similar to: XP-Clients loose file permissions

Hi, I have a strange error, that I haven't yet found in another posting on the archive. Here's my sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls callerid=No CallID [stefan] type=friend secret=blah username=stefan

Hi, the german telco Colt Telekom has assigned the phone number block 56830-xxx to one of our customers. In the diaplan we have setup extensions like the following ones: exten => 56830910,1,Answer() exten => 56830910,2,Dial(SIP/bduerring,10,tr) exten => 56830910,3,VoiceMail,u20 exten => 56830910,4,hangup exten => 56830910,103,VoiceMail,b20 exten => 56830910,104,hangup exten

Hi, I spent some hours working my way through the WIKI and a number of other documentations, but after all, three questions are still left: 1. I'm using a Fritz!Card with the i4l driver - no problem at all, my Grandstream BT 100 rings when I diall a regular phone number. Is there any need for me to configure the zapata.conf for the ISDN card to get a special asterisk feature or is this

Hi, a number of people reported problems when trying to join the ADS. The error message mostly looked like "krb5_cc_credentials not found" or "ads_connect: No such file or directory". I got these errors on my SuSE 9.0 (Samba 3.0.2a, heimdal-devels installed, Samba configured with --prefix=/usr/local/samba3 --with-ldap --with-ads) system, too. Well, now I'm a

Hello, today I tried to compile Samba 3.0.2a on an AIX 5.2 system with Visual C++ 5.0.2. configure failed with the error message "compiler cannot create executables". I have read this message before on a linux system and was able to fix it (if I remember correctly) by installing cpp. But cpp is already installed on AIX. Was anyone of you able to compile the current samba version on

Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten => 83086921,1,Answer exten => 83086921,2,Dial(SIP/stefan,5,r) exten => 83086921,3,VoiceMail,u111 exten => 83086921,4,Hangup exten => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 =>

Hi, is anyone on the list using the Siemens Openstage phones together with asterisk? If yes, is it possible to use the programmable keys of these phones together with Asterisk? Thanks for any hints, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93

Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan --

Hi, I have a strange problem with our Asterisk installation. Outgoing calls are handled by the following lines: exten => _0[2-9]X.,1,Set(CALLERID(num)=09999403${CALLERID(num)}) exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} = 0999940321]?099994030:${CALLERID(num)})}) exten => _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr) exten =>

Hi, I'm using an Aastra 57i together with Asterisk 1.4.13. The 57i is configured for call pickup as recommended by Aastra. When the LED flashes, I press the corresponding button and the display tells me "Call not possible". In the CLI is see the follwoing output. Why isn't the call transfered to user2 and what does "No target channel found for 31 mean"? User2 can

Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten => 202,1,ANSWER() exten => 202,2,PLAYBACK(tt-monkeys) exten => 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [202 at local:1]

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

Hello, I have read the articles on echo cancellation (http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a solution to my problem. We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M PCI (current driver from EICON) and some SNOM 300/360. There are few clients where we recognize echoes on both sides when we call them via ISDN. With some of these clients we

Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten => 83086921,1,Answer exten => 83086921,2,Dial(SIP/stefan,5,r) exten => 83086921,3,VoiceMail,u111 exten => 83086921,4,Hangup exten => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 =>

Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] ??? exten => 99,1,ANSWER() ??? exten => 99,2,PLAYBACK(tt-monkeys) ??? exten => 99,3,HANGUP() The phone

Hello, I have setup an asterisk server with two EICON cards, a 4BRI and 2FX. How do I know, which card is the first, so that I can setup capi.conf with the right entries? Thanks for your help, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93

Hello, one of our clients is currently testing three installations: - Cisco Callmanager 5 - Siemens HiPath 8000 - Asterisk To get an impression how these system behave under heavy load, he's going to use an ABACUS 5000 system (http://www.spirentcom.com/analysis/technology.cfm?media=7&WS=325&SS=111&wt=2) so simulate 14k users, calling each others, in other words 7k simultaneous

Hi, I'm currently trying to connect an Aastra 57i to our Asterisk Server. The strange thing is, that altough I have definitely entered the correct IP address of the server, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1:

Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten => *8200,1,Pickup(200) Asterisk tells me that the highest value for the Pickup command is 63. Wenn I enter the number of a callgroup instead of an extension, I can pickup the call.

Hello, I have read the articles on echo cancellation (http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a solution to my problem. We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M PCI (current driver from EICON) and some SNOM 300/360. There are few clients where we recognize echoes on both sides when we call them via ISDN. With some of these clients