similar to: Re: [Serusers] ASTERISK+SER+MWI

Waldo, How do you let your customers manage 'their' PBX. I too have a setup like you. However, I installed a * server for each customer, via vserver. I'd like to now what kind of software/webbased package you use for this. I also have SER installed as a front-end server for the * servers. But, as I'm still not very into SER, don't know exactly how this fits in. Should I use

lqbal, I do plan on having alot of users. Two markets I'm trying to get some volume users from are: residential consumers and business users. Residential consumers should get basic line services such as their own DID, voicemail, caller-id, call-waiting, three-way calling, and basically, all the standard features you get from companies like Vonage, etc. This particular market base

Hi Does anyone know of a retail outlet in the UK where you maybe able to purchase cards for asterisk. Iqbal

Hello all, I have just been trying to install the latest ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ into my cvs version of Asterisk and have found that the install patching fails. --------------------------------------------- patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c patching file

hello, I need help with ast_data I downloaded asterisk from cvs cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot co -r HEAD asterisk and the latest ast_data. When i run ./INSTALL.txt i get : serveur1:/opt/asterisk/ast_data# ./INSTALL patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c Hunk #1 succeeded at 27 with

Hi If I have ser sending calls to asterisk, is there a way to get a different block called in sip.conf for each call (based on some variable, NOT username, From:), if not and they all hit one block which has contect=abc, then when that context is called/matched in extensions.conf, how can I have diff features for various groups of users. EG lets say I have a large company with 4 departments

Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar

On Wed, Mar 28, 2018 at 6:25 PM, Steven Tardy <sjt5atra at gmail.com> wrote: > On Wed, Mar 28, 2018 at 3:16 PM Asif Iqbal <vadud3 at gmail.com> wrote: > > > It is not respoding to A server which is sending the tftp read request > RRQ. > > > > I do see the RRQ packets coming from A to S, but S never responds back > from > > a different port Y to A

On Thu, Mar 29, 2018 at 12:48 PM, Asif Iqbal <vadud3 at gmail.com> wrote: > > > On Thu, Mar 29, 2018 at 7:21 AM, Steven Tardy <sjt5atra at gmail.com> wrote: > >> A STATEFUL firewall with ?ip any any? can and will still block asymmetric >> communications due to the firewall keeping track of state (hence tha name >> stateful firewall). >> >>

Hi, Is there any way to pick up a remote phone/extension wich is ringing for a long time but no one is availabe to answer that call? OR the scenario is like, suppose my extension is 1000, and I am working at a colleagues desk. At this time my phone is ringing in my desk and I want to pick up the extension I am presently at and get the call. I have another question about the remote login. Is

Hi everybody, >From asterisk CLI we can add extensions in dial-plan dynamically using "dialplan add extension" command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal

I'm looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050515/d66e6335/attachment.htm

Anyone done a chat line app? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

Hi how do I tell with user type to use userdel with -r short of using exec type? i have a user account called `foo'' so i created a test puppet file local to the system called test.pp # cat test.pp user {"foo": ensure => absent, managehome => true, } when I apply it on solaris or linux the userdel does not use the `-r'' switch

Hi All, Is there a ubuntu repository managed by puppetlabs? I am using lucid and latest puppet package it offers is 0.25.4. I could of course just download source, or a deb from debian or newer ubuntu, or get it using gem. But would be nice if there is a PPA available for lucid, cuz it would work nicely with routine upgrade. puppetdashboard worked nicely with the puppet labs ppa deb

How do I find out why df -k is taking soo long to respond only sometimes? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing?

I have a tftpd server S running on centos 7 and managed by systemd It is not respoding to A server which is sending the tftp read request RRQ. I do see the RRQ packets coming from A to S, but S never responds back from a different port Y to A So this part is working fine https://en.wikipedia.org/wiki/Trivial_File_Transfer_Protocol#/media/File:Tftp-rrq.svg But I do not see any attempts to

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

I'm trying to patch the current asterisk CVS with spandsp-0.0.1k.tar.gz. Everything compiles fine but when I go to patch the asterisk/apps/Makefile it fails: asterisk:/usr/src/spandsp2# patch < Makefile.patch can't find file to patch at input line 3 Perhaps you should have used the -p or --strip option? The text leading up to this was: -------------------------- |--- Makefile.orig