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...the same issue. The issue was related to DNS, the reverse > lookup query failure caused the delay around(7-9 seconds). The purpose > of reverse lookup is to block IP Spoofing attacks. > > Regards, > Faheem > > On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson > <brent at texascountrytitle.com <mailto:brent at texascountrytitle.com>> wrote: > > I am having an issue with a couple of phones where they ring, but > there is a long delay after the phone is picked up before the > audio starts. > > My setup: > > * Server running Asteris...

I am having an issue with a couple of phones where they ring, but there is a long delay after the phone is picked up before the audio starts. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT. Server and phone are on the same subnet with only a gigabit switch between them. * Digium

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO NAT*_ involved. Phones and server are plugged into the same network switch, all on the same IP range. The server is running a Wildcard AEX410 analog card with 2 FXO modules receiving incoming analog lines. Occasionally, in the middle of a call, the

I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that

We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the

...k 1.6.1 + openais (Russell Bryant) > 3. asterisk+heartbeat (Nhadie) > 4. Re: ISDN (Wilton Helm) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 14 Oct 2008 11:11:23 -0500 > From: Brent Davidson <brent at texascountrytitle.com> > Subject: Re: [asterisk-users] is there a way > To: Asterisk Users Mailing List - Non-Commercial > Discussion > <asterisk-users at lists.digium.com> > Message-ID: <48F4C4AB.8070403 at texascountrytitle.com> > Content-Type: text/plain; charset="iso-8859-1...

I have several branch offices, each with their own Asterisk server (version 1.4.22.1) handling their PBX functions. All of these offices need to talk to each other. In sip.conf I created a peer entry for each office with a username of branch-user and a friend entry for every branch-user with the username being just the branch, for example: [Office2] username=Office1-user host=10.10.80.253

Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8

If I have a global variable in my dialplan and I change it, does that change immediately take affect for all calls that are active? Here is my situation. The company I work for has two office groups that share a building. The two offices are separate companies but support one another and want to be able to transfer calls as if they were all on the same phone system. Each company has 4

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and

Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them "try out" our dialtone

...> An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080925/e3ab5815/attachment-0001.htm > > ------------------------------ > > Message: 23 > Date: Thu, 25 Sep 2008 17:16:35 -0500 > From: Brent Davidson <brent at texascountrytitle.com> > Subject: Re: [asterisk-users] Create virtual extension > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <48DC0DC3.5050708 at texascountrytitle.com> > Content-Type: text/plain; charset=ISO-...

I have a custom application that is primarily an indexed database of scanned documents. The system can print either search records or the individual documents which are stored in Tiff format. Under windows, everything works perfectly, but there are some issues with speed at our remote offices. I am using FreeNX and Wine to set up remote access to the program to mitigate the overhead that

Is there a detailed explanation of the printer setup in the registry of Modern Wine (0.9.38) ? I can't seem to find any documentation and I'm having some weird printing problems that I would like to resolve and I think the key may be in the registry. Thanks, Brent Davidosn

I've recently installed Asterisk-based servers at several of our branch offices. Each server has 2 X100P cards to handle 2 incoming voice lines. I was having a lot of trouble with Echo until I got OSLEC running on all of the servers, but now we have a new problem. Incoming callers are not always able to dial extensions. I would say probably 95% of the calls go through correctly, but

I've been reading most of the day and can't seem to find a clear definition of the syntax for parking lot hints in AEL2. I have tried all of the following and they either do not light up the line button on my Snom 300 or give syntax errors: hint(park/701) 701 => { ParkedCall(701); } hint(park:701) 701 => { ParkedCall(701); } hint(park/701 at parkedcalls) 701 =>

Does anyone have any idea what would cause distorted audio but ONLY for DTMF tones coming in over our analog lines. (The analog interfaces are X100P's). I have carefully adjusted the gains in the zapata.conf using a local test line after trying various settings with no gain or just random gain settings. RelaxDTMF has no effect. I set up a monitor command in my dial plan to capture

I'm trying to test out Speex for our branch to branch connections, but am running in to a problem. I downloaded the Speex source code for 1.2rc1, did a ./configure, make and make install then went to my asterisk folder did a ./configure, make clean make menuconfig verified that speex is enabled, saved config then did make, stopped asterisk then make install and start asterisk. Did the

On my asterisk system, if an incoming call only has a number for the caller ID and no name, the system is using the channel name as in the Callerid Name field. I would like to use some sort of pattern match test to test for the presence of "Zap/" in the ${CALLERID(name)} variable and if it is present, replace it with "Unknown". I'm using the ael format for my