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Seshu, Whats with people who work at Morgan Stanley? You on the one hand bash the opensource software, and then on the other hand a few weeks later, ask for assistance from the open source community for assistance??????????????????????????????????????????? Today, May 11, you request assistance in installing the Areski Calling Card platform, after posting a couple of weeks ago, April 20, this

Why don't u attach the setup page of the phone ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di FaberK Inviato: gioved? 13 ottobre 2005 17.56 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] PA168S/AT320P Right now, but nothing changed. 2005/10/13,

For the benefit of everyone, having H323 Configuration problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems

List, The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk. Seshu Kanuri "G

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

Folks! Let me clarify something to the Asterisk community about the CDR tool. 1) This is *not* my code to start with. I picked the original code from this forum here... http://www.voip-info.org/tiki-index.php?page=Asterisk%20CDR%20Areski%20GUI 2) The original code was not working (for most part, as the MySql portion has bugs) and I fixed this and added a few bells and whistles. 3) The

I am using PBXware for configuring users and extensions. Pbxware uses Internal script called init.sh to process the calls based on its own version of extensions.conf defined in the GUI. I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. I have used IAX2 extension 101 and dialed SIP Extension 51 But the PBXWare's Init.sh AGI command identifies the DNIS as another IAX

----- Original Message ----- From: "chris" <chris@fivestartel.com> To: "Frank Tarczynski" <ftarz@mindspring.com>; <seruser@iptel.org> Sent: Tuesday, July 26, 2005 9:54 AM Subject: Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found > hi Tzafrir > > any idea how we can fix this error? > > include/asterisk/strings.h:232: parse

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

Hi, I didn't get any solution in the mailing list. [http://asterisk.linkx.net/asteriskusers/200409/msg01167] What should be the next step? Changing the machine??????? Is it machine dependent?... Regards, Somesh S. Shanbhag --- Tzafrir Cohen <tzafrir@cohens.org.il> wrote: > On Tue, Sep 27, 2005 at 12:13:21AM -0700, somesh s > wrote: > > Hi, > > > > I did

I have setup Grandstream to connect to my Asterisk Server. All the digits 0-9 are accepting dtmf. But When I try to send the call by Pressing # Key, nothing happens. Does anyone has a standard configuration for Asterisk and Grandstream as a PDF file or something to see. How do you send the connect signal? Seshu Kanuri -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hi My toy mail server runs dovecot. My client is mutt, which I use normally use a a remote screen session. I normally connect to imap locally: account-hook imap://localhost/ 'set tunnel="/usr/local/bin/imap-local"' After a long frustration from the new mail notification in 0.99, I have decided to bite the bullet and try the -test series. Ever since I have not had stable mutt

I originally wanted to answer with something ... tzarit and kevit Readed probably before you invent rapid biz. I am asking to share any info /experience not your high spirit. Thanks for less trivial answer, G >On Wed, Oct 05, 2005 at 12:44:27PM -0700, Thameem Ansari wrote: >> I am using the inter asterisk trunking and the article in >> voip-info.org<http://voip-info.org>will

Hi folks You are welcome to try our (Xorcom)'s latest debs (for Xorcom Rapid, or Debian Sarge in general) "Unstable": Asterisk and AMPortal: The repository is available at: deb http://rapid.dotsrc.org/ unstable/ #deb-src http://rapid.dotsrc.org/ unstable/ The commands you are looking for are: apt-get update apt-get dist-upgrade apt-get install amportal

Hi I must be missing something here: do the zaptel tarball (1.2.2 and 1.2.3) miss the file .version? Without it version.h will be generated with an empty version number and some bad things will happen, IIRC. I've added it mnually to the Debian package. But I have a feeling that I have somehow missed the way it should be generated. -- Tzafrir Cohen | tzafrir@jbr.cohens.org.il | VIM

If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

There was some discussion in the past about which one is the best Content Management System that can be used in conjunction with Asterisk. Mambo was supposed to be the best out there under GPL. The guys who developed Mambo have a new product now - Joomla. I am using this and it appears to be better than Mambo in many respects. Read the gist about Joomla below. ------------- If you've read

I agree, use either SCSI with hardware raid with a battery backed cache or use sata/ide with linux software raid. Linux raid SCSI also works well, but if you go for the scsi drives might as well get the controller too. The "firmware" raid on the cheap sata/ide cards have left me stranded several times, I have had experiences were both an HP and Promise IDE raid controller have SCRAMBLED