) and american airlines (800 433 7300)... i think they are trying to save money by not connecting the call. Thanks, David I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the End Call soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't picked up yet. Here are a couple examples of IVRs that elicit this behavior: 800 327 2177 800 433 7300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] latest cvs update
ok, ive just updated a server and now im getting these messages a lot on the console: -- Called g1/X WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to forward voice WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to forward voice WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to forward voice WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to forward voice any ideas what the problem is? i havent change the configurations or setup of anything just upgraded from the cvs duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] my asterisk experience (long)
I bought a Grandstream 101, then I bought 2 more. I also got a Cisco ATA186. I had looked into using the ATA186 with asterisk, and it looked like I could get it to work. When I got it, I realized that It didn't have the same firmware as I thought it would. In fact, as it was, I couldn't get it to work with asterisk at all. I tried to get a firmware update from the Cisco website. Their website is ridiculously complex and annoying. In the end, though the web site didn't tell me this explicitly, I found that they would not let me download a firmware upgrade. Luckily I was able successfully navigate their huge and annoying phone system to reach an engineer who was nice enough to email me the SIP firmware upgrade as a courtesy. After I loaded that firmware the Cisco ATA186 has worked good. How does the CISCO ATA sound quality, functionality and stability compares to the Grandstream phones? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adtran TA750 T100P
I crimped one using cat 5 cable since this is just a bench-test unit. Thanks for the link, though. Turns out this question is answered in the FAQ, but it's certainly much easier to crimp one using Jared's pictures than the terse pin-out in the answer. asterisk # ztcfg -vvv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Slaves: 16) Channel 17: FXS Kewlstart (Default) (Slaves: 17) Channel 18: FXS Kewlstart (Default) (Slaves: 18) Channel 19: FXS Kewlstart (Default) (Slaves: 19) Channel 20: FXS Kewlstart (Default) (Slaves: 20) Channel 21: FXS Kewlstart (Default) (Slaves: 21) Channel 22: FXS Kewlstart (Default) (Slaves: 22) Channel 23: FXS Kewlstart (Default) (Slaves: 23) Channel 24: FXS Kewlstart (Default) (Slaves: 24) 24 channels configured. And green lights all 'round. Thanks all. Ken Godee wrote: Jose Quinteiro wrote: Hello, So all the pieces are finally here, and I'm ready to play. I remember reading on this list that the connection Channel Bank - T100P requires a reverse cable. Is this a regular Ethernet reverse cable (i.e., only a couple of pairs reversed?) Please help me before I blow something up! Saludos, Jose. Here's a link to Adtran's site w/discrip for pin outs for loopback adapters and T1 crossover. http://www.adtran.com/adtranpx/Doc/0/BIAU1PH6DJBH39S2038BE81ID8/CU-94a6a9d76bfc11d78ff20c045003.html Here's a site that sells premade T1 cables by the foot. T1 cables should be 22awg solid, and each pair individually shielded. They also include shielded RJ connectors on there cables as well. $16.00 + .70 per foot, good cables. http://www.stonewallcable.com/product.asp?dept%5Fid=134pf%5Fid=SC%2D9598%2DX+++ ___ 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] latest cvs update
On Sat, 2003-10-18 at 15:33, duncan wrote: ok, ive just updated a server and now im getting these messages a lot on the console: Use cvs update -D date to rollback your source to the date you want then check again to see if its new cvs problem. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Outgoing call to IVR not being answered
On Sat, 2003-10-18 at 15:17, David Harris wrote: Does anyone how to make asterisk tell the calling sip-ua that the remote party answered the phone as soon as it sees rtp coming in even though the remote IVR didn't connect the call. I found a couple of remote IVRs that when I call them they don't connect the call until the caller chooses an option via DTMF... but my cisco phone won't send DTMF until asterisk tells it the call has been answered Its american express (800 327 2177) and american airlines (800 433 7300)... i think they are trying to save money by not connecting the call. I see the same problem, Im trunking 800 through IAXTEL. When I dial American I only see the end call button only but DTMF works; after choosing an option I get the rest of the buttons. So I think at least the DTMF problem can be solved. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID line with Adtran TA750 and T100p
Hello, I new to this, but with the help of mailing lists archives and IRC I am able to build my PBX. Thanks to all who had help me to reach till here. I am stuck at a point where I can't find the solution on mailing lists or even on IRC. I have individual 4 DID (Direct Inward Line) coming from Telco and terminating into TA 750 to FXS card. Many of them told that Phone instrument terminates to FXS card that is correct. When I check with Adtran Tech they told it should be terminated into FXS card, with DPO mode on each DID circuit. I checked this on there website also here it is: How do I extend a DID (Direct Inward Dial) line from the telephone company using voice FXO and FXS cards? This connection seems backwards when compared to the OPX line. Remember with a DID line, the telco acts like the switch (FXO) and the customer supplies the battery (FXS). The customer connects the telco DID line to our FXS card and the DID trunk of the PBX to the FXO card. These voice lines originate from telco and terminate into the PBX. They will never originate from the PBX. When a call comes into the telco's switch with your telephone number, the telco closes a switch connected to your cable pair. This causes loop current to flow from the FXS card. The FXS card sends signal bits across the T1 to the FXO card who then closes his switch causing loop current to flow from the DID interface card on the PBX. The PBX then signals the telco (with a wink) that it is ready for the call. The PBX does this by reversing the battery's polarity. When the Telco sees this wink, the Telco then passes the DNIS digits through the talk path into the PBX. The PBX uses the DNIS digits to route the call to the appropriate phone. The call can terminate from either end. If the person at the PBX hangs up the loop current (from the PBX to the telco) will stop flowing and Telco will return to an idle condition by opening their switch. The call can also be terminated from the telco side if the incoming caller hangs up. When this happens, the telco opens their switch and loop current stops flowing. The PBX then returns to an idle condition. My PBX communicates with TA750 on T1 with T100p card. I don't know what signaling needs to be set in Zaptel.conf and Zapata.conf files, so that once Telco send the signal it will see the PBX and rings that DID extensions. Has anyone had done this? Please let me know, I have to put this machine live by Monday and I stuck where I have no clue what to do. Thanks Regards, KD ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Outgoing call to IVR not being answered
Juan, That is a helpful discovery. If I too route the calls out of an IAX provider (voicepulse) rather then my usual Cisco SIP Gateway I can get DTMF through with the Cisco phone still not presenting the usual soft key options. So that means that the Cisco IP Phone IS sending DTMF to calls that are not yet connected and that either the asterisk box or the Cisco SIP Gateway are the ones not transmitting. Thanks, David I see the same problem, I´m trunking 800 through IAXTEL. When I dial American I only see the end call button only but DTMF works; after choosing an option I get the rest of the buttons. So I think at least the DTMF problem can be solved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oh323 cisco callamanager
Title: Message hi , i'm testing asterisk like and Automatic attendant with a callmanager and vg200 gateway with 1 t1 everithing works finw but some times asterisk didnt not disconnect calls and star growing the number of connections from asterisk to callmanager , and when this connections get to 35 g711 , the asterisk hang. some one , ?? i'm using asterisk-0.5.0 and oh323 5.5 regards , victor medrano
RE: [Asterisk-Users] RE: Are Cisco 7960 SIP versions 4 5 safe to use?
Howdy, Does anyone know if there are any problems running Asterisk when using later 7960 SIP versions like 04.04 or 05.03? Thanks! Paul Paul Mahler [EMAIL PROTECTED] phone: 650-207-9855 fax: 877-408-0105 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Creating new voicemail accounts
I have googled this one to death, and can't find anything. I added a number of new users to my asterisk (current CVS) system. I am using the Voicemail2 family. I added entries in extensions.conf and voicemail.conf for my new users, and I have tested leaving and retrieving new voicemails for them. All of this works fine. But if one of the new users tries to Administer personal greetings (or whatever exactly the menu choice is called) once they get to the point of recording, asterisk bombs them off the call without writing anything. Here's the CLI interaction: -- Playing 'vm-messages' -- Playing 'vm-opts' -- Playing 'vm-options' -- Playing 'vm-rec-unv' -- Playing 'beep' -- x=0, open writing: voicemail/default/5120/unavail format: gsm, (nil) == Spawn extension (home, 8, 1) exited non-zero on '[EMAIL PROTECTED]:4569]/2' Actual voicemail messages seem to be being saved in all the various (gsm, wav) formats, so the recording subsystem must have all its components. I'm sure I must be overlooking something obvious. Can anyone provide a pointer? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTA310 Config
I have been trying to get the DTA310 to work properly with Asterisk for the last week. It seems to connect but it does not play back any sound and I cannot dial it by using x-lite. Sip debug looks pretty good. I was wondering if someone has a working config that they could post so that i could see what I am doing wrong. Thanks Buddy Edwards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] my asterisk experience (long)
How does the CISCO ATA sound quality, functionality and stability compares to the Grandstream phones? Sound Quality using G.729: Grandstream phones are superior. They sound perfect even with slight packet loss. The ATA will sound very good with 0% packet loss but if you ramp it up a bit it degrades quite badly. Stability: This is where the ATA shines. It has proved very stable 99% of the time. Very few times have we found one that needs to be reset. On the other hand the Grandstreams do stop registering every now and then and there is also an issue with STUN and changing IP Addresses at the client side. Neverthless the Grandstream folks are quite on top of things and they assure us these issues are being fixed. We are quite happy with the Grandstream phones and if these issues get resolved and the ATA286 becomes available soon, we might just stop buying the Cisco ATA. ___ 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?
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. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Feedback request: AGI GET DATA change - termination digits
** 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
RE: [Asterisk-Users] Auto Start
Hi, Just back from the Rugby match, (we won), and am now trying again to get asterisk to auto start without being there when the server reboots. Wipe Out asked: Have you got the zaptel drivers loading at startup? This can either be done by using modprobe commands in the rc.local or by using the init script that comes with the zaptel source.. just run make config from /usr/src/zaptel (AFAIK only for redhat).. I can see them unload when I issue the reboot command. Just before the Logon prompt appears on boot I see a message but it goes before I can read it and the prompt to logon appears. If I scroll back up the screen the message is not there. I now have only ./safe_asterisk in my rc.local file. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: 18 October 2003 12:11 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto Start David J Carter wrote: I have put ./var/sbin/safe_asterisk in the rc.local file but it still doesn't start. Have you got the zaptel drivers loading at startup? This can either be done by using modprobe commands in the rc.local or by using the init script that comes with the zaptel source.. just run make config from /usr/src/zaptel (AFAIK only for redhat).. Later.. ___ 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] Auto Start
Dave, After the system boots up, you check to see which modules are loaded by doing a lsmod. zaptel and others that you need should be listed. If not you can manually add modprobe zaptel and the other drivers into your rc.local file. Paul Just before the Logon prompt appears on boot I see a message but it goes before I can read it and the prompt to logon appears. If I scroll back up the screen the message is not there. I now have only ./safe_asterisk in my rc.local file. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Creating new voicemail accounts
WipeOut wrote: If you are using VM2 and using something other than the [default] context then you will not create the correct directory structures when you run the addmailbox script to create the mailboxes.. I have attached a copy of my addmailbox2 script which takes into account the context when creating the voice mailbox directory structure.. Hope it helps you promlem.. Well it creates the mailboxes just fine. I have discovered the problem in the messages logfile. Somehow this invocation of asterisk is trying to put the voicemail in the wrong place: Oct 18 18:44:10 WARNING[262161]: File file.c, Line 769 (ast_writefile): Unable to open file /var/lib/asterisk/sounds/voicemail/default/5120/busy.gsm: No such file or directory I'm assuming asterisk must figure that out dynamically when it is started up, because the binaries and modules, etc., are the same for several incarnations of asterisk that I run, and only this one seems to be looking in the wrong directory for the voicemail stuff. I've grepped myself blind, and can't yet figure out how that's getting set wrong. Just for fun, I symlinked /var/spool/asterisk/voicemail to /var/lib/asterisk/sounds/voicemail, and things work just fine. What did I hose, I wonder, and especially, how?? Thanks. B. ___ 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
** 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 I can't comment on the usefulness of creating this additional syntax in an AGI (though it does seem like a useful shorthand method to recognize terminators and replay digits) but some of the same comments have been discussed in the bugtracker for dialplan routines. see: http://bugs.digium.com/bug_view_page.php?bug_id=181 JT ___ 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
While that change is fine, you could also just write the same functionality with get digit and deal with it inside the AGI app. I guess the digit handling part is doable with WAIT FOR DIGIT.. I was going to ask about interupting the prompt that played before that but on RTFM-ing I see that STREAM FILE returns the interrupting digit if interruptions are allowed. Sooo... Knock myself up a subroutine in Perl to handle flexible digit termination as I want.. or have a look at patching the GET DATA function in AGI, potentially making easier for others whilst not breaking anything that currently exists. Would bug tracker be the way to go for further discussion? Or I could go with the second option, stick it in there, and see if it makes it to CVS? Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny XML Files for 7920
More appropriately, Skinny is still considered beta quality in Asterisk. Patches are welcome and encouraged. Jeremy McNamara Tomica Crnek wrote: Skinny is not well supported by Asterisk. This was the answer when I asked the same question few days ago. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 18, 2003 4:42 AM Subject: [Asterisk-Users] chan_skinny XML Files for 7920 Hi, I have a Cisco 7920 that I'm trying to get working with my * box. When the phone boots it requests XMLDefault.cnf.xml and SEPMACADDRESSHERE.cnf. I assume I set the line number, etc in the latter of the two. However I cannot find any reference to how this file is structured. Anyone know? I assume this is why I'm getting the errors below: Oct 17 19:47:24 WARNING[1357974832]: File chan_skinny.c, Line 1807 (handle_message): Client sent message #0 without first registering. Oct 17 19:47:24 ERROR[1357974832]: File chan_skinny.c, Line 1823 (handle_message): Rejecting Device SEP000D282E95F0: Device not found Oct 17 19:47:24 WARNING[1357974832]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 17 19:47:24 NOTICE[1357974832]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Connection reset by peer Thanks, Justin ___ 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] 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
RE: [Asterisk-Users] Feedback request: AGI GET DATA change -termination digits
While that change is fine, you could also just write the same functionality with get digit and deal with it inside the AGI app. I guess the digit handling part is doable with WAIT FOR DIGIT.. I was going to ask about interupting the prompt that played before that but on RTFM-ing I see that STREAM FILE returns the interrupting digit if interruptions are allowed. Sooo... Knock myself up a subroutine in Perl to handle flexible digit termination as I want.. or have a look at patching the GET DATA function in AGI, potentially making easier for others whilst not breaking anything that currently exists. Would bug tracker be the way to go for further discussion? Or I could go with the second option, stick it in there, and see if it makes it to CVS? Cheers Paul I would suggest that if you pursue the second path (patching GET DATA) that you start a new feature request in the bugtracker, since the original item (#181) was about a request to program this into a new application for the dialplan, not AGI. Just because something can be done one way with Asterisk doesn't meant that a shorter/better/different way of doing it won't be well received if it is significantly easier and makes the life of the scripter/dialplanner less complex. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
