similar to: application meetme

Does anyone Know if tha avaya 4621 IP phone work wiht asteisk? if it work it has featuras working Thanks Ignacio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051014/d3a65784/attachment.htm

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198

Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I

Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'.

http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is working fine in my Setup, but I want Extern1 to talk to Extern2 directly whitout going over Asterisk as the uplink is slow. When I set for Extern1/2 canreinvite=yes it works, but "Intern-2-Extern" doesn't work because Asteisk gives out the private IP-Adresses of Int1/2 I defined localnet=10.0.0.0/255.0.0.0 (Private

Dear list i am using Asterisk 1.2.5 with A@H . here is my problem. if i dial a number (consider 79) i have to wait around 20 seconds before my Asteisk box response. now i want to decrease this waiting time . any idea how to do that ? thanks Salaque

You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Then again, in my system, the in and out channels are mixed first before they are written to the disk. ________________________________ From:

On Thu, Jun 14, 2012 at 11:44 PM, Jacob Carlborg <doob at me.com> wrote: > On 2012-06-15 07:25, Chandler Carruth wrote: >> Figured it was good to ask this as James is busy making sure the Clang >> doxygen is actually all well formed, and reasonably linked together so >> we have proper docs for the ever growing number of Clang library users... >> >> There are

I did (re-)compile dovecot (1.1rc8) today, with managesieve (0.10.2) and sieve-plugin (1.1.5). I did modify the existing dovecot-devel port, adding managesieve patch as an OPTION, and I added a 'dovecot-managesieve-devel' port, and a 'dovecot-sieve-devel' one to be complete... Not sure everything is very complete, but at least it works for me=E2=84=A2= , and if it can help anybody

On Fri, Jun 15, 2012 at 1:21 AM, James Dennett <james.dennett at gmail.com>wrote: > On Thu, Jun 14, 2012 at 11:44 PM, Jacob Carlborg <doob at me.com> wrote: > > On 2012-06-15 07:25, Chandler Carruth wrote: > >> Figured it was good to ask this as James is busy making sure the Clang > >> doxygen is actually all well formed, and reasonably linked together so

Hi all, Our asterisk PBX, randomly restarts all the channels of the E1 connection. It sends this message "There is no D-Channel, using channel 16 anyway".Then the asteisk recive (or it thinks it recives) yellow alarms at all the B-channels, after that it restart all the channels. When restarting the B-channels it cut all the conversations that is handling at that moment. Does anyone

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

Hello Anyons has tested this configuration with success ? I'll test it in few days and I am wondering if I am alone :-) Thanks

On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote: > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote: > > Hi. > > > > (Asterisk 16.2.1) > > > > I'm using AMI Originate to initiate calls, and I'm passing some > > additional data in to the dialplan context using the Variable: > > parameter. Works fine. > > > >

Hello, I am trying to configure ZFS with IBM sdd. IBM sdd is like powerpath, MPXIO or VxDMP. Here is the error message when I try to create my pool: bash-3.00# zpool create tank /dev/dsk/vpath1a warning: device in use checking failed: No such device internal error: unexpected error 22 at line 446 of ../common/libzfs_pool.c bash-3.00# zpool create tank /dev/dsk/vpath1c cannot open

Hey, I've been strugling to get quota working on my enviroment. I have one storage server that exports the 'home' dir. My mail server mounts the exported 'home' server at '/home' # dovecot --version > 1.1.14 I'm also running Postfix and using Dovecot/LDA as MDA so that we can use the Dovecot's Quota plugin. The thing is that if the user has exceeded

For those who wants testing 1.1rc3 on FreeBSD, this is a simple patch against /usr/ports/mail/dovecot. To apply: # cd /usr/ports/mail/dovecot # patch -p1 < /tmp/patch_dovecot_1.0.13-1.1.rc3.diff Hope I did not forget something important... This is an adaptation of the existing port. PS: for port's maintainer: Which way will take 1.0 vs 1.1 upgrade ? a new mail/dovecot10 port or ? Hope

I'm testing dovecot as courier-imap replacement. My setup: FreeBSD 6.1 nfs homedirs on FreeBSD server ~/Maildir 's LDAP auth - bind_auth=yes Seems to work well in most cases, but: - No way to read a maildir with 35000+ messages (a freebsd mailing-list for some years) => Nov 21 12:11:42 host dovecot: child 7217 (imap) killed with signal 11 - No way to have quota plugin reporting

Hi. (Asterisk 16.2.1) I'm using AMI Originate to initiate calls, and I'm passing some additional data in to the dialplan context using the Variable: parameter. Works fine. https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_Originate Now I need to do the same thing but from another context in my dialplan, so I was expecting to use the Originate() dialplan command,

Hi, We are using dovecot for some times (a little before 1.0), for IMAP and (a little for the moment) deliver. We have ~20000 logins/day, Maildirs are over NFS (with indexes too, for the moment...) Up to 1.0.5, we sometime have this error in maillog: ________________________________________________________________________ Nov 1 13:55:41 tic dovecot: IMAP(login): file mail-index-view-sync.c: