[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:

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:

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:

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:

RE: [Asterisk-Users] Asterisk Compilation with H323 working on it

-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

[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?

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

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

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

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

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

[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

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

RE: [Asterisk-Users] tx(rx)_fax for *-1.2.0.beta

-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

[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

[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.

[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

[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

RE: [Asterisk-Users] Calls between SIP and IAX

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

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

[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

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

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

[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'

[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] 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

RE: [Asterisk-Users] Dial more then 9 digits

... 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. ___ Asterisk-Users mailing list Asterisk-Users

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

RE: [Asterisk-Users] AVAYA Asteris H323 chanel

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

[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