similar to: SNOM Softphone on windows 2000

Displaying 20 results from an estimated 3000 matches similar to: "SNOM Softphone on windows 2000"

2006 Dec 07
1
Asterisk accepting calls to fast
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
2006 Mar 30
1
Benchmarking an Asterisk Server with 14k users
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
2006 Jun 28
2
2 or more ISDN cards: which comes first ??
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
2006 Apr 19
2
Unable to allocate socket: Too may open files
Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create socket Apr 19
2007 Aug 08
3
Siemens Openstage & Asterisk ?
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
2008 May 14
2
Setting CallerID UNKNOWN on an outgoing call
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 --
2008 Jun 02
1
Why doesn't Pickup() work??
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
2008 Oct 31
5
twice normal beep before busy tone ??
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 =>
2006 Jun 10
1
Voicemail records nonsense, but record() works (??)
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 =>
2008 Mar 05
4
Problem between Asterisk and an Aastra 57i
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:
2008 Jan 13
2
Question about queues and the definition and agents
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
2007 Dec 03
4
Soundcard necessary on an asterisk server to get output of playback()??
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]
2007 Oct 21
1
Sometimes echoes & Asterisk sometimes connects too early
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
2007 Nov 12
3
No sound from playback and voicemail
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
2006 Jun 12
2
Bug in Voicemail ??
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 =>
2008 Feb 15
0
Question about DIALSTATUS NOANSWER
Hi, according to the wiki the value NOANSWER for the channel variable DIALSTATUS means: No answer. The dial command reached its number, the number rang for too long, then the dial timed out. In out dialplan we grap all these events with exten => s-NOANSWER,1,Playback(sometext) exten => s-NOANSWER,2,WAIT(1) exten => s-NOANSWER,3,Hangup() The dial commands for internal and external
2010 Mar 21
0
dahdi_monitor doesn't show data on RX & TX: broken card or cable?
Hi, on one of our clients asterisk server we have the problem that you hear nothing on external calls. Here are the details abount the system: Asterisk 1.6.0.22 DIGIUM Wildcard B410 quad-BRI card (rev 01) dahdi-linux-complete-2.2.0+2.2.0 I have setup the following test extension: exten => 9216992,1,ANSWER() exten => 9216992,2,WAIT(2) exten =>
2007 Dec 18
0
Doorbell Siedle DCA 612 and Asterisk?
Hi, has anyone already set up a configuration between the doorbell Siedle DCA 612 and an Asterisk Server? I have used a Grandstream HT 286 to connect the doorbell and the asterisk. When I press the button, the phone ring and when I pick up the call I hear a beeping. At the door I hear nothing. According to the wiki, this doorbell should work with Asterisk, but I haven't found a dialplan
2008 Feb 05
1
Mistake in the wiki's description of cmd Pickup() ?
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.
2007 Nov 12
0
No sound from playback and voicemail (Atis Lezdins)
Hello, >> > 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 =>