[Asterisk-Users] Fax with wildcards

to send concurrently faxes on one/two/three/four E1? Is it possible either? Thank you in advance, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] PRI numbering plan

number? In other case I can't place call or won't see the calling number on the phone. Thanks in advance, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] rxfax problem

Hello, I tryed out spandsp with libtiff-3.5.7 and with Asterisk from CVS. I tryed to receive a fax on a CAPI channel. Finally I got a file with 8 byte length (/tmp/testfax.tif). How can I do next? Thanks in advance, Thomas ps: what are hardware requirements for sending/receiving

[Asterisk-Users] voicemail files in Asterisk have rights 600 , I need 644

Hello, I would like to access the Audio-Files from my Apache Web Server and would need the voicemail files generated with rights 644. Any idea how to do this? thanks Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] voicemail files in Asterisk have rights 600 , I need 644

, thanks for the information. I will try the solution next week and will report the results. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Re: chan_sip.c:5262 sip_reg_timeout Probably a DNS error for registration

. best regards Thomas ___ --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] sipgate.de question

On Saturday 25 February 2006 23:56, Michiel van Baak wrote: Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18

[Asterisk-Users] call files and cdr I need src different from CallerID(number)

(number) in the cdr file. So I dont have the choice to set CALLERID(number) different to the peer-name (src in the cdr file). How this can be fixed. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Zap channels busy. Have to soft hangup.

in at the same time it happens that instead of dialing out I get the person who just dialed in. Rather confusing for both of us. Honestly I did dig around a lot and could not find any specifics about my issue. Any help highly appreciated. -- Thomas

[Asterisk-Users] pattern matching in dialplan problems matching _NNN

I hear playtones invalid. It seemed to be that pattern matching with N is not working as designed. Any ideas? best regards Thomas extensions.conf - [internal] #include /etc/asterisk/x_internal.conf x_internal.conf - exten = 210,1,Macro

Re: [Asterisk-Users] pattern matching in dialplan problems matching _NNN

thanks... _NXX works for me best regards Thomas On Saturday 07 January 2006 16:37, Peter Bowyer wrote: On 07/01/06, Thomas [EMAIL PROTECTED] wrote: Hi, I have a problem with pattern matching N what should digit 2 to 9 in Asterisk 1.2.1. If I dial 220 I did not get an PlayBack

[Asterisk-Users] chan_sip.c:5262 sip_reg_timeout Probably a DNS error for registration

(Attempt #11) chan_sip.c:5345 transmit_register: Probably a DNS error for registration to [EMAIL PROTECTED], trying REGISTER again (after 60 seconds) best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

[asterisk-users] Asterisk in debian Wheezy 1.8.13.1 vs. Squeeze 1.8.23.1

Hello, in Squeeze Asterisk 1.8.23.1 is installed, in Wheezy older version 1.8.13.1~dfsg1-3+deb7u3. With version 1.8.13.1 I have some problems so I would like to install version 1.8.23.1 used in Squeeze whats running fine for me. How I can do this? thanks for help Thomas

[asterisk-users] asterisk segfault debian jessie asterisk 11.13

Hi, every two weeks the asterisk process has a segfault. Any idea whats reason or what I can do... thanks pc kernel: [1780743.239296] asterisk[11362]: segfault at 0 ip (null) sp 7f1e396b04a8 error 14 version is debian jessie Asterisk 11.13.1~dfsg-2+b1 built by buildd @ brahms on

[asterisk-users] AMI StarAstAPI.php not working for commands like SIPpeers with more output

; string(6) " start" ["Message:"]=> string(29) " Peer status list will follow" But how I receive the peer list? best regards Thomas -- _ -- Bandwidth and Colocation Provided by http:/

[asterisk-users] nagios asterisk check SIP

Hi, Iam loocking for an programm to check the SIP port of an Asterisk asterisk. Ome time ago I have used #/usr/bin/sipsak but it seemed that it is not working anymore? Any ideas what I can use instead? best regards Thomas

[asterisk-users] looking for soft phone can be manged like Snom phones

http://my.serverIP/pbx/snomcom.php?mac=$mac=1401 I would need an softphone for Linux and/or Mac with same or similar functionality. best regads Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Queue logfile txt format in mySQL needed

in the dialplan direct in an mySQL database. Is there any way that Asterisk write this information direct in an mySQL database instead of using var/log/asterisk/queue_log? best regards Thomas -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Queue logfile txt format in mySQL needed

Am Donnerstag, 21. Januar 2016, 09:52:53 schrieben Sie: > > From: Thomas <thomasit...@gmail.com> > > To: asterisk-users@lists.digium.com, > > Date: 01/21/2016 04:17 AM > > Subject: [asterisk-users] Queue logfile txt format in mySQL needed > > Sent by: ast

[asterisk-users] ConfBridge increase talking volume as standard

Hello, is it possible to increase talking volume for caller in ConfBridge as standard without need to press buttons after joining an conference room. best regards Thomas -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] softphone instead of desktop phones

with an headset? I want to save cost for desktop phones. thanks Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk

[asterisk-users] AMI Originate not working

Hello, I want to call an phone and if phone picked up I want to ring another phone. Or I want to connect to an running channel and then call another phone or move to an ConfBridge Iam using PHP $channel = 'IAX2/556-1696'; or $channel = 'SIP/0019736363636@outbound.patton'; $exten = '';

[asterisk-users] Confbridge or Bridge

or congestion I have problem when agent is the leaving before callee, that hangup should send to callee channel. Also habe somebody an source for an soundfile with every 5s an beep that agent is hearing when he is the only person in confroom. thanks Thomas

[asterisk-users] Conference bridge profile does not exist

nux on 2016-10-24 19:32:53 UTC best regards Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? St

[asterisk-users] UNIQUEID not unique in different channels

Hello, since al long time I have used UNIQUEID for identify calls in my dialplan, statistics... Now I have had an problem, after I have checked log file I found out following: calls same time ( hours:seconds) came in. CallID, DID, channel name (3cf9 to 3cfa) are different. Only

[asterisk-users] Read sometimes disconnects user

Hello, I am asking the user to enter his mobile phone followed by # using Read(). From time to time the Read() application disconnects the user while he is typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. *same =

Re: [asterisk-users] Read sometimes disconnects user

But even then, it should do another attempt, play the file again and accept DMTF input, as I specified that there should be 2 attempts. *same = n,Read(mobileNumber,app/input-mobile,10,,2,15)* * * 2012/10/30 Steve Edwards asterisk@sedwards.com On Tue, 30 Oct 2012, Thomas Thomas wrote: I

Re: [asterisk-users] Read sometimes disconnects user

, ... seems random. 2012/10/30 Thomas Thomas debussy...@gmail.com But even then, it should do another attempt, play the file again and accept DMTF input, as I specified that there should be 2 attempts. *same = n,Read(mobileNumber,app/input-mobile,10,,2,15)* * * 2012/10/30 Steve Edwards

[asterisk-users] Uprading to Asterisk 11 issues

Hello, I installed Asterisk 11 via the following command * svn co http://svn.asterisk.org/svn/asterisk/branches/11* (as written in asteriskdocs.org http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Installing_id240883.html ) But it seems that I have a development version instead of

[Asterisk-Users] busy signal not in cdr

hi list. i have the following problem. if i dial an ip endpoint from my ip phone and the endpoint is busy, in my cdr i see (answered). I think there must be busy. why is that? any hints? thx, thomas ___ Asterisk-Users mailing list Asterisk-Users

[Asterisk-Users] TE110p card with Euro ISDN (Ericsson switch)

I am trying to use TE110p card for Euro ISDN with Ericsson AMS switch. I consistently get one of the following errors: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 or PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 My zaptel.conf file:

[Asterisk-Users] Channel name (and substring)

How do I get the bit like IAX2/white_phone in extensions.conf eg from pre-defined variables or variants thereof ? What I *do* get is strings like this IAX2/[EMAIL PROTECTED] from ${CHANNEL}, but that's the full channel name. What am I missing here ? Thanks, Thomas

Re: [Asterisk-Users] Channel name (and substring)

in the ${CHANNEL} variable because the convention seems to change from '@' to '-'. It means I can't write a generic translation. Thanks, Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Channel name (and substring)

On Thu, Mar 17, 2005 at 12:24:00PM -0800, Sean Kennedy wrote: Thomas Andrews wrote: How do I get the bit like IAX2/white_phone in extensions.conf eg from pre-defined variables or variants thereof ? What I *do* get is strings like this IAX2/[EMAIL PROTECTED] from ${CHANNEL}, but that's

[Asterisk-Users] leaky reload

- ie if I uncomment the line and reload then it learns about the caller id Uniden Dead. Why is this a one-way process ? Thanks, Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] AGI-like calls in the [globals] section

I'd like to set up some global parameters once at startup using an external program. (eg like one would with AGI) How can I do that ? Thanks, Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

Re: [Asterisk-Users] :: What does it take to upgrade? :: Newbie Q ::

On Sat, Mar 19, 2005 at 10:40:10AM +0100, Reuben Grech wrote: How can I upgrade Asterisk to the latest version ?? You will have to use cvs: http://www.automated.it/guidetoasterisk.htm#_Toc49248761 Will I need to re-compile?? Yes. -Thomas

[Asterisk-Users] Asterisk compatible IP Phones

, Thomas Lee ___ 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] new release of chan_misdn !

-beta-0.1.0.tgz Greets, Thomas. ___ 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] capi segfault when incoming call is answered

I have a Fritz! card set up to use capi, however when incoming calls to the card are answered, asterisk segfaults. Here is the output of gdb: #0 0x4014f7af in memcpy () from /lib/tls/libc.so.6 #1 0x081316b0 in ?? () #2 0x08130680 in ?? () #3 0x40432da6 in pipe_msg (PLCI=257, CMSG=0x405052c0)

Re: [Asterisk-Users] capi segfault when incoming call is answered

On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote: I have a Fritz! card set up to use capi, however when incoming calls to the card are answered, asterisk segfaults. Just for the record, my capi.conf looks like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8

[Asterisk-Users] ignorepat changing the sound of dialtone

, Thomas ___ 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] ignorepat changing the sound of dialtone

On Sun, Apr 10, 2005 at 09:47:36AM -0500, Andy Hamilton wrote: On Apr 10, 2005 7:30 AM, Thomas Andrews [EMAIL PROTECTED] wrote: Is it possible to play a different dialtone as soon as a user dials say '0' for an outside line ? Ignorepat is an inadequate solution because local users

[Asterisk-Users] semantics terminology

Hi, Where can I get a good reference for terminology terms. For instance I'd like to know if there's an accepted difference between the terms divert and forward. Thanks Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] is deadlocking with the Manager API still a problem?

I have read on the wiki that deadlocking is a problem when using the Manager API. I have the manager api running. I am making hundreds of telnet connections from a remote server (running windows) to the asterisk server via telnet / manager api.The remote machine is telnetting in to the manager

[Asterisk-Users] spandsp-0.0.2pre6

Hi Folks, hi Steve I get following error on loading app_rx/txfax.so: ...WARNING[10458]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: symbol errno, version GLIBC_2.0 not defined in file libc.so.6 with link time reference Unable to load app_rxfax.so Spandsp compiled and

Re: [Asterisk-Users] spandsp-0.0.2pre6

Hallo Thomas Niesel On Wed, 29 Dec 2004 22:03:05 +0100 you wrote: Hi Folks, hi Steve I get following error on loading app_rx/txfax.so: ...WARNING[10458]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: symbol errno, version GLIBC_2.0 not defined in file libc.so.6

[Asterisk-Users] Problems to use asterisk with mysql /odbc

hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version. i like to store usernames and passwords in a sql database. i like to log failed authentification-passwords, to create a blacklist for securityreasons. i thingk a sql-database is a good way to log these actions. i don.t find

[Asterisk-Users] how can setup mysql in sip.conf?

Hi, i have a problem: asterisk 1.0.3 is my current version. i copy from asterisk-sources: contrib/scripts/retrieve_sip_conf_from_mysql.pl in my /usr/asterisk/etc/asterisk directory. I edit this script and remove -T -option #!/usr/bin/perl -w i create the mysql database: mysqladmin create

[Asterisk-Users] calling with out registration

hi, i am using Asterisk CVS-05/31/04. i have the problem that sip clients can make calls over asterisk without registering befor. the xlite is not loged in with any username/secret bit still can make calls over asterisk. how can that be? thx for help. thomas

Re: [Asterisk-Users] calling with out registration

hi, Am 06.01.2005 um 11:23 schrieb Ronald Wiplinger: Thomas Küpper wrote: hi, i am using Asterisk CVS-05/31/04. i have the problem that sip clients can make calls over asterisk without registering befor. the xlite is not loged in with any username/secret bit still can make calls over asterisk

[Asterisk-Users] Question to authenficate client automaticlly

Hi, i have setting up asterisk for mysql. i using the template-database: sipfriends. i have a vpn in the office. i like to setup asterisk: when a client make authentification request: username and password stores automaticlly in the sql database. any users in the vpn can setup the own name

[Asterisk-Users] chan_misdn - new release ! Please test it.

/download/chan_misdn-beta-0.0.3-rc4.tgz You can report bugs and feature requests to www.beronet.com/bugs Have fun! Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Friedrichstr. 231 Haus D, 4. OG 10969 Berlin FON:+49 (0) 30 259389-14 FAX:+49 (0

Re: [Asterisk-Users] EuroISDN BRI 2 or 4 wires?

On Wed, Jan 12, 2005 at 11:06:18PM +0100, Remco Barende wrote: Hi List! Have a weird problem with ISDN in The Netherlands. The line that is coming in from the telco is 2 wires. The line is connected to an NT1 using the middle pair of a UTP connector. So far sop good. The incomming 2

Re: [Asterisk-Users] ISDN + chan_capi

On Tue, Jan 18, 2005 at 02:53:26PM +0100, Dave Cotton wrote: I really can't understand what is happening. I have the same set up in two locations:- 2 fritz cards in each with the patches and the same capi.conf except for the MSNs. One installation calls in on controller 1 and starts calls

[Asterisk-Users] chan_misdn 0.0.3-rc5 - new release ! Please test it.

/download/chan_misdn-beta-0.0.3-rc5.tgz You can report bugs and feature requests to www.beronet.com/bugs Have fun! Thomas. -- *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Friedrichstr. 231 Haus D, 4. OG 10969 Berlin FON:+49 (0) 30 259389-14 FAX

Re: [Asterisk-Users] AVM Fritz crash

On Mon, Jan 24, 2005 at 04:31:33PM +, Steve Hill wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Not strictly Asterisk related, but I'm trying to get an AVM Fritz! card to work with Asterisk on a Fedora Core 3 box. I've grabbed the official AVM Fritz CAPI driver from the AVM

Re: [Asterisk-Users] Menu tree for voicemailmain application

On Mon, Jan 24, 2005 at 10:55:03AM -0500, David Brodbeck wrote: Is there a menu tree diagram somewhere for the Voicemailmain application? I know my users will ask for one, and before I started drawing my own I thought I'd see if someone already had. Ask the wiki for voicemailmain, use the

Re: [Asterisk-Users] AVM Fritz crash

Hallo Steve Hill On Tue, 25 Jan 2005 09:49:05 + (GMT) you wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 25 Jan 2005, Thomas Niesel wrote: Does anyone have any suggestions? Quick shot: SMP, HT? It is an SMP machine but it's only running a uniprocessor kernel

Re: [Asterisk-Users] SIP UDP ports on firewal to open

Hallo Steve Radich On Tue, 25 Jan 2005 11:48:44 -0500 you wrote: I notice most things say to open ports 1-2 for UDP for SIP, however from time to time this range isn't where Asterisk is opening the ports: We're at xxx.xxx.xxx.xxx port 8542 Answering with capability 0x2(GSM)

[Asterisk-Users] Re: Polycom Phones

I purchased some 20 Polycom phones (brand new) for a very good price of around $165 each. Now I am having a nightmare in configuring them. I pulled the bootrom, SIP and config files from freedomphones.com, modified them for my need and and started configuring the phones. First couple of phones

[Asterisk-Users] Re: Polycom phones

From: Walt Reed [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Polycom Phones To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On

Re: [Asterisk-Users] Attended call transfer

Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Any news? I am also looking for #-Transfers for asterisk-stable. Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Problmes compilling *

/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/lib/gcc-lib/i586-suse-linux/3.3.3/../../../../i586-suse-linux/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 Can anyone know what it is? -lssl??? Thanks Thomas -- Thomas Hardb [EMAIL

Re: [Asterisk-Users] HFC-5/S + Asterisk

On Tue, Feb 01, 2005 at 06:13:41PM +, Nigel Kukard wrote: Hi Guys, I got 2 internal HFC-S[5] (not sure what character this is) internal modem cards yesterday, thought I would get different makes as the one was a named brand, the other not... but anyway... Its picked up by lscpi as a

Re: [Asterisk-Users] HFC-5/S + Asterisk

On Wed, Feb 02, 2005 at 08:06:35AM +0100, Peer Oliver Schmidt wrote: Thomas Niesel wrote: [..] = Your Card should work with i4l (bad) and zaphfc from junghanns (good) Forget about Capi and mISDN, go for kernel 2.6.10 along with zaphfc, ztdummy together with uhci for timing and ask the wiki

Re: [Asterisk-Users] HFC-5/S + Asterisk

On Wed, Feb 02, 2005 at 10:49:26AM +0100, Klaus-Peter Junghanns wrote: Am Mittwoch, den 02.02.2005, 10:41 +0100 schrieb Thomas Niesel: On Wed, Feb 02, 2005 at 08:06:35AM +0100, Peer Oliver Schmidt wrote: Thomas Niesel wrote: [..] = Your Card should work with i4l (bad) and zaphfc

[Asterisk-Users] wcte11xp Trouble

Hello List, i am trying to set up an * box with a debian 2.6.10 Kernel. the compilation of asterisk and the zaptel drivers worked almost perfectly. so i did modprobe zaptel modprobe wcte11xp (which is the correct driver for my card) ztcfg -v and this results in ZT_SPANCONFIG failed on span 1: No

Re: [Asterisk-Users] 4xHFC-s cards vs 1 quadbri HFC-4S card ?

Hallo Robert Rozman On Tue, 15 Feb 2005 13:54:48 +0100 you wrote: Hi, I wonder what makes the difference between inserting 4 HFC-S cards (cca. 120 EUR) and using 1 QuadBRI card (approx. 700 EUR) ? What makes such difference ? Is it possible to do first configuration ? 1st: 4Slots for

[Asterisk-Users] Voicemail and busy tone

to record the busy tone that the provider sends. How can I avoid this behaviour? Thx. Thomas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] Re: Voicemail and busy tone

: [Asterisk-Users] Re: Voicemail and busy tone Thomas == Thomas RULMONT [EMAIL PROTECTED] writes: Thomas When I call asterisk from outside, I leave my message, but, Thomas after hanging on, voicemail continues to record the busy tone Thomas that the provider sends. How can I avoid this behaviour? First

Re: [Asterisk-Users] capiECT problem

On Wed, Feb 16, 2005 at 08:58:41PM +0100, Robert Rozman wrote: Hi, I'm trying to get capiECT working. I'd like to transfer call to another ISDN router connected extension and free channel from router to Asterisk. I get incoming call on CAPI and would liek to transfer it to dialed local

Re: [Asterisk-Users] Strange MSN issue with HFC-s

On Thu, Feb 17, 2005 at 03:02:07PM +0100, Marc SCHAEFER wrote: Hi, I have two HFC-s boards I configured in NT and TE mode respectively. When I connect the two boards together, I can dial extensions and I see the correct called and caller ID numbers: -- Executing SetCallerID(Zap/2-1,

Re: [Asterisk-Users] Re: capiECT problem

On Thu, Feb 17, 2005 at 11:52:16PM +0100, Aldo Bergamini wrote: ...cut This makes me wonder if I understand your hint; is this what you are suggesting? [outbound-capi-local] exten = _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn CAPI/${CALLERIDNUM}) exten =

[Asterisk-Users] CDR for callback

Some of my clients of hosted PBX service want to use it for callback when they cannot use the ATA. This is the scenario 1. Asterisk calls Party A at numA. 2. When A picks up the phone, he hears the announcement to enter the destination number, numB. He enters numB 3. Asterisk Dials numB and

Re: [Asterisk-Users] Problem with ISDN Dialin via CAPI

On Mon, Feb 21, 2005 at 11:36:45AM +0100, Müller, Thorsten wrote: Hello List, because this is my first post to this list, i'd like to introduce myself. My name is Thorsten Müller, 26 years old and live near Frankfurt/Main in germany. Okay, now to the reason for this posting: I just

Re: [Asterisk-Users] Language Problems

On Sat, Feb 26, 2005 at 08:53:23AM +0100, Peter Svensson wrote: On Sat, 26 Feb 2005, Anton Krall wrote: Asterisk will add sp after the last directory name. You will need the following directories (or symlinks): /var/log/asterisk/sounds/sp /var/log/asterisk/sounds/phonetic/sp

Re: [Asterisk-Users] dialout with PPP on ISDN to an ISP

On Sun, Feb 27, 2005 at 10:32:21PM -0600, Steven Critchfield wrote: On Mon, 2005-02-28 at 00:43 +0100, Ilija Poznic wrote: Hello my name is Ilija Poznic and I have a problem. My configuration is 1. Digium TDM4000P with one FXS. 2. AVM Fritz ISDN adapter

[Asterisk-Users] Manager Message: Originate failed being generated when callee does not pick up

I am using the Manager tooriginate calls. I am getting "Message: Originate failed" even the phone is ringing on the other end of the line. How can I reliably know if the phone on the other end of the line is receiving the call? Thanks, Tom Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do

[Asterisk-Users] how to increase max number of simulatneous outgoing calls

Hello, I would like to know how to maximize the number of simultatenous outgoing calls. The application I am working on uses the Manager API to originatethe outgoing calls, and the callswill hang up one or two seconds after the callee picks up the phone. I know it sounds strange but that is what

Re: [Asterisk-Users] Suse 9.2 + CAPI Driver

On Mon, Feb 28, 2005 at 05:06:47PM -, Victor Alvarez wrote: Hello, I'm trying to install CAPI Driver for Suse 9.2 and I found the documentation for this pretty old since It refers to Suse 8.2 ( http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install ). This is

RE: [Asterisk-Users] Manager Message: Originate failed beinggenerated when callee does not pick up

Thanks for your help. From what you said it looks like I should not use Originate, but there is no alternative to the Originate action if I just want to make an outgoing call is there? This is what my code is sending to the Manager API: clientSocket.Send(Encoding.ASCII.GetBytes(Action:

RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls

Hi Matt, in your experience is there a 100% reliable way to know that the callee phone is ringing? In my situation I don't need to know if they pick up or not, I need to know (as reliably as possible) if the calee phone number is ringing. Thanks, Tom --- mattf [EMAIL PROTECTED] wrote:

[Asterisk-Users] Teles GW authentification

for your help, regards, thomas ___ 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] Cisco 7960

for the migration. Is there a special order of firmware-upgrades? Thanks a lot Thomas ___ 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] Cisco 7960 Problem - Phone Unprovisioned

to 192.168.0.83, 0 bytes So i think it now wants to load the file P0S3-07-3-00.bin. But i have not modified the OS79XX.TXT. Does anybody know, where i can get this file and what next steps to perform? Thanks a lot Thomas ___ Asterisk-Users mailing list

[Asterisk-Users] Call through. with 2xT1 .configuration

Hello all, It 's dificult to explain; The system I need is an box option (based on *), that I would add to an existing PABX (ie: Nortel with 600 ext). I need two E1/T2 card to plug the system between Telco (FT) and PABX (Nortel)! One card for France Telecom Side (E1a) and one other to Nortel

[Asterisk-Users] Data calls (ISDN/64k) through * PRI

is an audio call. Any ideas ? Thanks, Thomas. ___ 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

AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

Thanks Maik, i try it -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Maik Schmitt Gesendet: Donnerstag, 22. Januar 2004 11:21 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Any ideas ? exten =

AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

Hi Maik, is there any special version from libpri or asterisk necessary since it works ? I'am runnig version: CVS-11/11/03-11:49:55 and it don't work :-( Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Maik Schmitt Gesendet

AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data Here my pri debug ouput Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: SETUP (5) Bearer Capability (len= 2) [

AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

in 64K? Regards, Gus - Original Message - From: Thomas Haeger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 12:28 PM Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Hi , maybe someone knows what's going wrong... The incoming data call

AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

Has somebody got it work at all ? I mean data calls (ISDN 64k) through asterisk. Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Donnerstag, 22. Januar 2004 19:07 An: [EMAIL PROTECTED] Betreff: AW: [Asterisk

[Asterisk-Users] WG: Reference projects which using Asterisk !?

it on the Asterisk website or the asterisk wiki. This would bring trust to the disbelieving folks What do you think ? Or, is there someone who can report something about such projects ? Maybe it is something for the documentation project Thanks, Thomas

AW: [Asterisk-Users] Data call transfer

to bugs.digium.com and look at bug number 960 at libpri project Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Tomica Crnek Gesendet: Donnerstag, 5. Februar 2004 10:05 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Data call transfer

Re: [Asterisk-Users] Problems with ATA's locking up..

using MGCP-Image Version: v2.16.1.ms ata18x (Build 030814a) Thomas PS Has anyone got Transfer with Flash working perfectly? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

AW: [Asterisk-Users] Need to interface to BRIs

Look at this: http://www.junghanns.net/asterisk/page17.html Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jim Archer Gesendet: Montag, 16. Februar 2004 10:11 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Need to interface

[Asterisk-Users] nasty segfault with previous strange Zaptel warnings

minute cleanups Have somebody an idea what the problem is ? The Zap Hardware which is running on my machine is a E400P. Thanks for help. Best regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] callerid will not be set

: Newchannel Channel: Zap/13-1 State: Rsrvd Callerid: unknown Uniqueid: 1077819120.3 Event: Hangup Channel: Zap/13-1 Uniqueid: 1077819120.3 Is this a bug ? Or may i do somthing wrong ? Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas

[Asterisk-Users] 2 Linphones communicating through Asterisk?

Thomas Sip read: REGISTER sip:192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 From: sip:[EMAIL PROTECTED];tag=680816676 To: sip:[EMAIL PROTECTED];tag=680816676 Call-ID: [EMAIL PROTECTED] CSeq: 0 REGISTER Contact: sip:[EMAIL PROTECTED] max-forwards: 10 expires: 3600

  1   2   3   4   5   6   7   8   9   10   >