[Asterisk-Users] Web interface

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing l

[Asterisk-Users] IAX call rejected.....who was trying to reach 's@'

dear All i signed up with an Aussie provider who gives me a DID in Aust... when I call my number I get the following on the console Mar 21 05:54:15 NOTICE[68071]: chan_iax2.c:6123 socket_read: Rejected connect at tempt from 203.13.163.245, who was trying to reach 's@' the s part i can understand

[Asterisk-Users] blind xfer works atxfer doesn't...help!

Hi all I am having problems with atxfer if I do the extact same thing with blind xfer it works fine when I hit press #2 (defined in conf for atxfer) i get "transfer" I dial the number I want and i get the following on the console -- Playing 'pbx-transfer' (language 'en') -- Executing Dial("L

[Asterisk-Users] looking for DID in spain

Dear all I am looking for a per minute DID # in spain..either IAX/SIP Thanks Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] vars for transfered calls

areacode - used by dial() and a cellnum used to forward the call when it is not answered can someone help Thanks Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] diffrent area codes for diffrent phones in dialplan

I have 3 sets of SIP phones all in diff area codes that need to access the PSTN I need to it so that a 7 digit number is converted to a 10 digit with the correct ara code eg a call coming from sip-phone1 needs aera code AAA and a call coming fom sip-phone2 needs BBB how can this be setup in the

Re: [Asterisk-Users] PAP2-NA point to poitn calls ??...(Direct IP Dialing)

At 11:40 AM 3/11/2005, you wrote: the only way I found to do this was have them register with a * server and have * connect them Hello, I need to know if there is an option in the PAP2-NA Web Configurator like "Enable IP dialing: yes/no" I need to make "point to point" calls with two PAP2-NA by IP

Re: [Asterisk-Users] looking for cheap 4 port FXS card

At 04:43 AM 3/8/2005, you wrote: it doesnt have to be a card it can be a device. long as it has 2-4 ports Jer Dear list I need to find a way to hook up 4 analog phones to * was wondering if anyone had old hardware not being used and they are willing to sell or know of any places to

[Asterisk-Users] looking for cheap 4 port FXS card

Dear list I need to find a way to hook up 4 analog phones to * was wondering if anyone had old hardware not being used and they are willing to sell or know of any places to buy that are cheaper then digium :/ Thanks I need 2 ports but would perfer 4 let me know Thanks

[Asterisk-Users] CVS compile error utils.c

it to ast_pthread_create(thread, attr, start_routine, data); and it compiledbut when running asterisk it gives me a nasty bus error and dies :/ now i'm feeling stupoidcan someone help??????? :) Jer ___ Asterisk-Users mailing lis

Re: [Asterisk-Users] Call transfer questions

=> *2 it still seems to want to accept only # as transfer I am running Asterisk CVS-v1-0-03/07/05-06:50:06 Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or upd

[Asterisk-Users] Call transfer questions

is T in dial() Thanks sorry for the noob question Jer ___ 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/lis

Re: [Asterisk-Users] cant compile app_meetme2

At 04:34 AM 3/5/2005, you wrote: The error messages are Postgres related. You need to have a special postgres include file (postgres-dev files) to make it compile or disable postpres support somehow. I'm using debian and the the concering include file resided in a subdirectory of what asterisk was

[Asterisk-Users] cant compile app_meetme2

Dear all I am get the following problem when trying to compile app_meetme2 using mysql...it seems to want to use pgsql.? anyone my Makefile looks like app_meetme2.o: app_meetme2.c #$(CC) -pipe $(CFLAGS) -c -o app_meetme2.o app_meetme2.c $(CC) -pipe -I/usr/local/include/mysql

[Asterisk-Users] iconecthere and *

am I missing something all that happens if * never even sees the call (or more to the point i dont see it with sip debug!) THanks Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-use

[Asterisk-Users] chan_modem dialout

can a voice modem make an outbound call? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Some (lack of) answers regarding the wakeup call application...

At 11:27 AM 6/1/2004, you wrote: Rob I would be very interested Since I only seem to get questions, and no feedback, from the Wiki page, I'll ask here. There seems to be no lack of opinions here... I have a working wakeup call system on my home * system. The architecture is something I'm not pe

[Asterisk-Users] STREAM FILE question

Dear all I was wondering is there a way to advance/rewind in playback?(STREAM FILE) say 5 seconds somehow i don't think so but I thought I' would ask Thanks Jer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailma

[Asterisk-Users] unable to use EXEC in AGI

dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[12

[Asterisk-Users] Phone_read Resource temporarily unavailable

Could someone tell me why I would get this using a linejack POTS May 20 17:43:05 WARNING[1217602880]: chan_phone.c:417 phone_read: Error reading: Resource temporarily unavailable May 20 17:43:05 WARNING[1217602880]: app_festival.c:181 send_waveform_to_channel : Null frame == hangup() detected Ma

Re: [Asterisk-Users] AGI/php script not working

At 08:47 AM 5/20/2004, you wrote: -vvvc mode I see *CLI> -- Executing Wait("Phone/phone0", "1") in new stack -- Executing Answer("Phone/phone0", "") in new stack -- Executing AGI("Phone/phone0", "test.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -

[Asterisk-Users] AGI/php script not working

Dear all I am just getting started with AGI so I wrote the following script as a simple test but all that happens is silence before it times out and hangs up can someone help to get me started? yet if i use the agi-test.agi script everything works I don't see the difference Thanks php -q yet al

[Asterisk-Users] Linejack dialout

Dear all I read on the list back in 2003 that * does not support IXJ LineJACK dialout yet is this still the case? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] Asterisk not answering phone

Dear list.. I am trying to use * to answer a call coming in from the PSTN port of a line jack I am using mode=fxo in phone.conf but the line just rings and gins in mode.=dialtone it works fine on the POTS port any ideas what I am doing wrong? Thanks ___

[Asterisk-Users] failed compile

Hi all I am trying to compile Asterisk on RH9 I am have installed pwlib_1.5.2 readline-4.3-5 readline-devel-4.3-5 openssl-devel-0.9.7a-2 openssl-0.9.7a-2 openh323_1.12.2 I read that it has to do with pwlib not being installed correctly so i made sure it was in /usr/local correctly When i try to

RE: [Asterisk-Users] problems compiling h323 support

l Business [EMAIL PROTECTED] | 910 16th Street, #1220 (303) 228-0070 x821 --The Future is Now!--| Denver, CO 80202(303) 228-0077 fax > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jer > Sent: Monday, Ma

[Asterisk-Users] problems compiling h323 support

Hi all I am trying to compile Asterisk on RH9 I am have installed pwlib_1.5.2 readline-4.3-5 readline-devel-4.3-5 openssl-devel-0.9.7a-2 openssl-0.9.7a-2 openh323_1.12.2 I read that it has to do with pwlib not being installed correctly so i made sure it was in /usr/local correctly When i try to