similar to: queue/agent and macros?

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the

Here is a section of my dialplan (macro) exten => s,200,Wait(1) exten => s,201,read(holdopt|screen-onhold|1) exten => s,202,GotoIf($[${holdopt} = 1 ]?4) exten => s,203,GoTo(200) it's simple really it loops telling you the caller is on hold until you press 1 and then it sends you off to another area. The problem right now is that if the read() times out i get these warnings...

Hi I would like to send a text to the called person when he picks up the phone before the call gets connected through. Is there a way to do this? Example: I'm registered to multiple SIP providers. They come in to a context each and then get through to my phone. Now I would like to send myself an announcement about from which SIP provider this call came from. -- Beno?t Panizzon,

I remember reading some on here a week or so ago about one of you having a problem doing firmware updates over PXE. Right now i'm trying to do the same and update the firmware on a intel SE7520DB2 board. The firmware utility just stays at reading file ..... 0x00000000 (0%) And doesnt change, the utility is 'AMI firmware Update Utility - Version 1.32' Just wondering if i could be

Hi folks! Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double checked if I've forgotten to remove an

Anybody know the proceedure to factory reset the a 7960 phone running 6.3 SIP software? I've tried holding # when booting the phone and nothing, i can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. Also **# doesnt work either.. -- ~Shaun

Hi, I have one Q: why doesn't work h-extension in macro? I want to setup fax macro, to send tif-file after fax channel hang-up, but h-extension doesn't work in macro, so I must put them to global context. But in this setup, exten => h,1,system(/var/lib/asterisk/bin/mailfax ${FAXFILE} ${EMAILADDR} "${CALLERIDNUM} ${CALLERIDNAME}") mailfax binary will be executed after any

I can't seem to get a Windows XP Media Center Edition system to see a very simple print share. [global] workgroup = HOME realm = SHAUN server string = Samba Server security = SHARE encrypt passwords = No log file = /var/log/samba/%m.log max log size = 50 dns proxy = No wins support = Yes valid users = shaun admin users = shaun read list = shaun write list = shaun printer admin

Okay my smb.conf file is correct. Firewall is configured correctly to allow Samba on local network. smbd and nmbd are running. Her computer still cannot connect to my printer share. I read my Samba version wrong its actually 3.0.23a-1.fc5.1, here is my smb.conf: [global] workgroup = HOME realm = SHAUN server string = Samba Server security = SHARE encrypt passwords = No log file =

I used the xensource install script for RHEL4 and after trying to boot on the new kernel the system just continually rebooted. I appended a noreboot to the xen line and now i''m seeing the following... Freeing unused kernel memory: 184k freed dm_mod: no version for "struct_module" found: kernel tainted. device-mapper: 4.4.0-ioctl (2005-01-12) initialised: dm-devel@redhat.com

I'm looking for somehting like the standard house hold linksys/dlink router. Basically it needs to have at least 1x100mbit port, wireless G capabilitys and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2 which appears to do what i want. Anybody use this? Does it require vontage's service? I'm looking for any recommendations. Thanks -- ~Shaun

I've been trying to get the polycom 550 phones to show a idle display bitmap but have not been successful. Anybody have any experience with this? The manual gives instructions (http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf) but they do not seam to work. So far i've done the following in my

I guess this really isnt the right place to be posting this it's probably the best place for me to find the answer. I would like to know if any of you are doing windows 2000/2003 installs over PXE and if so how you had it setup. I've done some searching on google and mostly found info about somthing called RIS or RAS or whatever.... it required a windows box and thats not a option.. I

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=

Hi I have a problem with the queues on Asterisk. The setup is Asterisk@Home v1.0 with Asterisk 1.0.7. I have 1 queue (4500) set up, with leastrecent strategy. There are no agents configured in this queue. Agents log in by dialing 4500* on their phones. All incoming calls are sent to the queue. Calls wait 120 seconds in the queue, and are then sent to voicemail extension 310. My problem is

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call home from my cell phone. This is what I tried in the context

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:

Hi All, I am trying to achieve something pretty simple. I have a samba share and I would like all new directories and files within that share to have 775 permissions. My config for that share is below. Whenever I create a dir I get 755 permissions. I have set every force permission their is. Both configs below produced the same directory. I feel like this should be very easy and my settings are

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to