[Asterisk-Users] g729 Codec
Hi, Dotheg729 codec licensesfor Asterisk workon aSIP environment (only SIP UAs running g729+ Asterisk)? I would liketo buy a couple for a SIP test labbut I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa http://www.telesip.net
[Asterisk-Users] Nortel 350
Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are generic, and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a stupid one. Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time and date stamp in voicemail
Mark: Will you add this to cvs? On Sun, 2003-07-27 at 04:06, Andy Hester wrote: Tilghman, I applied your voicemail_prompts patch and it works like a charm. Thanks for donating the code and thanks to those that donated the voice prompts! Another win for Asterisk Sincerely, Andy Hester Consero -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher Sent: Friday, July 25, 2003 10:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] time and date stamp in voicemail On Friday 25 July 2003 14:12, Andy Hester wrote: Dan, the page is actually http://asterisk.drunkcoder.com/patches/ . However, I didn't see the patch there. I just added it. It's available there now. Note that there are three files: a patch, sounds, and some instructions. -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
[Asterisk-Users] * behind ISDN pbx - Forwarding to extensions with in primary pbx
Hi, I have asterisk behind my primary PBX connected via ISDN (chan_capi). Calling out and calling in works just fine, however I can't connect to my primary pbxs' extensions. If anyone has an example extensions.conf, I'd be grateful for a copy. I tried (the MSN of the ISDN card is set to 30) exten = 22,1,Dial,CAPI/30:22 but this does not work. Changing it to exten = 22,1,Dial,CAPI/30:12345 will call the outside number 12345. Any and all help is greatly appreciated. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD-gateway prefix
Hey all, As there seem to be some problems with DTMF-signalling between chan_sip and several clients, due to which many could not properly dial a number at the dial-tone of the XS4ALL-gateway at FWD-number 42442, I've now arranged for a prefix on FWD for this gateway. From FWD, you can now dial 1010-666, followed by the Dutch toll-free number or IAXTel-number you wish to reach, as you would have dialled it from the dial-tone at FWD-number 42442. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Banks
What channel banks are best supported by asterisk, and available new at preferably decent prices?? It would seem that for small offices with less than 15 users, a single port T1 card with a channel bank, with say 15 FXS and 9 FXO (or similar config) would be ideal. So I would like to find channel banks that suit that sort of environment Also, I suppose you would also want channel banks for larger offices, generally as 100% FXS ports and using digital trunks for FXO (ie, pri). Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get musiconhold to work
So you mean a just simple blank line at the end of the musiconhold.conf file or the extensions.conf file? Second question, though it might seem a bit stupid, do I perhaps need a sound card on the box that asterisk is running on? I don't think this should be the case but I'm just wondering. Is there anything I can do to manually make it run with asterisk? I guess what I'm trying to say here is in ps aux I see no example of mpg123 running that tells me it has not been executed. What is the process that asterisk uses to execute it? Is it executed each time a caller is put on hold or are instances started in the background when asterisk begins (listen state)? AJ On Sat, 26 Jul 2003, WipeOut . wrote: Only things I can suggest is.. 1. Execute it from a command line and make sure it runs.. If not you may hevr to compile it from source.. 2. Make sure you have a new line at the end of your .conf file cos * often freaks out about that.. Other than that I don't know why its not working for you.. No instances of it running when I look at processes. AJ On Sat, 26 Jul 2003, WipeOut . wrote: Sorry I though you had compiled from source... When * is running do ps-aux | grep mpg123 and make sure it is actually running.. Later.. Wipeout I'm using the exact mpg123 binary that you sent me. When I execute a whereis mpg123 it returns /usr/bin. To take it a step further I've done whereis mpg321 and rpm -q mpg321 just to make sure mpg321 is not on the system. The one thing that's confusing the heck out of me is the fact that the rpm that I installed seems to have installed in /usr/bin whereas everybody else's installed in /usr/local/bin. Any other ideas? I'm growing very frustrated. AJ On Sat, 26 Jul 2003, WipeOut . wrote: IIRC I had the same problem becasue the package will install the mpg123 binary to /usr/local/bin and * seems to look in /usr/bin so just copy the mpg123 executable to /usr/bin and it should work.. Later.. I can't seem to get musiconhold to work. I'm running asterisk on a RH9 box, I have the mpg123 package installed. In my zapata.conf file I have the line MusicOnHold=default . In my musiconhold.conf file, in the classes section I uncommented default and loud. In my extensions.conf file I have a set musiconhold line. However if I get a call and I either put it on hold or hit flash I get no music. The sample mp3 file is in the mohmp3 directory. Does anyone know what I might be doing wrong or how I might be able to correct it? Also I have tried assigning a extension with the MusicOnHold application and it still doesn't seem to work. AJ ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P startup
I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A red flashing light circles around the 4 RJ48C sockets. I load the wct4xxp driver, and the flashing light stops. Whether I connect an E1 signal or not, no lights are shown, and no alarms are reports in the /proc/zaptel/XXX files. What is supposed to happen? I expected all the ports to show a continuous red LED until I plugged in an E1, and then to go green. That is what the other Digium cards do. First, if you want to run in E1 mode you either need to put jumpers on the T1/E1 select jumpers located just south-west of the middle of the board (and labeled). Place a jumper on each span you want to be E1. Then you'll need to edit your /etc/zaptel.conf and make sure you have E1 spans defined. Finally, if the lights don't go red, then you might need to manually run ztcfg. If all else fails, be *sure* you're running latest CVS :) Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P startup
we have now perfect results with yesterdays cvs and the te410p todays cvs allways thinks that immediate is set to yes in zapata.conf. weird ... cheers Michael On Sunday 27 July 2003 7:12 pm, Mark Spencer wrote: I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A red flashing light circles around the 4 RJ48C sockets. I load the wct4xxp driver, and the flashing light stops. Whether I connect an E1 signal or not, no lights are shown, and no alarms are reports in the /proc/zaptel/XXX files. What is supposed to happen? I expected all the ports to show a continuous red LED until I plugged in an E1, and then to go green. That is what the other Digium cards do. First, if you want to run in E1 mode you either need to put jumpers on the T1/E1 select jumpers located just south-west of the middle of the board (and labeled). Place a jumper on each span you want to be E1. Then you'll need to edit your /etc/zaptel.conf and make sure you have E1 spans defined. Finally, if the lights don't go red, then you might need to manually run ztcfg. If all else fails, be *sure* you're running latest CVS :) Mark ___ 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] TE410P startup
OK Funny guy, Mark Spencer wrote: I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A red flashing light circles around the 4 RJ48C sockets. I load the wct4xxp driver, and the flashing light stops. Whether I connect an E1 signal or not, no lights are shown, and no alarms are reports in the /proc/zaptel/XXX files. What is supposed to happen? I expected all the ports to show a continuous red LED until I plugged in an E1, and then to go green. That is what the other Digium cards do. First, if you want to run in E1 mode you either need to put jumpers on the T1/E1 select jumpers located just south-west of the middle of the board (and labeled). Place a jumper on each span you want to be E1. We know this Then you'll need to edit your /etc/zaptel.conf and make sure you have E1 spans defined. ... and we how to do this Finally, if the lights don't go red, then you might need to manually run ztcfg. and with a mix of cards we always have to do this. If all else fails, be *sure* you're running latest CVS :) but this is *highly* relevant. Gee I wasted ages experimenting with this today. Now I get throbbing red lights! Attach an E1, and even get green lights. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCM Voice Quality Issue on CVS Version
Hi, You can try to apply Michael's patch... for me it works perfect. BR, Dan - Original Message - From: Ricardo Villa [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 26, 2003 5:47 PM Subject: [Asterisk-Users] PCM Voice Quality Issue on CVS Version Hi, I have asterisk-0.4.0 running. When I make a call between an ATA186 and Asterisk using ulaw or alaw codec, all is fine. I installed the CVS version and tried the same thing but the voice is choppy. The installation was done on the same linux server. The stats on the ATA186 show no packet loss but a great number of late packets. The stats when running version *-.0.4.0 do not show late packets. I have gone back and forth installing each version again with the same results. Bandwidth is not an issue as the ATA and * are on the same LAN. Is there something that can be fixed on the CVS version to prevent this problem? Thanks, Ricardo VIlla http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCM Voice Quality Issue on CVS Version
Hi, I tried todays CVS and it works fine now. Thanks, - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 27, 2003 12:40 PM Subject: Re: [Asterisk-Users] PCM Voice Quality Issue on CVS Version Hi, You can try to apply Michael's patch... for me it works perfect. BR, Dan - Original Message - From: Ricardo Villa [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 26, 2003 5:47 PM Subject: [Asterisk-Users] PCM Voice Quality Issue on CVS Version Hi, I have asterisk-0.4.0 running. When I make a call between an ATA186 and Asterisk using ulaw or alaw codec, all is fine. I installed the CVS version and tried the same thing but the voice is choppy. The installation was done on the same linux server. The stats on the ATA186 show no packet loss but a great number of late packets. The stats when running version *-.0.4.0 do not show late packets. I have gone back and forth installing each version again with the same results. Bandwidth is not an issue as the ATA and * are on the same LAN. Is there something that can be fixed on the CVS version to prevent this problem? Thanks, Ricardo VIlla http://www.telesip.net ___ 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] can't get musiconhold to work
Yes always end your conf files with blank lines otherwise you may get weird results from asterisk.. as for the sond card requirement I don't know all my systems have onboard sound.. So you mean a just simple blank line at the end of the musiconhold.conf file or the extensions.conf file? Second question, though it might seem a bit stupid, do I perhaps need a sound card on the box that asterisk is running on? I don't think this should be the case but I'm just wondering. Is there anything I can do to manually make it run with asterisk? I guess what I'm trying to say here is in ps aux I see no example of mpg123 running that tells me it has not been executed. What is the process that asterisk uses to execute it? Is it executed each time a caller is put on hold or are instances started in the background when asterisk begins (listen state)? AJ On Sat, 26 Jul 2003, WipeOut . wrote: Only things I can suggest is.. 1. Execute it from a command line and make sure it runs.. If not you may hevr to compile it from source.. 2. Make sure you have a new line at the end of your .conf file cos * often freaks out about that.. Other than that I don't know why its not working for you.. No instances of it running when I look at processes. AJ On Sat, 26 Jul 2003, WipeOut . wrote: Sorry I though you had compiled from source... When * is running do ps-aux | grep mpg123 and make sure it is actually running.. Later.. Wipeout I'm using the exact mpg123 binary that you sent me. When I execute a whereis mpg123 it returns /usr/bin. To take it a step further I've done whereis mpg321 and rpm -q mpg321 just to make sure mpg321 is not on the system. The one thing that's confusing the heck out of me is the fact that the rpm that I installed seems to have installed in /usr/bin whereas everybody else's installed in /usr/local/bin. Any other ideas? I'm growing very frustrated. AJ On Sat, 26 Jul 2003, WipeOut . wrote: IIRC I had the same problem becasue the package will install the mpg123 binary to /usr/local/bin and * seems to look in /usr/bin so just copy the mpg123 executable to /usr/bin and it should work.. Later.. I can't seem to get musiconhold to work. I'm running asterisk on a RH9 box, I have the mpg123 package installed. In my zapata.conf file I have the line MusicOnHold=default . In my musiconhold.conf file, in the classes section I uncommented default and loud. In my extensions.conf file I have a set musiconhold line. However if I get a call and I either put it on hold or hit flash I get no music. The sample mp3 file is in the mohmp3 directory. Does anyone know what I might be doing wrong or how I might be able to correct it? Also I have tried assigning a extension with the MusicOnHold application and it still doesn't seem to work. AJ ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ordering digital trunks?
OK, this is probably a dumb question for a lot of you, but I have no experience with digital lines outside of a tiny bit of ISDN, so I'll just bite the bullet and ask some newbie questions. I am attempting to plan an asterisk installation with about 20 SIP phones and the following incoming lines: 1) At least 6 (as many as 10) lines for voice to the SIP phones 2) 2 incoming/outgoing fax 3) 4-10 lines for an IVR application I feel like a T1 with 24 channels should suffice, but what exactly do I order and what to I have to have in my asterisk unit to interface? Does the line they terminate just plug into a T100P or do I need some extra hardware? What services do I need to be sure I order on the T1? Is there a way using the T100P to dynamically allocate unused voice channels for data? Finally, how do I get the best fax performance for two analog fax machines out of this setup? Will an ATA-186 serve, or will I get much better performance using a TDM400P with zaptel bridging? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival talks fast...
Ok I have festival on RH8. It speaks fast and you can't understand it. I don't have any FXO cards in this box yet. Can someone shed some light on this? Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ordering digital trunks?
On Sun, 2003-07-27 at 13:29, John Laur wrote: OK, this is probably a dumb question for a lot of you, but I have no experience with digital lines outside of a tiny bit of ISDN, so I'll just bite the bullet and ask some newbie questions. I am attempting to plan an asterisk installation with about 20 SIP phones and the following incoming lines: 1) At least 6 (as many as 10) lines for voice to the SIP phones 2) 2 incoming/outgoing fax 3) 4-10 lines for an IVR application I feel like a T1 with 24 channels should suffice, but what exactly do I order and what to I have to have in my asterisk unit to interface? Does the line they terminate just plug into a T100P or do I need some extra hardware? What services do I need to be sure I order on the T1? Is there a way using the T100P to dynamically allocate unused voice channels for data? There is config options for this, but you will want to have a PRI to do this. A PRI only gives you 23 channels to work with. This will also help make sure your FAX lines are clean as can be. You will want at least a block of DID numbers. With your question about dynamic allocation of empty channels for data, you would have to find someone who will let you dial in and bond up channels. Then you need to calculate the estimated number of lines used a any time, add in a bit for the idle lines so new calls come in, and your left with the number of possible 64k channels for data. To take your low estimates, 6+4+1 = 11 + 3 for spares and you have 14. 23-14 = 9*64kbit. So your low usage, high speed bandwidth is ~576kbit. If you don't like this speed, you really need to think about maybe a DSL line to augment or be your bandwidth. Finally, how do I get the best fax performance for two analog fax machines out of this setup? Will an ATA-186 serve, or will I get much better performance using a TDM400P with zaptel bridging? The TDM400P would be a good choice, and price isn't too much different. Not to mention that it will help making sure your T1 is up as it is easier to configure. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Fritz RedHat 8.0
Title: Message Has anyone got the BT Speedway (AVM Fritz) card working on a RedHat 8.0 system with *. If so could someone give me some pointers on getting the right sequence of installingthe drivers and which versions to use. Thanks, Stuart
RE: [Asterisk-Users] Nortel 350
On Sunday, July 27, 2003 3:32 AM, Brian Capouch [SMTP:[EMAIL PROTECTED] wrote: Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are generic, and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. generic may not mean much. Connecting a generic ADSI phone to a locking ADSI script would change it to a LOCKED ADSI phone. So even though it may not be branded, it still could be locked. The important question is, has the phone been locked. By the way, I would like to find a vendor for unlocked ADSI phones. To this point it has seemed like hit or miss whether a phone will be locked. Any recommended vendors for ADSI phones? Don Pobanz Of course I am green as could be on this topic, so this question may even be a stupid one. Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Language
Hi, I am in the process of recording voicemail prompts in german. How do I specify the language for the voice mail messages? I want to offer both language files, based on the calling party. Any and all help is greatly appreciated. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN Fritz RedHat 8.0
Assuming it is a suitable Fritz card your best bet is to get the CAPI library/driver from AVM and then check this out http://www.junghanns.net/asterisk/ - chan_capi is reportedly the best performing ISDN channel driver for asterisk, although I personally haven't used it ;-) Iain --On Sunday, July 27, 2003 8:25 pm +0100 Stuart Hirst [EMAIL PROTECTED] wrote: Has anyone got the BT Speedway (AVM Fritz) card working on a RedHat 8.0 system with *. If so could someone give me some pointers on getting the right sequence of installing the drivers and which versions to use. Thanks, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Language
Use setlanguage. Then organize the language files by directory e.g. /var/lib/asterisk/sounds/de /var/lib/asterisk/sounds/digits/de Also, say.c will have to be modified to support German style number handling. Mark On Sun, 27 Jul 2003, Peer Oliver schmidt wrote: Hi, I am in the process of recording voicemail prompts in german. How do I specify the language for the voice mail messages? I want to offer both language files, based on the calling party. Any and all help is greatly appreciated. rgds pos ___ 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] Ordering digital trunks?
I feel like a T1 with 24 channels should suffice, but what exactly do I order and what to I have to have in my asterisk unit to interface? Does the line they terminate just plug into a T100P or do I need some extra hardware? What services do I need to be sure I order on the T1? Is there a way using the T100P to dynamically allocate unused voice channels for data? There is config options for this, but you will want to have a PRI to do this. A PRI only gives you 23 channels to work with. This will also help make sure your FAX lines are clean as can be. You will want at least a block of DID numbers. OK, that basically covers the main question. Thanks very much for the response. With your question about dynamic allocation of empty channels for data, you would have to find someone who will let you dial in and bond up channels. Then you need to calculate the estimated number of lines used a any time, add in a bit for the idle lines so new calls come in, and your left with the number of possible 64k channels for data. To take your low estimates, 6+4+1 = 11 + 3 for spares and you have 14. 23-14 = 9*64kbit. So your low usage, high speed bandwidth is ~576kbit. If you don't like this speed, you really need to think about maybe a DSL line to augment or be your bandwidth. This kind of speed is fine for my purpose. This connection would be only for emergency use when our primary (6 megabit wireless) link fails, and our ISP would allow us to use it for backup purposes only at no additional cost. Finally, how do I get the best fax performance for two analog fax machines out of this setup? Will an ATA-186 serve, or will I get much better performance using a TDM400P with zaptel bridging? The TDM400P would be a good choice, and price isn't too much different. Not to mention that it will help making sure your T1 is up as it is easier to configure. Well, I was asking because I already have some ATA-186's that I am only dinking around with that could actually be used. I guess I should test them to see how (non-T.38) faxing works via asterisk and if the performance is not good, I'll move to the TDM400P. Thanks again, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Package asterisk-oh323?
I am working on one (chan_h323). On Thu, Jul 24, 2003 at 07:50:08PM +0200, Peer Oliver schmidt wrote: Is there a Debian package available for asterisk-oh323, or the chan_h323? If yes, where might I find one? Thanks rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] * behind ISDN pbx - Forwarding to extensions with in primary pbx
Hi Peer, at my site it is working exactly as you wrote in your 1st example. How is your PBX setup? I remember that there is a way to set a pbx to spontanic trunk access. At least my Agfeo has got such a setup possibility. Try to switch this off for your ISDN Card. Cheers Andreas -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Peer Oliver schmidt Gesendet: Sonntag, 27. Juli 2003 14:52 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] * behind ISDN pbx - Forwarding to extensions with in primary pbx Hi, I have asterisk behind my primary PBX connected via ISDN (chan_capi). Calling out and calling in works just fine, however I can't connect to my primary pbxs' extensions. If anyone has an example extensions.conf, I'd be grateful for a copy. I tried (the MSN of the ISDN card is set to 30) exten = 22,1,Dial,CAPI/30:22 but this does not work. Changing it to exten = 22,1,Dial,CAPI/30:12345 will call the outside number 12345. Any and all help is greatly appreciated. rgds pos ___ 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] time and date stamp in voicemail
On Saturday 26 July 2003 21:06, Andy Hester wrote: Tilghman, I applied your voicemail_prompts patch and it works like a charm. Thanks for donating the code and thanks to those that donated the voice prompts! Another win for Asterisk Is anybody at all using the variable substitution and/or the expression logic at all? I'd like to know if and how well it works for you. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] time and date stamp in voicemail
I would also like to see a patch to ignore voicemail messages x number of seconds long.. ususally those 1-4 second voicemails are nothing anyway.. bkw On Sun, 27 Jul 2003, Tilghman Lesher wrote: On Saturday 26 July 2003 21:06, Andy Hester wrote: Tilghman, I applied your voicemail_prompts patch and it works like a charm. Thanks for donating the code and thanks to those that donated the voice prompts! Another win for Asterisk Is anybody at all using the variable substitution and/or the expression logic at all? I'd like to know if and how well it works for you. -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] time and date stamp in voicemail
Tilghman, I'm not sure how to use this logic. Would this be for something like, for example, deleting of forwarding a message that a certain age? Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher Sent: Sunday, July 27, 2003 11:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] time and date stamp in voicemail On Saturday 26 July 2003 21:06, Andy Hester wrote: Tilghman, I applied your voicemail_prompts patch and it works like a charm. Thanks for donating the code and thanks to those that donated the voice prompts! Another win for Asterisk Is anybody at all using the variable substitution and/or the expression logic at all? I'd like to know if and how well it works for you. -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] ISDN Fritz RedHat 8.0
Title: Message All you really should need is: modprobe hisax type=27 protocol=2 id=isdn0 and in modem.conf: driver=aopendriver=i4ltype=i4l; ISDN example;group=1msn=xxxdevice = /dev/ttyI0device = /dev/ttyI1 Has anyone got the BT Speedway (AVM Fritz) card working on a RedHat 8.0 system with *. If so could someone give me some pointers on getting the right sequence of installingthe drivers and which versions to use. Thanks, Stuart
