[Asterisk-Users] DTMF on Planet VIP153
Hi all. Does anybody use VIP 153 phone with asterisk and has DTMF works. Thank, Bob. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp / txfax exit codes / logging?
I'm looking for that one too. I had not been succesfull up to now. Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomasz Chmielewski Sent: Thursday, October 27, 2005 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] spandsp / txfax exit codes / logging? Is it possible to somehow read spandsp / txfax exit codes? What I mean, I never know if the fax sent through the Asterisk box was sent successfully, or not (i.e., a real person picked up the phone instead of a fax machine). A possibility of reading an exit code, or a log file would allow to build some kind of fax confirming (via email/web page/etc.). Are exit codes (or logging, or something similar) possible with spandsp / txfax? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2.6.13 zaptel incompability?
I'm using zaptel on FC4 with 2.6.13. and it works good. Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Friday, October 21, 2005 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 2.6.13 zaptel incompability? hi i heard some talk about something in zaptel is currently incompatible with 2.6.13. is this so? if so, will this be fixed soon? thanks roy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Compilation with H323 working on it
I did use it on Debian and now use it on FC4 and H323 is working good on both systems. Im using asterisk own h323 driver. Bob. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Thursday, October 20, 2005 2:24 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Compilation with H323 working on it Hi Folks, Can recomend a asterisk compilation for Mandrake or Debian that has on it H323 WORKING ? I try use H323 with Asterisk for some implementations but that cant good results. So any tip ? Thanks alot ! Carlos. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Compilation with H323 working on it
I dont use Microsoft Netmeeting. Sorry I use HW H323 devices only. AVAYA S8300 and some Planet telephones. Bob. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Thursday, October 20, 2005 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Compilation with H323 working on it Hi Did it work well with Netmeeting from Microsoft ?? Thanks for answer. Carlos. On Thu, 20 Oct 2005 14:41:38 +0200, Bohuslav Coufal wrote: I did use it on Debian and now use it on FC4 and H323 is working good on both systems. Im using asterisk own h323 driver. Bob. From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Thursday, October 20, 2005 2:24 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Compilation with H323 working on it Hi Folks, Can recomend a asterisk compilation for Mandrake or Debian that has on it H323 WORKING ? I try use H323 with Asterisk for some implementations but that cant good results. So any tip ? Thanks alot ! Carlos. Carlos Arnt Key soluçőes em Internet Av. das americas 500 bl 03 sala 204 Tel: (021) 2492-1666 Voip rede mundial: 9000 ou 9500 E-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap channel does not hangup
Hi, I have the [585228900] exten = s,1,SetCallerID(5228900) exten = s,2,Dial(H323/[EMAIL PROTECTED],20) exten = s,3,Hangup commands in the context [585228900] where zap channel come when inside call is coming. But when the call isn't answered it isn't hangup after 20 sec. What is it wrong? Thanks, bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compiling Asterisk 1.2 with zaptel and h.323
On FC4 is better to use pwlib 1.9.1 and openh323 1.17.2. I think, that OPENH3232DIR= is wrong. Better is OPENH323DIR= :-). If You use standard prefix for instalation o packages there is a better way instad copy library edit /etc/ld.co.conf and use /usr/local/lib/ as next source of shared library. Anyway, your text is very usefull. Bob. Dne pondělí 17 říjen 2005 14:55 Lenz napsal(a): Hello list, I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with a TDM400 card and H.323. You can find it at http://www.oinko.net/astrecipes/index.php?n=102 Any comment / suggestion / modification /bugfix is welcome! I was wondering: is there any way to build a version of Bristuff for 1.2 beta 1? Bye for now, l. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ooh323c and calls to pri
Does anybody has more information about internal structure of ooh323c and should tell me how can i setup startup information about transfer rate of call? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Coufal Bohuslav Sent: Monday, October 17, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ooh323c and calls to pri And the next information is that in header of call is information about transfer rate zero and should be 64k (codec ulaw). Bob. Dne pondělí 17 říjen 2005 13:36 Coufal Bohuslav napsal(a): The next information is that calls send from ooh323 to PRI has packet mode and it shall be circuit. Bob. P.S. - I did use old H323 driver form asterisk up to now and it works fine. Dne pondělí 17 říjen 2005 13:10 Coufal Bohuslav napsal(a): Hi I have a trouble with calls coming form ooh323c channels and going to PRI. This calls are rejected by telecom. Incoming calls form PRI and going to ooh323c works good. When i spoke with man on telecom thay said to me that there is wrong in something called information element. Does anybody knows if i can change some values for it or what i can do. Thanks, Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Email to FAX
I didnt try it up to now ill try it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eddie Sent: Monday, October 17, 2005 5:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Email to FAX Bob, Have you tried faxing multiple pages? I'm facing problem with multiple fax. The receiver have only received the first page of two pages I sent out. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Email to FAX
I think, that mistake is between PC and chairs. When i have not outgoing lines it's too hard to call out. Now i'm in state, that example form README dialed and i'm trying to receive fax on other side. Thanks, Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Thursday, October 13, 2005 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Email to FAX Yeah I missed that in the original, sorry bout that. are you sure that the other end didnt hang up? You may want to test this by calling a number you have access to so that you can at least rule that out. The only other thing I can think of is that txfax itself is aborting and returning prematurely. I wonder if its a negotiation failure. You say it hangs up immediatly, how immediatly? 1 second? 5? On Thu, 2005-10-13 at 11:52 +0200, Coufal Bohuslav wrote: But it seems that Asterisk understand that he has to dial (the dialed number is correct), -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) it seems that zap channel had answered (but nothing to try dial), Channel Zap/4-1 was answered. and lunching txfax Launching txfax(/tmp/ast_fax-1129191936.10240.1804289383.0|caller) on Zap/4-1 and immediately hungup -- Hungup 'Zap/4-1' May be something wrong in zapata.conf? ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [channels] ; language=us signalling=fxs_ks context=default ;context=fax channel = 3-4 Thank for any other sugestions, Bob. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Email to FAX
All works very well. Last question is if there is a chance to get result of sending by mail (for example as answer to my mail). Thanks, Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Thursday, October 13, 2005 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Email to FAX Yeah I missed that in the original, sorry bout that. are you sure that the other end didnt hang up? You may want to test this by calling a number you have access to so that you can at least rule that out. The only other thing I can think of is that txfax itself is aborting and returning prematurely. I wonder if its a negotiation failure. You say it hangs up immediatly, how immediatly? 1 second? 5? On Thu, 2005-10-13 at 11:52 +0200, Coufal Bohuslav wrote: But it seems that Asterisk understand that he has to dial (the dialed number is correct), -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) it seems that zap channel had answered (but nothing to try dial), Channel Zap/4-1 was answered. and lunching txfax Launching txfax(/tmp/ast_fax-1129191936.10240.1804289383.0|caller) on Zap/4-1 and immediately hungup -- Hungup 'Zap/4-1' May be something wrong in zapata.conf? ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [channels] ; language=us signalling=fxs_ks context=default ;context=fax channel = 3-4 Thank for any other sugestions, Bob. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Email to FAX
Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Thanks, Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Email to FAX
Thanks, I'll try it. Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Email to FAX Hi Bob, I've justed looked at inter7 solution and perhaps that is what you're looking for (http://www.inter7.com/?page=astfax) Greetings Otto Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Thanks, Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] tx(rx)_fax for *-1.2.0.beta
Sorry, I could not find it there. I found only version for *-1.1.0. Could You send right URL to me. Thanks, Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roman Sent: Friday, October 07, 2005 4:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] tx(rx)_fax for *-1.2.0.beta On Friday 07 October 2005 13:52, Bohuslav Coufal wrote: Hi all, does anybody have $subj apps. Thanks, Bob. you can download them from spandsp website ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tx(rx)_fax for *-1.2.0.beta
Hi all, does anybody have $subj apps. Thanks, Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls between SIP and IAX
Hi all, I have a trouble when I try to configure asterisk to make calls between IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have phones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection is loss. When I'll make connection between asterisks on SIP then all work fine. Does anybody has any suggestions? Bob. P.S. - I'm using asterisk 1.0.9 on FC3. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Now can I tranfer call form one SIP phone to other during call (unattended transfer)
I have both t and T options in dial command. SIP phones configured with canreinvite=no and when I pres #1 (as I have in features.conf) during call there nothing to happened. Thank for any suggestions. Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I tranfer a call form one SIP phone to other during the call (unattended transfer)
Hi all. I have both t and T options in dial command. SIP phones configured with canreinvite=no and when I press #1 (as I have in features.conf) during call there is nothing to happened. Thanks for any suggestions. Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls between SIP and IAX
Thank You for answer. As I try, the problem occurs when the call come to IAX channel in unknow format of codec. When the calls come in IAX channel with correct codec format (ulaw in my case) calls are O.K. Is it possible to set generally, that im using in all devices ulaw format (calls from H.323 trunk doesnt set it correct). Thanks, Bob. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Saturday, October 01, 2005 7:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Calls between SIP and IAX asterisk console output and details about config files and networking are welcome, and i think, desirable. best regards On 10/1/05, Bohuslav Coufal [EMAIL PROTECTED] wrote: Hi all, I have a trouble when I try to configure asterisk to make calls between IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have phones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection is loss. When I'll make connection between asterisks on SIP then all work fine. Does anybody has any suggestions? Bob. P.S . - I'm using asterisk 1.0.9 on FC3. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IP's (aliases) on asterisk box?
Thats works without any problems. Bob. Dne neděle 28 srpen 2005 21:46 Rich Adamson napsal(a): Anyone have any experience running an asterisk box with a single nic and multiple IP's (aliases)? Have a six class-c production network that needs to be completely re-IP'ed and need to run the box with both an old and new IP for a few days. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring RTP protocol
Hi all, is it possible to monitor RTP protocol (latency, errors, ...) by Asterisk or other software. Thanks for answer, Bob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX compatible phones
For example TEK SIP-IAX 323. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dr. Marios Moutzouris Sent: Wednesday, August 17, 2005 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAX compatible phones Hello, I would like to know which phones are IAX compatible. Thank-you Marios Moutzouris -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/8/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soft Phone
It works very fine for me. Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker Sent: Monday, July 25, 2005 11:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Soft Phone Any suggestions for IAX phones on Linux (without Wine preferred)? Thanks, JASON WALKER - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 25, 2005 11:05 AM Subject: RE: [Asterisk-Users] Soft Phone On Mon, 2005-07-25 at 17:17 +0200, Alex Ongena wrote: Any recommendation for Linux environments (without WINE) ? Thanks Alex Xten runs on linux. http://xten.com/index.php?menu=productssmenu=download -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Receiving fax by app_rxfax over h.323 trunk
Hi,
does anybody has working this konfiguration? For me app_rxfax start receiving,
fax start sending, but after few seconds at begining of the page it stop with
error 400.
My HW PBX configuration is:
ISDN PRI - AVAYA S8300 - H.323 channel - * with app_rxfax
My extensions.conf is:
'7406211' = 1. Goto(fax|666|1)
[fax]
'666' = 1. SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
2. rxfax(${FAXFILE})
'h' = 1. system(/usr/sbin/mailfax ${FAXFILE} [EMAIL PROTECTED] ${CALLERIDNUM})
Thanks for help,
Bob.
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[Asterisk-Users] app_rxfax does not receive
Hi all, I try to use app_rxfax. Aplication app_rxfax start O.K., fax trying to send, but it will stop at the beginning of page and after few seconds it stop with error 400. Does anybody has any suggestions? Thanks, Bob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial more then 9 digits
Could you kick me, I can't dial more then 9 digits. Is anyone some default length of extensions or dialed number. Thanks, Bob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial more then 9 digits
my exten
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[default]
; If the number dialed by the calling party was 2000, then
; Dial the user 2000 via the SIP channel driver. Let the number
; ring for 20 seconds, and if no answer, proceed to priority 2.
; If the number gives a busy result, then jump to priority 102
;exten = s,1,Dial(SIP/${EXTEN})
;exten = s,1,Dial(SIP/7406100)
exten = 7406100,1,Dial(SIP/7406100)
exten = 7406101,1,Dial(H323/[EMAIL PROTECTED])
exten = 7406105,1,Dial(SIP/7406105)
exten = 7406106,1,Dial(SIP/7406106)
exten = 7406200,1,Dial(SIP/7406200)
exten = _74068XX,1,Dial(H323/[EMAIL PROTECTED])
exten = _OO.,1,Dial(H323/[EMAIL PROTECTED])
exten = _X,1,Dial(H323/[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of altus
Sent: Wednesday, June 15, 2005 12:31 PM
To: asterisk
Subject: Re: [Asterisk-Users] Dial more then 9 digits
no
I can?
how is your dialout rules ?
I have a client where you have to dial a 4 digit pin and then the rest
of the number
I simply have a
exten = _1234.,1,Dail...
On Wed, 2005-06-15 at 12:20 +0200, Bohuslav Coufal wrote:
Could you kick me, I can't dial more then 9 digits. Is anyone some
default length of extensions or dialed number.
Thanks,
Bob.
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RE: [Asterisk-Users] Dial more then 9 digits
This is double-zero international prefix. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hamill Sent: Wednesday, June 15, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Dial more then 9 digits On Wednesday 15 June 2005 12:40, altus wrote: exten = _OO.,1,Dial(H323/[EMAIL PROTECTED]) Sorry, I couldn't help but notice this... Is that really meant to be _OO (capital letter 'Oh') rather than _00 as the double-zero international prefix? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AVAYA Asteris H323 chanel
Thanks, now it works. Problem was in CVS and libraries versions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, June 15, 2005 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AVAYA Asteris H323 chanel Yes. I configured it for a former employer. We had an S8700 talking to * via h.323 with no problems. oh323 did need to have it's rtp frame size adjusted initially for some sound quality issues, and we needed to dbl check that oh323 wasn't trying to negotiate for codecs that * didn't want to handle. Aside from that, it's been working flawlessly since. On 6/14/05, Bohuslav Coufal [EMAIL PROTECTED] wrote: I'm trying to make H.323 trunk between AVAYAAsterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AVAYA Asteris H323 chanel
I'm trying to make H.323 trunk between AVAYAAsterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
