similar to: When to use 'Answer' and when NOT to...

Hello, I have a TDM400P with 2x2 configuration of FXOs and FXSs. I set a test extension of '444' to dial out a specific zap trunk and call a local #. First time I call out to '444' everything works fine. If I hang up the call, and within 10 seconds dial the same number again, I get a fast busy. Seems it isn't letting go of the trunk or something, and I don't have a

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten =>

I have a Polycom 500 and would love one of the line-appearance keys to show me if a certain person/people are on the phone upstairs. This 'hint' priority seems to have little-to-no documentation. So, if anyone out there has a clue about this, here are a couple of questions: Can you 'hint' Zap and IAX2 extensions? Can you concatenate extensions together? (i.e. exten =>

Is there an easy (or even a hard) way to save to the CDR a userfield value with the call's codec in it? Chris Coulthurst chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051007/7ab00352/attachment.htm

Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? I know this has to work somehow. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/4a3c3025/attachment.htm

Hi, I'm trying to write a tool, which shows me the state of the current calls. For this purpose I'm reading from Pipe the Asterisk output and parse it... asterisk -vr | mytool However, the problem ist how to get the information about who got this call in the group. The Zap channels are assigned dynamical. Only thing I can see which channel is connect to the caller but not who is

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Is there a way to make an outside call hear "The person at phone number XXXX is unavail", but when an internal extension calls another extension, they hear "The person at extension number XXXX is unavail"? I swear I've read this somewhere before but I'm not typing in the right search. I probably found it before by complete accident. Of course, we want the outside

Get the new firmware - it's supposed to have changed the callerid display presentation to include name and number. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Coulthurst Sent: Friday, May 20, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Displayed CallerID on

Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? All I get, no matter what I've tried is "Unable to upgrade firmware". tftpd is working because the dialplan freshens, and I have aastra.cfg whatevermacaddressfile.cfg and firmware.st in /tftpboot Am I missing something stupid? Is there another way to upgrade it? Chris Coulthurst chris@shuksan.com

Does anyone have a quick-n-dirty context to implement *66 automatic callbacks? I have a few people who like to have no call waiting on their phone (can you really blame them?) It would be nice to have something like *66, and also like 'Camp On', but instead of waiting something like 30 seconds, monitor the channel until it becomes available, then immediately ring back your phone to

Can anyone explain the Polycom Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to it? I've seen vague references to MSN Messenger, and somehow that's mentally disturbing. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Lately when I issue a 'reload' from the CLI, I find that it will sometimes hang forever, completely locked up. I can press enter and see the CLI prompt move, but no commands are taken. "top" shows asterisk eating everything up: PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 20669 root 25 0 10068 9.8M 5392 R 88.4 1.9 1:02 0 asterisk

I'm trying to find a way to generate an 'internal extensions' tonelist but I can't seem to find anything on how to do this. My idea was to start a Playtones(intercom) tonelist and not indicate ringing to the line (dead air). But then, somehow StopPlayTones needs to be run once the ringing telephone picks up. This seems like a dirty way to do this. I envision an option to the

Does anyone know how to change the display format of caller id on the screen of a polycom 300/500/600? When I call FROM my 'shop phone 203' TO my 'office phone 201', a Polycom 500, it only says 'Shop' as the calling party. More specifically, the two lines look like this: Incoming call from: Shop I'm looking for a way to make it use both lines for caller id,

Has anyone ever tried to roll out a *66 auto-callback-redial feature on asterisk? I'm sure that implementing this for outbound Zap calls would be a nightmare, but what about something easier, like internal extensions? On my old Panasonic key system, it used to be such that, if the called extensions were busy, you could press 6 while hearing the busy signal, it would beep twice and hangup.

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Hi, I'm having trouble with dialing out on analog lines. Asterisk can't seem to detect "answers". I have two zap groups. Group 1 is connected to an external analog PSTN provider. This group seems to work fine, especially with "answeronpolarityswitch". Group 2 is a group of "GSM gateways", ie. devices that host SIM cards so that you can dial from any PBX

Ever since I upgraded my Polycom 500 to the newest sip.ld (kept the old bootrom), when I dial things like "*98" for voicemail, the screen shows "9*8" and doesn't dial my voicemail system! Is this user error, or errors in the new firmware? Chris Coulthurst chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I try and park a call by transferring to 700, it audibly says to me "701", and then instead of hanging up with me, it puts me on hold. The only way to park the call is to send it blind to 700, but then I wouldn't know which parking spot the call is in! Before I send any .conf files to the list, does anyone recognize this behavior, and have a workaround? Chris Coulthurst