similar to: Queues +Exiting

> -----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 >

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

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

> 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

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, 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:

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 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

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

> I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. > All seems to work normally with realtime voicemail, reads vmbox > parameters from the db fine. When I try to change the password, > asterisk operates normally, "enter new password" ok, "re-enter new > password" ok, "password has been changed" > > There are no entries in

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

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''

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

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

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

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

Hi there, see subject. I'd like to be able to use the vmbox prompt of VoiceMailMain2 and use 1234 and 4321 to point to the same mailbox. Will it be sufficient to create a soft link for 4321 --> 1234 in /var/spool/asterisk/default or will I get myself into horrible trouble? Background: I like to be able to map certain functions ("boss", "peasant",

I've setup the voicemail that auths against the mysql db. Now, everything works ok, except voicemail() calls fail with Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517 leave_voicemail: No entry in voicemail config file for '' all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the

Hi All, I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, "enter new password" ok, "re-enter new password" ok, "password has been changed" There are no entries in the mysql.log setting the