similar to: Using callgroups (was: Taking a call for someone elses extension from my extension)

Callgroups/pickupgroups are allocated per channel, not in the dialplan. sip.conf and zapata.conf are the two files you're interested in. -wade ---- Original Message ---- From: wipeout@linuxmail.org To: asterisk-users@lists.digium.com, Subject: RE: [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extensionfrom my extension) Date: Sun, 20 Apr 2003 16:39:15 +0000

On Wed, 30 Apr 2003 00:24:19 -0400, Uriel Carrasquilla wrote: > >Gary: >I just copied the content from chan->exten to chan->dnis. I am calling from How are you doing this coying ? >one extension to another. >Have you got DynExtenDB to work? nope, haven't got over the first problem yet. Gary .

Every time I send an e-mail to the * list, I receive this "AntiSpam UOL" E-mail. is anybody else experiencing the same? How can I get rid of it? Uriel -----Original Message----- From: AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br] Sent: Wednesday, September 24, 2003 11:51 PM To: uriel@adelphia.net Subject: RE:RE: [Asterisk-Users] SIP / GrandStream Configuration Ol?,

Where can I learn the syntax for DBput and DBget? is it working with MySQL? do I need to set up tables? URiel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030508/a2598dc8/attachment.htm

Hi all, Can any one please help me in intergrating PHP/Mysql with my running asterisk server to configure IAX or SIP users? I will highly appreciate any help in this regard. Regards Nawaz. --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, >

I have one of my asterisks off an xDSL modem with a dynamic IP address. The IP address keeps on changing so I am curious as to what other people are doing to work around this issue. I am contemplating to register for a dynamic DNS service but then my communications become dependent on the availability of this service. The "host=dynamic" for SIP seems interesting. Any suggestions?

I need to set up one user so he can use the Windows Print Manager to delete any print job, he is the print manager. However I do not want him to have any other kind of supervisory privelages on the linux server for now. How do I do this? Thanks in advance, Nicholas Waltham

vmail.cgi rocks (if I can borrow the expression for Mark Street). As Mark pointed out, the /vm/INBOX messages are created with 0700 security and vmail.cgi is not happy. Apache/cgi/vmail.cgi cannot play them unless I fool around with the Apache wrapper or chmod 755 *.* thefiles myself. (Tedious, that is why I like computers). Obviously this is not acceptable. I took a trip to the

I have been fooling around with DynExtenDB and run into two glitches. 1) The code is looking for (chan->dnis) and in my case I find (null). I forced (chan-dnis) to be the same as (chan->exten). So far so good. Now I can connect and talk. This lead me to the second glitch. 2) As soon as the call ends by hanging up, the code issues a (ast_spawn_extension). This causes asterisk to drop

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

I have seen a couple of messages on the Xten and the work done by William Walsh (Kudos). It is not clear in my mind the advantages of SIP phones versus using GnoPhones (once we complete the work for the Windows version). Since I lack the experience with IP SIP phones, can someone, high level, tell me when it makes sense to use them. Is it complicated to set up on the Asterisk side? Thank you.

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Hi, I started to get some Zaptel compile errors after a 'make update' I did a clean zaptel install with: svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel I am still getting the error, is this someelse seeing this ?. CC [M] /usr/src/zaptel/kernel/zaptel-base.o /usr/src/zaptel/kernel/zaptel-base.c: In function 'zt_reallocbufs':

Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel --------------------------------- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. -------------- next part -------------- An HTML attachment was

Author: wyllys Repository: /hg/zfs-crypto/gate Revision: 1b97a96daa581c4f53b4fd8acceb76a27a9fe324 Log message: 6310540 6290437 causes gss_accept_sec_context not to output ret_flags whn no deleg cred; breaks ssh Files: update: usr/src/lib/libgss/g_accept_sec_context.c

callgroups & pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John

Hi all, at time i am trying to get a better idea of callgroups and pickupgroups (especially within the SIP Channel) A Pickupgroup is relative clear - everyone in the same pickupgroup may pickup a call And a callgroup does what ? - The same ? I thought that a callgroup would act like the ZAP groups - so that you then can dial SIP/g1 - and every SIP Client which is in the callgroup 1 does then

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

So I am beginning the long process of learning puppet... and it seems that I am missing some vital things. My systems are RHEL-4, and i have installed the puppet/facter/puppetmasterd from dlutter@redhat.com''s archive. [root@canopus puppet]# rpm -qa | egrep ''puppet|facter'' | sort facter-1.3.6-1.el4 puppet-0.22.2-1.el4 puppet-server-0.22.2-1.el4 on the client I

I have just installed R on a Windows NT system. Unfortunately I am unable to open any of the data files I wish to work with. I have tried using read.table and scan and to the best of my knowledge am using the correct syntax. The error message I receive is Error in file(file, "r") : cannot open file [file name] I have the data in text files in white-space delimited form. I put them