similar to: queues and the '*' key

>From voip-info.org: Queue(queuename|options|optionalurl|announceoverride|timeout) 'optionalurl' allows you to send a URL to devices that support it. Does anyone have details on the "devices" that support the optionalurl method of the Queue application? I am wondering if there is a softphone that supports this. The only thing that seems to happen is the

Hey all, First of all, thank you for the help I've gotten on this list in the past. Very helpful, and I apprecaite it. Now, what I'd like to do is send a message to my snom 320s. I'd like to have the message display regardless of what the phone is doing. I have been trying SMS, or the sipsak method on the wiki but I have had no luck thus far. Does anybody have this working,

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Hi folks, For the longest time, I've had a problem changing or modifying ACLs from my window clients. Whenever I tried, I'd get this in the logs: [2004/07/29 12:36:26, 0] smbd/posix_acls.c:create_canon_ace_lists(823) create_canon_ace_lists: unable to map SID S-1-5-21-1292428093-651377827-xxxxxxxxx-1333 to uid or gid. I could change the ACLs using getfacl/setfacl, btw. After a

hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info mailbox=1234@default disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind

Hi group. I have huge problem. My pickup exten #8 isn't working. This is what I have configured. pbx*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 #8 In sip.conf I have callgroup=2 pickupgroup=2 For called party and same for person that is trying to pick up the call. The person that is trying

Hi, I have several queues configured at my call center for different support levels. Today, something weird happened: - A client called queue 1 and was answered by an agent - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf - The user transferred the client to another Queue, by using the second channel and the XFer key of her

This evening I posted a new set of Digium G.729 codec modules to our FTP server and web site, for Linux x86 and x86-64 processors. They were built using GCC 4.0.1, and they now report the processor they were optimized for when they are loaded. The previous x86-64 module required a non-standard Asterisk binary configuration, so this was corrected. In addition, there was only a generic version

Greetings everyone, I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy allows you to communicate with multiple Asterisk boxes from a single point of contact using a variety of I/O formats, now including support for XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. Astmanproxy is

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

] Hey ho, A few days ago we released the linux version of the phone, today we are very happy to have the mac version ready for a little field test. Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php At the same time, we also put a newer version of the windows and linux versions online. Let us know how you feel about it, a more mac look (brushed metal) is coming.

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes

i have quake v1.06 installed and running fine under windows. however, running it in wine gives a bunch of errors. see below: prophet% wine --winver win98 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD. Try --winver nt40 or win31 ! prophet% wine --winver nt40 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD.

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

Hi, Is there any way to run a command (AGI script, whatever else) at that moment when the call that was in the queue is being bridged to a specific agent? An examples of what I would want to do with this is, for example, have Asterisk ask the caller for his 4 digit customer ID before being put in the Queue. Once I know who the caller is being connected to (which agent) I'd run a

Hi, I've dug a bit through the wiki and the mailing lists, and haven't really seen anything like this, but there must be someone out there doing this. Basically, there is a fax line that I don't want to answer inbound, but I want it available to do dial out from. Right now, we are using a busy wait around the ringing line, but I was hoping for something that might be a little more