similar to: waitfordialtone patch

I have an interesting observation which I thought I'd pass along to save other people from spending time trying to 'fix' it. One of my clients uses Charter's so called "business phone service". They provide 'analog' phone lines over IP. In general, they've worked OK. End users were saying that the phone are "cutting out" at times. What

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Why pay 1.1 cent's a minute for interconnecting to another Asterisk server for a high volume call center. Do people really understand what they are trying to sale and take an honest look into what they advertise. As a high volume user like a call center I would not connect my Asterisk Box to there Asterisk Box to a third Sip provider who then hands of to the Level 3 and so forth. With LD

Is it possible with asterisk to setup aliases for SIP? For example, direct sales@mysipdomain.com to 55544@mysipdomain.com If this isn't possible directly with asterisk, does SER offer anything along those lines? A search of the usual sites didn't turn up anything conclusive. Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com

The AstLinux Team is happy to announce the release of AstLinux 0.7.1. This is a bugfix release which includes updates to Asterisk (1.4.30), Dahdi and several other items as detailed in the Changelog. http://www.astlinux.org/release/071 Existing users can upgrade from the web interface or from the CLI. From the CLI execute the following: upgrade-run-image check

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

I'm considering switching my incoming phones lines from standard analog to a T-1 service from XO communications. They propose to bring in an "IAD" which has 12 lines of voice and 768k of internet bandwidth as part of a package deal. Since I want to keep the voice traffic in the digital domain the equipment they're proposing is a "Lucent Digital Vina Integrator" IAD

Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I?m trying to find an IAD which works with Asterisk. I?ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it?s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it?s nice to control using MGCP. Any information will be apreciated ! Thanks. --

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

The AstLinux Team would like to announce that the 0.7.0 version of AstLinux is available for download. There have been many significant updates in this release including updating to the latest Asterisk Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other system updates. For a complete list of changes, read the changelog available on the download page:

The AstLinux Team has released 1.2.0. All current users are encouraged to upgrade as this release addresses the bash "ShellShock" bug. New in 1.2.0: * New Linux Kernel 3.2.x * "igb" ethernet driver for Intel Atom C2000 * Enable AES-NI support * New "sip-user-agent" firewall plugin * New versions of Asterisk 11 and 1.8 * Bash "ShellShock" security fixes A

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Hi Stephen, Thanks again for getting back to me, Ivan Krylov responded also and suggested windows binaries and I must confess I was only familiar with installing from files via the package sources (apart from the conventional install.packages method), so the solution was as simple as installing via the binaries. Thanks again, best wishes, James From: stephen sefick <ssefick at gmail.com>

All, I am posting this here to announce I have finally managed to get my Cisco IAD2421 to speak MGCP with Asterisk. Due to an acute lack of reading on the subject as searched on Google, I'm putting this out with the hope that it helps whomever should need to do this in the future. This should also apply to the IAD2420 and the other models in the line, but as I do not have access to those,

Hello Alice, On Wed, 2016-10-19 at 14:22 -0700, Alice Wonder wrote: > I formerly used secp521r1 but suddenly Google with no warning stopped > supporting it in chrome. That company is too powerful. Actually this is something the NSA insists on:

Hello, after upgrading from Samba 3.5.6 to 3.6.5 I encounter some strange NT_STATUS_ACCESS_DENIED trouble when trying to access group readable directories from Windows 7 or Windows 2008 Terminal Server. Very strange stuff is that it works fine using smbclient from a Linux machine using the same userid and Kerberos authentication. The Server is fully integrated into active directory and uses

P.H.B. is insisting on having the ability to create a transparant SIP tunnel between old style ISDN telephony PBX with overlap dialing: PBX - ISDN - IAD - SIP - * - DAHDI - PRI The idea is that dialed numbers a the PBX are transmitted to the PRI as they are typed, whenever the PRI gets the signal that the number is complete the dialer instantly gets a ringing. This behavior is described in RFC