similar to: Asterisk Regular expression to validate any phonenumber

I''m trying to modify the SaltedHashLoginGenerator to where it separates the users table into two tables: a users table which contains login info: token, salt, etc) and userprofiles table (which contains firstname, lastname, authlevel, etc) I''m trying to use the existent helper for user under app/helpers/user_helper.rb and copied to app/helpers/userprofiles_helper.rb and

Hello, Can anyone suggest us how to got the phonenumber of the caller. In the environment variable, I just see the ip of the gateway. Environment: 'agi_callerid' is 'XXX.xxx.XXX.xx' Should I do some changes in the GW conf, or is it just not possible when we got a call from PSTN ?!? Thx in advance, Ares

Hello! I finally found a working solution. calling divactrl with the parameter -n [0..20] gives the DID-length means, if you wanna have 123-XXX in digit-wise mode, then call divactrl load -c 1 -n 3 -f ETSI and the card will wait for n digits. regards, Sebastian -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]

If I have a module with a callback in it: module Sanitize module PhoneNumber def before_validation self.phone.gsub!(/\D/, '''') if attribute_present?(''phone'') end end module MobileNumber def before_validation self.mobile.gsub!(/\D/, '''') if attribute_present?(''mobile'') end end end

Hello, How can Set(language()) in asterisk-1.6.x ? Cordialement, P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090901/f0f3e73d/attachment.htm

Hello, I have a var like ?blabla? with the ? I need to suppr the ? Is it possible with the ${var:x:y} ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have conferences in my database. I need at some moments, to access the database without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

I'm attempting to create an array of treatment comparisons for modelling data generation. This involves comparison of one treatment (c2) with another (c3), relative to a common comparator (c1). Attached code gives me the correct array but need to remove duplicates. Duplicates relate only to c2 and c3 such that I need to remove r3 because c2 and c3 are same as r1 with c2 and c3 swapped r5

Hi I know this is far from best practice but is it possible to authenticate a sip peer on the IP address it's coming from so that it doesn't need to use a UN and Pass? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hello, I am looking for a good prepaid solution. What is the best ? Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081212/5188f96b/attachment.htm

Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 & 2002 are behind one firewall, and 2003 & 2004 are behind another. Tim

Hello, I work on voicemail.conf and I need that ${VM_DATE} is in french! How can I do it? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090319/7c1de958/attachment.htm

Hello, I am trying to do : Exten =>_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:10000)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:10000)' Why? I forgot something ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. *same => n,Read(mobileNumber,app/input-mobile,10,,2,15)* In the logs: When it fails: - - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr') - - User disconnected When it succeeds: - - <SIP/ipbx-iwred-000002e> Playing

hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part --------------

Hello to you all! We are implementing here at the University KVM virtualization for our servers and services and i was wondering if anyone virtualized domain cotrollers to KVM. Does anyone done this before? Any advice? -- Andry Michaelidou Papa | IT Systems Administrator|Department of Computer Science| University of Cyprus Tel: +357.22.892734 | Fax: +357.22.8927231 | http://www.cs.ucy.ac.cy

Hello, I need to execute an agi in php. I have that: == Using SIP RTP CoS mark 5 -- Executing [0170725000 at mnupprx1:1] Answer("SIP/33179977999-b6c18478", "") in new stack -- Executing [0170725000 at mnupprx1:2] GotoIf("SIP/33179977999-b6c18478", "0?6:3)") in new stack -- Goto (mnupprx1,0170725000,3) -- Executing

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming