similar to: sounds /vm-*.gsm / apps_voicemail.c

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

How is a callgroup used in the dial plan?? I can't see an example in the extensions.conf.. > See configs/sip.conf.samples search for pickupgroup > > RTFSC (Sample Configs) > > > Jeremy McNamara > > > > > WipeOut . wrote: > > >Is there a way to pickup a call whn using a SIP phone? > > > > > > > >> From

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

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?

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

TC: You were right on the money. The new firmware L-34 fixed my problem with the Channel Bank. Now the lines are properly closed and not hanging up the lines. One of these days I should read more about the 750. I saw some parameters for "network on-hook/off-hook + test" that I don't even know what they do. Thank you so much. Regards, Uriel -------------- next part --------------

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 .

I was using the Asterisk::Manager perl module and had some troube with it so I decided to make my own. I have a pretty good prototype after a few hours (My main point about the protocol is at the bottom of this example.) http://asterisk.650dialup.com is where you can download it. my $man = init Asterisk::AstMan ({ -user => "asterisk",

Hi: I''ve met an ext4 Bug in dom0 kernel 2.6.32.36. (See kernel stack below) 32.36 kernel commit: http://git.kernel.org/?p=linux/kernel/git/jeremy/xen.git;a=commit;h=ae333e97552c81ab10395ad1ffc6d6daaadb144a The bug only show up in our cluster environments which includes 300 physical machines, one server will run into this bug per day. Running ontop of every server, there are about 30

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

After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/fc494bb4/attachment.htm

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.

All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

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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, >

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

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.