similar to: Strange dropped calls

Hi list, sorry if this has been discussed in the past, but I couldn't find anything wise about it. Since we had some trouble with the builtin hold function of some (all?) SNOM 320/360 phones, we decided to use the call parking feature in asterisk instead. Assume, a call comes in with CALLERID(num) 1234567 for extension 10. Extension 10 parks this call into 801, dials extension 11 and

Hi, anyone out there with the same problems and a possible solution to the following? The functions callparking and hold use the same transferdigittimeout in features.conf. While I think 3 to 5 seconds are enough to let 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

> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking >

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

> 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

Hmm i found the problem... I?m using a Grandstream BT100. The transfer just works in a queue if I first acknowledged the call using the # key, and then press the TRANSFER key in the Grandstream. In the asterisk console I receive a: -- SIP/4002-4563 acknowledged Then I can transfer the call... Weird because i?m using ackcall=NO in agents.conf ... Diego Magalh?es diego@redetaho.com.br +55 24

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:

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

> 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

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

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:

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

Hello List, we have 3 Asterisk boxes running under Fedora Core 2. Every box hangs/crashes from time to time. These installations are image based, means we made an image from our testserver with an image tool, which is able to manage ext3 partitions and deployed it to different server hardware. These servers run very stable and I could not find any failures in the logs. As these crashes appeared

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

Trying to add a new samba 4 domain controller to an existing Windows 2003 domain. There are two existing domain controllers: dc1.home.aaronson.com and dc2.home.aaronson.com. As you can see below, samba 4 dies during the join. I am stumped. Dcdiag throws no errors on the existing controllers. Any ideas/ ubuntu at sulu:/usr/local/samba# sudo bin/samba-tool domain join home.aaronson.com

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

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''

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

Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible