search for: christian08

Displaying 20 results from an estimated 24 matches for "christian08".

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2008 Apr 06
3
Need help with Cisco 7960
Hello all, I need some help with my Cisco 7960 enabling TFTP. Does anyone know what numbers to press in the menu? Or can I enable this through telnet? Many thanks, Christian
2006 Nov 12
3
Looking for a simple TFTP server for Linux
Hi, I am looking for a TFTP server that is easy like the tftpd32 for Windows that I have been using. Just want to start it with a command and my Cisco can connect and retreive the config files from it. Many thanks, Christian
2008 Dec 16
5
Installing Asterisk v1.6 on Ubuntu Intrepid?
Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c -> manager.o manager.c: In function ?action_getvar?: manager.c:1732: error: ?SENTINEL? undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears
2015 Jun 08
3
Fritzbox 7490
Hi, Sorry if off topic, but is anyone here on this list using it? I am currently searching for a good router for my home network wich supports SIP. Many thanks!
2005 Aug 09
2
Asterisk and Wave files problem
Hi, I'm recording wave files but I cant get Asterisk to play them, only if they are in 8000 Hz. What is the maximum sample rate Asterisk can handle? I have been using 16-bit 44.1, 22050 and finally 8000 kHz. Many thanks, Christian
2006 Mar 06
1
Asterisk on MacOS?
Hi, I am just curious, does anyone know if I can run Asterisk on the Mac? I've read something that it should be possible, but cant find an eventual download page or what is supported. And also if the Zaptel driver is supported as well as Ztdummy. Many thanks, Christian
2006 Nov 07
1
Why dont my messages get through
Hi, My messages to the list don't get through. This must be the tenth message i am trying to send! Please ignore this test message.
2006 Nov 08
1
talking caller ID
Hi all, Lets say I have my incoming calls transfered to my mobile phone. When a call comes in, Asterisk will answer the call and ask the caller to hold the line while the call is being transfered. I know how to do this, but i dont want the caller to hear me answer the mobile phone. They can hear some music on hold. When I answer Asterisk will read the callerID to me and I can then decide if this
2011 Jun 12
2
A question about Caller ID
Hi all, Sorry if this is a little off topic, but I just want to know a thing here. What system is used for sending out the caller's number in the US? Here in Sweden we use DTMF to send the number out. I just need to know what is used in the US since I don't think I will be able to use an American caller ID device over here. Many thanks for any info, Christian
2005 Aug 04
1
Callback question
Hi, I'm interested in a callback feature where I can dial my Asterisk, then hangup and Asterisk will call me back and I can then place phone calls or whatever I want to do. And also, if I've got voicemail I want Asterisk to call me back as well. Are there any scripts for this available? Any help would be apreciated! Best regards, Christian
2005 Aug 09
2
Playing GSM files in Windows?
Hi, Is there any program that will play GSM files in Windows? I'm going to translate the audio files and need some player to play it with. All the best, Christian
2006 Nov 09
2
Latest Debian and latest zaptel
Hi all, Since i cant get latet beta of zaptel installed on the latest test version of Debian with kernel 2.6.17-2-686 can someone who is using debian give me some tips on how to get it working and installed? Many thanks, Christian
2006 Nov 13
1
Music on hold question
Hi all, Using the latest 1.4 of Asterisk. I have noticed that the music on hold files are in wav, isn't mp3 supported anymore? Many thanks, Christian
2006 Nov 15
2
Question about TFTPD server
Hi all, I have installed this package onto my Debian and placed the files i want the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to work. Are ther any special settings I should do to this server? Many thanks for all your help, Christian
2006 Nov 18
0
Cant record phone calls
Hi all, I tried enabling the automon feature, but for some reason it doesn't start recording when I or the caller press *1 when the call is iniciated. exten => 200,1,set(DYNAMIC_FEATURE=automon exten => 200,2dial(SIP/phone1,30,rWw) Any thoughts? Many thanks, Christian
2008 Feb 28
1
Problems with setting up Zaptel
Hi all, I've just got an OpenVox A400P card with 1 FXO and 1 FXS module and I am just trying to get it working. But no luck as of yet. In /etc/zaptel.conf I've set the following options: fxsks=2 fxoks=1 loadzone=se defaultzone=se And in /etc/asterisk/zapata.conf I've not sure what to set exactly. For example, under [trunkgroups] what to specify there? Under [channels I set something
2008 Feb 28
2
Problems with removing zaptel
Hi all, Using the latest test version of Debian but when I have done modprobe -r and removed a few of the zaptel modules some of them cannot be removed. The other module is in use. Also if I reboot my system they're all loaded again. Any thoughts? Many thanks, Christian
2008 Sep 28
0
Need help with Cisco 7960
Hi all, This might be a little off topic, but I need some help with this phone and hopefully someone on this list is able to assist me. When establishing a conference call I am not able to hang up the call I connected to my original call. I have tried pressin ghte conference button, but nothing happens. Any help would be apreciated, many thanks! Christian
2008 Oct 31
1
Asterisk installation
Hi all, I've just installed the latest v1.6 release of Asterisk. First, I isntalled libpri. Then i installed zaptel with make config at the end of the isntallation as I usually do. Then I installed Asterisk. However, there is no zapata.conf file in /etc/asterisk. I isntalled the sample configuration files. Any tips? Many thanks, Christian
2009 Mar 25
0
A Cisco 7960 question
Hi all, Is it possible to have the Cisco 7960 dialing a SIP address to a service that you are not registered with, for example: sip:xxxxxx at xxxxx.org and asign that to some spee dial button? I have heared that it should be possible to define in the dialplan.xml file, but not sure. Any info is really apreciated, many thanks! Christian