similar to: How is it best to initialize specific SIP peer settings

The setvar command below works fine in iax.conf and in sip.conf but not here in zaptel.conf. I need it to retrieve info from the AstDB. Advice is apreciated, can't seem to find an answer. ; define channels group=1 context=longdistance_users signalling=fxo_ks ;FXO Sig for Phone callerid="John French" <103> mailbox="101" callwaiting=yes threewaycalling=yes

Hi all, I used to configure each of my sip clients with a unique identifier via setvar. These clients were all configured as "friends." However, now that I've got some Polycom phones, which MUST be "peers," I am unable to define this variable. For example, this works: [friend-client] context = default accountcode = pcc type = friend username = username secret =

Hello, Will I be breaking something if I have two users share the same home and/or mail_location (maildir format) in dovecot? Mail will only be delivered to the primary user, the other would be used every now and then to login via imap (webmail). This would be on a CentOS 7 box running dovecot from the packaging system, i.e. dovecot-2.2.10-8.el7 currently. The reason I'd like to do this

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

I'd love to understand what the function of the peer list returned by 'sip show peers' is for, especially when Realtime is used. If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As each peer (phone) registers, or a call is made to the peer, Asterisk adds them to the list returned by 'sip show peers'. Correct? Apparently Asterisk doesn't

Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload

hi everyone I''ve got two machines here: a Powerbook and a Linux box running MySQL. The MySQL user privileges seem to be configured properly, e.g. I can connect to a database using "mysql" from the Powerbook, alter the tables etc without any problems. However, I can''t get Rails to connect to the same database. (just a simple app with one model called "Test"

Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

Hi Everyone, We want to be able to momentarily or temporarily provide CONGESTION or DE-REGISTER a SIP PEER to asterisk through WEB GUI. I don't want to indulge into dial-plan and write changes to .conf file every-time. Is there any way that a SIP PEER can be de-registered for an amount of time or maybe deactivated? or there isn't such facility available in asterisk? Thanks --------------

Skipped content of type multipart/alternative-------------- next part -------------- asterisk1*CLI> soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' --

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060

I am using this site (http://www.digium.com/asterisk_handbook/manager.html) to access port 5038 (manager port). But I have problem below. Is there another step that I need to do? ----------------------------------------------- Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.0 Action:Login Username:theuser Secret:somepassword Response:

CentOS-6.6 Is there any command line program that determines and reports what pam_cracklib.so returns for a given password; subject to variation in the command line options and values provided? For example, assuming a cli driver program called cli_driver_pgm: cli_driver_pgm pam_cracklib.so difok=8 minlen=14 dcredit=3 ucredit=3 lcredit=3 ocredit=3 minclass=2 maxrepeat=3 maxsequence=3

Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file? It seems to only read it on startup. Thanks.

Hi listers, We want to implement one call center with asterisk. The idea is it should be scalable, with openser as an dispatcher and bunch of asterisk servers to do ACD, Queues, Agents things... Easy to say :( Look closely to the current asterisk, we do see some problem: - SIP registrations was stored in astdb. - And queue members also was stored in astdb. - ... asterisk was built as

Hi, I have tried several different line options, but I cannot seem to get the pxe master password option to work? I have basically tried all combinations of: DEFAULT local PROMPT 0 ALLOWOPTIONS 1 TIMEOUT 500 MENU TITLE sometitle MENU MASTER PASSWD somepassword But just can't seem to get the menu to prompt for a password. version I have is pxe 3.31 Thanks for any help

I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a

Are there any plans to allow us to specify UPS variables in ups.conf, eg: [evo_s_3000] driver = usbhid-ups port = auto desc = "Eaton Evolution S 3000" vendorid = 0463 productid = ffff ups.delay.shutdown = 120 ups.delay.start = 130 The reason I ask is that I'm using FreeBSD & gmirror and need to extend the gracetime

Hi, On Lenny, I typed "apt-get install db4.5-util " then (as root) : # db4.5_dump /var/lib/asterisk/astdb db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument # file /var/lib/asterisk/astdb /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3, native byte-order) Is db4.5_dump appropriate to dump an