search for: gwsnettech

...gh the include statement, asterisk has to look first in capi-in-sub, there it should find this extension: exten => 95562315,1,DoSomethingAnybodyWouldExpect() ... and follow the dialplan under capi-in-sub since a valid extension was found. What's wrong, any ideas? Regards, Guido Hecken gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 K?nigswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:guido.hecken at gwsnettech.de

I have a patton 1400 setup to handle the bri interface. As a trixbox user, I wanted a sip trunk rather than having to re-compile bri support into trixbix. Anyway, I have it working now so that asterisk can make calls and they are passed properly to the telephone network. Incoming calls however are another matter. I have (after turning on cli debug in the 1400) determined that its getting stuck

...et the user "find" their keys on the phone, the double ammount of time ( 2 x 5 secs) you have to wait before a call is parked and the parkposition is announced, is really too long. Did I miss something in the documentation? We are using SVN-branch-1.4-r96449. Regards, Guido Hecken gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 K?nigswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:guido.hecken at gwsnettech.de

...edCall(${EXTEN}) in features.conf: [general] parkext => 800 parkpos => 801-805 context => parkedcalls parkingtime => 60 ------------------------------------------------- We are using SVN-branch-1.4-r96449 and other, older versions of asterisk Regards Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 K?nigswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web <http://www.gwsnettech.de/> http://www.gwsnettech.de <mailto:guido.hecken at gwsnettech.de> mailto:guido.hecken at gwsnettech.de ------...

Hello, I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a different port. It is my understanding that I just need to set the port in sip.conf

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

Here is a part of a working sip.conf for Asterisk with SNOM190 Phones. Try using host=dynamic and have a closer look at the configuration in the snom 190. Also, try using dtmfmode=rfc2833 . [general] realm = hallinux2.gwsnettech.local port = 5060 bindaddr = 0.0.0.0 context = default disallow=all allow=alaw allow=ulaw allow=gsm register => 081503:xxxxxx@sipgate.de/081503 language=de tos=0x04 [6301] type=friend username=6301 secret=xxxxxx host=dynamic disallow=all allow=alaw allow=ulaw allow=gsm dtmfmode=rfc2833 ma...

> We have our Asterisk server running smoothly with a SIP BRI gateway > for inbound calls. However if the Internet connection goes down and a > DNS server becomes unreachable Asterisk basically does not function. > By this I mean it does not answer call coming in from the gateway > (which is on the local LAN) and you can't even reload it - just hangs > there. If I change the

Hello, I am trying to setup an Asterisk GUI with the help of astman(please visit http://astman.sourceforge.net/am-user-guide.html). I have installed astman and currently assessing my GUI using; http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl I am trying to get the menu options in my GUI to work but to no avail. Currently my parameters are set to; Asterisk Install Directory:

Anyone have a Snom they're happy with? How did you manage that? :) I have a system of: Asterisk 1.2.3 2 Wildcard TDM400P Rev I and E/F 1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841 ~15 Grandstream 101 Everything (currently) is on the same network, not a router to be seen between any two. Also everything, except the snom, is working sweetly. The main problem is ECHO.. awful echo and

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

Hello everyone, I am trying to install windows xp as a guest OS. It runs through the installer just fine, but every time I reboot, it keeps booting back to the installer. Here is my config: ------------<code>--------------------------------------- kernel="/usr/lib/xen/boot/hvmloader" builder="hvm" memory=512 device_model=''/usr/lib/xen/bin/qemu-dm''

We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this

Hi ! I have install the xen4+centos5 on my server following the post of the kernel version is 2.6.32.27 which is download from git://git.kernel.org/pub/scm/linux/kernel/git/jeremy/xen.git But after I reboot the server ,and run the "ifconfig", whose output likes the following [root@localhost scripts]# ifconfig eth0 Link encap:Ethernet HWaddr 00:15:C5:EB:96:C3

hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

> When I do click on the phpconfig.php link from > http://ip-of-machine/phpconfig/, it returns a page with the actual > contents of that file (phpconfig.php) and doesn't load the page. See some > of the output below; Try a simple php-script in this directory. Something like this, name it test.php <?PHP print (test) ?> If it does not print "test", and you see the

before inventing the wheel once more, I would like to ask the list Problem: the dial command does it's job by exten => _X.,1,Dial(SIP/${EXTEN}@192.168.161.135,60,tTr) now, if the given number (extension) is one of these +49(2244)870663 or +49(02244)870663 the dial command fails, since it cannot handle "(" ")" and "+" My idea was a regex, I use normaly

Hello List, has anybody seen (and perhaps decoded) this messages in Asterisk debug: ... Jun 30 17:46:23 DEBUG[11033]: Didn't get a frame from channel: SIP/192.168.1.30-ed86 Jun 30 17:46:23 DEBUG[11033]: Bridge stops bridging channels SIP/5602-c545 and SIP/192.168.1.30-ed86 ... This happens always, when a call get's dropped and today we had more than 10 calls which ended this way. The

Which kind of transfer do you use? Try using the # transfer. Hope that helps.. Guido Hecken -----Urspr?ngliche Nachricht----- Von: Diego Magalh?es [mailto:diego@redetaho.com.br] Gesendet: Mittwoch, 2. Februar 2005 17:21 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] AgentLogin / AgentCallbackLogin transfer problem Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and