[asterisk-users] mobile integration

hi all, a customer does want to have mobile integration within his asterisk based pbx - i have already an idea how to provide it - but wanted to ask here if someone already has an better approach - or other ideas... What the customer basically wants is to see the status of employees mobile

[asterisk-users] Attended Transfer does not release channels

Hi all, i have the following setup PSTN - routing server (asterisk 1.6.2.11) - IAX - callcenter asterisk 1.6.2.9 - SIP - agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does

Re: [asterisk-users] Attended Transfer does not release channels

2010/9/17 Olivier oza_4...@yahoo.fr 2010/9/17 Wolfgang Pichler wpich...@yosd.at Hi all, i have the following setup PSTN - routing server (asterisk 1.6.2.11) - IAX - callcenter asterisk 1.6.2.9 - SIP - agent Does work quit fine - then agent does have the abibility to transfer a call

[asterisk-users] call deflection support in chan_dahdi, libpri

Hi all, i do have the following setup ISDN BRI Line - openVOX Card/Asterisk 1.6.2.6/libpri 1.4.11.2 - Dialplan Dial DAHDI - ISDN PBX - ISDN Equipment The user on the ISDN Equipment das enable call forwarding - Teilrufumleitung / Call deflection - so that call will get forwarded by the telco

[asterisk-users] IAX2 Call Transfer

IAX2 channel - go to agent machine - agent does create outbound IAX2 channel - agent does transfer - now asterisk can handle it native - and will bridge channels on the first machine. Would this work ? best regards, Wolfgang Pichler

[asterisk-users] timing source problem

hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going

Re: [asterisk-users] timing source problem

old is your Digium card? MATT--- On 4/24/09, Wolfgang Pichler wpich...@yosd.at wrote: hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving

Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5

Hi, we are using version 2.0.4 (vicidialnow distribution) now for some time in productino - working quit nice. Is there any upgrade instruction out there - or will a simple yum update do the job in the feature. PS: On the astguiclient site you have April 3, 2008 - Released version 2.0.5 - i

Re: [asterisk-users] OT: Accountless, free, skinnable, browser based SIP client wanted

softphone functions using javascript. I would also suggest do add a call me on my phone option for users which will have problems with the softphone... There are already some cmpanies out there which will provide such a service - take a look at http://www.mexuar.com/ for example regards, Wolfgang

Re: [asterisk-users] Asterisk is not designed for University with largeuser base?

Hi, i do have a request for an installation with about 1800 sip extensions - as addon to a exisiting system - connected to it using qsig. The requirement here is also that the system should have SIP over TCP with TLS and SRTP (snom phones should get supported) I know there are patches out there

[asterisk-users] Looking for a patch cable for my SPA941 Phones

Hi all, i know this question is not directly asterisk related - but i have no idea where else to ask. We do have around 50 pieces of LinkSys SPA941 - these phones do have a 2.5mm plug connection - and we do have many many headsets we used with normal PC's before (so 2x3.5mm plug connection).

[asterisk-users] Java IAX Implementation

Hi all, i have now created a sourceforge project for the source made public by mexuar - you can find it at http://sourceforge.net/projects/javaiaxphone/ Take a look at http://lists.digium.com/pipermail/asterisk-users/2009-January/224730.html for more information about it. best regards,

Re: [asterisk-users] Vicidialnow

Hi, i have vicidial installed - and the company i have installed it for is using it. Take the VicidialNow install cd - install it - use it... If you need to combine it with your own system - then follow the step by step guid. regards, Wolfgang David @ULC schrieb: Anyone using VicidialNow

Re: [asterisk-users] dead sip channel

hi, try to set the rtptimeout value in sip.conf to a resonable value - so asterisk will kill the channels if it does not receive rtp traffic for the specified time regards, Wolfgang Jerry Geis schrieb: I have ran into a case using 1.4.22 where a SIP call to an asterisk client (running a

Re: [asterisk-users] Asterisk queues sending calls to members on the phone

Hi, take a look at the rininuse setting - if set to yes than you have the behaviour you described. If set to no - then you will get nearly the behaviour you want. Try upgrading to 1.4.22 if using queues - some concerns regarding the member status have been fixed there. regards, Wolfgang

Re: [asterisk-users] IAX Java Softphone?

Hi all, here you can find the demo site: http://www.yosd.at/corraleta/ I have also opend a forum for further discussion of the corraleta sdk... http://www.yosd.at/index.php?option=com_joomlaboardItemid=39func=showcatcatid=7 regards, Wolfgang Wolfgang Pichler schrieb: Hi all, thanks Tim

Re: [asterisk-users] IAX Java Softphone?

will also have no problem if you will create it as new project on sourceforge... (i think you would be the better project owner) regards, Wolfgang Tim Panton schrieb: On 15 Jan 2009, at 07:30, Wolfgang Pichler wrote: Hi all, thanks Tim and Mexuar for releasing this here... I have

Re: [asterisk-users] IAX Java Softphone?

Hi all, thanks Tim and Mexuar for releasing this here... I have already taken the source - and compiled a little java applet which is self signed to test the whole thing. I will put it on my site (and allow users to enter host/user/pass/Calling Number,Calling Name,Number to dial...) for demo

[asterisk-users] devicestate / inuse issue with 1.4.21.1

Hi all, we do have a callcenter system running with 1.4.21.1 - the agents are connected used sip phones. SIP accounts are configured using realtime (sip buddies) - and are configured with call-limit=1. It is operating just fine - but from time to time it does happen that an agent with an

Re: [asterisk-users] database queries from extensions.conf

Hi, you yould also use DBQuery (does only support mysql) - take a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+DBQuery (it does also contain a cdr backend to write customzied cdr entries to the database) regards, Wolfgang Klaus Darilion schrieb: Hi! What is the preferred way to

Re: [asterisk-users] database queries from extensions.conf

before - but i like it that way... Also the cdr backend with a customized sql querie is handy. regards, Wolfgang Klaus Darilion schrieb: Wolfgang Pichler schrieb: Hi, you yould also use DBQuery (does only support mysql) - take a look at http://www.voip-info.org/wiki/view/Asterisk+cmd

[asterisk-users] ISDN Cause Code 100, Bosch Integral Management Connection

Hi all, first off all - sorry for the cross posting - i did already posted this message to asterisk-dev - after that i realized that it isn't really a -dev related question - more a -users questions. So ignore it on -dev we have the following setup PSTN 3 PRI Lines --- Asterisk

[asterisk-users] zap channel media volume

Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk

[Asterisk-Users] SIP Callgroups

Hi all, at time i am trying to get a better idea of callgroups and pickupgroups (especially within the SIP Channel) A Pickupgroup is relative clear - everyone in the same pickupgroup may pickup a call And a callgroup does what ? - The same ? I thought that a callgroup would act like the ZAP

RE: [Asterisk-Users] TE410P PINS

hi all, Am Fr, den 02.07.2004 schrieb Steven Critchfield um 17:20: On Fri, 2004-07-02 at 09:56, Wolfgang Pichler wrote: hi, Am Fr, den 02.07.2004 schrieb Scott Stingel um 15:31: Hi- Only pins 1-2 and 4-5 are used, so one of the two cables should work. (probably the straight

[Asterisk-Users] TE410P PINS

hi all, i am getting crazy with my TE410P - it won't work now we already think that the only thing which can be wrong is the cable. At our telecom endpoint we have 1-2 tx, 4-5 rx So - which cable do i need to get it working Already tried a cross over cable 1-2 - 4-5 / 4-5 - 1-2 (with this cable

RE: [Asterisk-Users] TE410P PINS

, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Friday, July 02, 2004 4:42 AM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P PINS hi all, i

Re: [Asterisk-Users] Modified Prepaid Error

hi, Am Do, den 17.06.2004 schrieb oi geli um 1:13: I am trying to install the Modified Prepaid App. I have installed PostgeSQL, created the tables, etc. Make Install runs ok. The when I try to launch asterisk (asterisk -vgc), it fails to run. I get the following errors, 1st error:

[Asterisk-Users] pri with TE410P not working (Austria)

hi all, i am trying to get my TE410P (see previous posts) working in Austria (telekom Austria - i am still waiting for an answer for my questions). my /etc/zaptel.conf looks like span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31

Re: [Asterisk-Users] pri with TE410P not working (Austria)

Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43: On Thu, 17 Jun 2004, Wolfgang Pichler wrote: ... on the card i can see the two leds pulsing red (i think thats the yellow alaram - or i am wrong) ? Are you sure it is not a red alarm? That would indicate a loss of link. I think you

Re: [Asterisk-Users] pri with TE410P not working (Austria)

needs crossed wires, some needs straight wires. Crossed would be 1-4 2-5 i've already tried to use crossed wires - didn't worked either cheers Michael On Thu, 2004-06-17 at 10:28, Wolfgang Pichler wrote: Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43: On Thu, 17 Jun 2004

Re: [Asterisk-Users] pri with TE410P not working (Austria)

Am Do, den 17.06.2004 schrieb Peter Svensson um 10:38: On Thu, 17 Jun 2004, Wolfgang Pichler wrote: Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? The jumpers are on E1 - the cables should be ok (they are working with other

[Asterisk-Users] TE410P in Austria

hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already

RE: [Asterisk-Users] TE410P in Austria

Of Wolfgang Pichler Sent: Monday, 14 June 2004 8:11 PM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P in Austria hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3

RE: [Asterisk-Users] TE410P in Austria

: Is it the same as Telstra -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Monday, 14 June 2004 8:11 PM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P in Austria hi all, i've now

Re: [Asterisk-Users] Prepaid application error

hi, you have to also install the postgresql function's - the are included. There also exists a mailling list especialy for the prepaid application - take a look at asteriskbilling @ sourceforge best regards Wolfgang Am Mo, den 14.06.2004 schrieb [EMAIL PROTECTED] um 14:58: Hi, I successfully

RE: [Asterisk-Users] TE410P in Austria

Asynchronous Balanced Mode Extended what does this all means ? best regards Wolfgang Am Mo, den 14.06.2004 schrieb Wolfgang Pichler um 14:06: i've changed it now - but i still get the yellow alarm (i've changed it to: span=1,1,0,ccs,hdb3,crc4,yellow) this is my dmesg output

[Asterisk-Users] ast_log(LOG_DEBUG

hi all, at time i am working on the app_prepaid.so module (see: http://www.voip-info.org/tiki-index.php?page=Modified-Prepaid-Application ). In the source file a have many ast_log(LOG_DEBUG statements for debugging - but even if i start asterisk with asterisk -dvvvgc i wont get the

Re: [Asterisk-Users] Chan Capi Audio Quality Issue...

hi, what happens if you change the rxgain and txgain to something lower than 1.0 ? best regards Wolfgang Am Mo, den 31.05.2004 schrieb Stefano Finetti um 17:11: Hello all, I've just finished to install chan_capi with 3 AVM Fritz PCI cards. It correctly loads the 3 drivers, and * starts

[Asterisk-Users] CAPI / Channels

hi all, i have a probably very stupid question/problem. for testing purpose i am trying to get asterisk running with two isdn cards. I'd only like to here the demo sound when i call the number - but nothing works. The output of show channels is not showing any channel - should there be 4

Re: [Asterisk-Users] CAPI / Channels

hi, Am Do, den 27.05.2004 schrieb Thorsten Huber um 15:24: Hi Wolfgang, On Thu, May 27, 2004 at 02:49:51PM +0200, Wolfgang Pichler wrote: ... Here is my actual config (capi.conf) ... [interfaces] isdnmode=ptmp msn=073266 in my setup I'm using only the localpart as msn without

Re: [Asterisk-Users] CAPI / Channels

Am Do, den 27.05.2004 schrieb Julian Pawlowski um 15:29: Hi Wolfgang, I think everything is just fine because * says you that it has found 2 CAPI devices. (you can strip isdnmode=ptmp from your capi.conf anyway). thats the problem - it seems that everything is fine - but nothing works I

Re: [Asterisk-Users] CAPI / Channels

Am Do, den 27.05.2004 schrieb Thorsten Huber um 16:42: Hi, On Thu, May 27, 2004 at 03:53:08PM +0200, Wolfgang Pichler wrote: ... Here is my actual config (capi.conf) ... [interfaces] isdnmode=ptmp msn=073266 in my setup I'm using only the localpart as msn

Re: [Asterisk-Users] CAPI / Channels

can someone tell me how does the extensions.conf have to be - so that it is only configured to answer the phone ? (i've noe tested the card with capifax and capifaxrecvd - everything works fine) best regards Wolfgang Am Do, den 27.05.2004 schrieb Wolfgang Pichler um 14:49: hi all, i have

Re: [Asterisk-Users] CAPI / Channels

with that capi.conf it now works - thanx best regards Wolfgang Am Do, den 27.05.2004 schrieb Julian Pawlowski um 16:59: At time i am only testing / playing aready to learn more about asterisk. So I've simple installed it - installed the sample config files - installed two AVM Fritz Cards

[Asterisk-Users] calling card application

hi all, i'd like to know if is possible with asterisk / standard pc(server) / Wildcard TE410P (4 port T1/E1) to implement a calling card application. This calling card application should work as follows: - User's can buy credits at out company (this credit gets stored in a database - with the

[Asterisk-Users] would it be possible to...

hi all, i'd like to know if it would be possible with asterisk (and which hardware would i need) to implement the following (or is it not possible with asterisk - but possible with ...) I'd like to set up something like a Mobile to Conventionel Network Gateway - so that users (with there Mobile

Re: [Asterisk-Users] would it be possible to...

Die GSM Tailnehmer wählen nicht die eigentlich Auslandsnummer - sonder unsere SIP Gateway Nummer + als Durchwahl die Auslandsnummer. Unser SIP Gateway sollte dann die Durchwahl(=Auslandsnummer) wählen und das Gespräch verbinden. So dachte ich mir das auf jeden Fall - obs möglich ist weiß ich nicht