Re: [asterisk-users] Asterisk on Windows

I know who is lost here :) for sure not digium ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] Telecom HU cannot callforward to external number

Hi List! I have a Problem with Telecom Hungary, if I set a callforwarding on the Snom, to an external number (mobile). Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0, snom-7.7.30 When I call the Snom (Extension 68), it responds with 302 moved temporarily, and Asterisk try to

Re: [asterisk-users] Telecom HU cannot callforward to external number

, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Wed, Jul 18, 2012 at 4:23 PM, Christian Gansberger

Re: [asterisk-users] asterisk 1.8.9.2 channel.c: Channel allocation failed

Yes, I can give a higher ulimit, but I want to know why there were so much fd's. As I found out yesterday, the reason of running out of available file descriptors was: Some Agents in the Callcenter made ChanSpy on several Calls, but they didn't stop spying with *-key, just hangup the phone, and

[asterisk-users] SIP Provider Russia, Ukraine, Poland

Hello List! I'm searching for SIP-Providers in the following countries: Russia Ukraine Poland I need a geographical number for each country, maybe a prepaid SIP-Account, trunking is not important. Has anyone some experience with these countries? yours christian --

Re: [asterisk-users] sip issue

Hello, You have to disable RTP-Encryption on your Snom under Identity, RTP. It is set to on per default. On 31 October 2011 13:33, salaheddine elharit salah.elharit...@gmail.com wrote: hello list i have installed asterisk 1.8.7.1 and i have configured 2 account for sip in order to do

Re: [asterisk-users] sip issue

:) On 31 October 2011 15:36, salaheddine elharit salah.elharit...@gmail.com wrote: thank you so much all works without issue now 2011/10/31 Christian Gansberger christian.gansber...@accm.at Hello, You have to disable RTP-Encryption on your Snom under Identity, RTP. It is set to on per

Re: [asterisk-users] asterisk-1.8 crash if no extension specified in Dial

I had that problem too, I wastesting with asterisk 1.8.3.2 and come across this: Call from one extension to another with: [macro-internal-call];ARG1=extension to call exten = s,1,Set(TOCALL=${DB(SIP/${ARG1})}) exten = s,2,Dial(SIP/${TOCALL},60,tT) ... As I had no entry in the asteriskdb, so

[asterisk-users] Asterisk 1.8.3.2 core dump chan_sip.c

Hello, I'm testing with asterisk 1.8.3.2 and come across this: Call from one extension to another with: [macro-internal-call];ARG1=extension to call exten = s,1,Set(TOCALL=${DB(SIP/${ARG1})}) exten = s,2,Dial(SIP/${TOCALL},60,tT) ... As I had no entry in the asteriskdb, so the SIP uri was

Re: [asterisk-users] Question about EuroBRI final 2 digits

and if so jumps to the right extension in the context. Overlapdial should be yes. yours christian gansberger www.accm.at On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number

[asterisk-users] queue autopause status

hi all! Does anybody know, how to get the status autopaused from queues. I want to display the status to the agent. I'm using asterisk-1.4.29.1 thanks chris -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Queue autopause

On Thu, Jul 9, 2009 at 4:41 PM, Miguel Molinammol...@millenium.com.co wrote: Christian Gansberger escribió: On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co wrote: Christian Gansberger escribió: Hi all! I want to autopause my queue member when

Re: [asterisk-users] Queue autopause

On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co wrote: Christian Gansberger escribió: Hi all! I want to autopause my queue member when they are not answering within 20 seconds, and the autopause should affect all queues they are member of, not only the queue where

[asterisk-users] Queue autopause

Hi all! I want to autopause my queue member when they are not answering within 20 seconds, and the autopause should affect all queues they are member of, not only the queue where the call was not answered. Is there a way to do that? The members gets dynamically added. I'm using asterisk

[asterisk-users] asterisk 1.4.21.2 a caller waited in queue, after connect to agent hears silence

configuration settings. the Agent/22 is not in use, there are no open channels and queue show is also reporting not in use. So why i m getting this Warning? i really don't know whats the reason of the silence. yours christian gansberger ___ -- Bandwidth

[asterisk-users] Call recording - posible to remove recorded file at the end of the call

the recording, the agents is pressing 0 for deleting the file or 1 for leave the file stored. thanks christian gansberger ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] TEST MAIL

sorry just a testmail to the list, becausemy last mail does not show up on the list. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] PRI CallerID - leading zero added

,Hangup When i make a call from extension 10 the macro is called with that: exten = _0.,1,Macro(dialout-isdn,10,${EXTEN:1}) Christian Gansberger ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] [Serusers] user meeting (beer drinking in Vienna)

Hi, so not too much replies yet, i would like to come and meet some asterisk-users. what about werkzeug-h ? Nice place to drink some beer :) cu Chris On 9/14/07, SIP [EMAIL PROTECTED] wrote: Curses! I just got BACK from Vienna yesterday. I should have stayed another week. :) N. Klaus

[asterisk-users] Hint a sip account

. But, is there another way to do this? christian gansberger -- ... Do you know what this is? Neither do I, I made it last night in my sleep, but I put a button on it ... (Goon) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [Asterisk-Users] IVR menus on different DIDs

So ...where can I get some help on my problem?thxchristian ___ --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] IVR menus on different DIDs

i didn't thought of that, and i tried it - it works when i use the Goto commandbefore i had one incoming context like [iax] which includes the different sub-contexts ofthe three ivr-menus - and the menupoints of the first listet included context were played. Interesting.thanks to you Filip !On

[Asterisk-Users] IVR menus on different DIDs

Hi all,So the first ivr-menu is reached with 123456-10 and the context is first-ivr[first-ivr]exten = 12345610,1,Answerexten = 12345610,2,Set(TIMEOUT(digit)=2)exten = 12345610,3,Set(TIMEOUT(response)=10) exten = 12345610,4,Background(${FIRST_IVR_MENU_SELECTION_MSG})exten = 1,1,Noop(First IVR-Menu

Re: [Asterisk-Users] IVR menus on different DIDs

Vexler [EMAIL PROTECTED] wrote: On 7/1/06, Christian Gansberger [EMAIL PROTECTED] wrote: Hi all, So the first ivr-menuisreached with 123456-10 and the context is first-ivr [first-ivr] exten = 12345610,1,Answer exten = 12345610,2,Set(TIMEOUT(digit)=2) exten = 12345610,3,Set(TIMEOUT(response)=10) exten

Re: [Asterisk-Users] pattern matching

hi,try thatexten = _[46-50],1,Set(LANGUAGE()=de)exten = _[46-50],2,CDR(userfield)=INTERNexten = _[46-50],3,MusicOnHold(0.5)exten = _[46-50],4,SIP/1000144|60|wWexten = _[46-50],5,HangupChristian GansbergerACCM, Vienna ___ --Bandwidth and Colocation

[Asterisk-Users] zt_pri-error

-2my zaptel.conf:loadzone=nldefaultzone=nlspan=1,1,3,ccs,amibchan=1-2dchan=3System: Slackware 10.0 Linux 2.4.32 with 1 HFC-S card (cologne). Thanks in advance Christian Gansberger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Echo cancel on HFC-S cards and CIDNum setting on outgoing calls

hi all! I'm running an Asterisk-box with bristuff-RC8n and 2 HFC-S cards. I m located in Vienna/Austria. I have the problem that on outgoing calls i hear my voice as a short echo (about half a second). This occurs not on every call. I tried some changes in my zapata.conf, with rxgain and

[Asterisk-Users] Echo cancel on HFC-S cards and CIDNum setting on outgoing calls

hi all! I'm running an Asterisk-box with bristuff-RC8n and 2 HFC-S cards. I m located in Vienna/Austria. I have the problem that on outgoing calls i hear my voice as a short echo (about half a second). This occurs not on every call. I tried some changes in my zapata.conf, with rxgain and

[Asterisk-Users] DID problem with calls from analog to ISDN

hi all, I have this anoying problem with Direct in Dial, when someone calls from an extern analog phone to my isdn number, Asterisk is not waiting for the DID numbers the caller dials after the main-number. Version: Asterisk 1.0.9-BRIstuffed-0.2.0-RC8n, with junghanns quadbri zapata.conf:

[Asterisk-Users] DID from an analog phone

hi all, I wanted to call my asterisk on the Zapchannel with mainnumber+DID number: it's ok for calls from handy and from sip i get the right extension(DID) on phone, but when i call from an analog telephone the DID number is not mentioned by asterisk. in my zapata.conf: overlapdial = yes

[Asterisk-Users] queue - ringing members in order

hi all, i m new to this list, I have a big problem, how to configure a Queue to follow the behaivor of: every incoming call should first ring the member listed first (in queues.conf) - then the second and so on. Is there a way to always start ringing with the first member of the queue? Here

Re: [Asterisk-Users] queue - ringing members in order

with next queue: SIP/1 SIP/2 SIP/3 and roundrobin, all calls stars with SIP/1 and with rrmemory first call starts with SIP/1, second call with SIP/2 and so on. Regards, srsergio -Mensaje original- De: Christian Gansberger [mailto:[EMAIL PROTECTED] Enviado el: martes, 30