similar to: Delay after Answer

Well, I thought I had the problem solved. Ported everything over to PJSip and build RDNS records for the phones and the server, but I am still experiencing the problem on incoming calls. ** On 6/7/2016 1:00 PM, Faheem Muhammad wrote: > I've faced the same issue. The issue was related to DNS, the reverse > lookup query failure caused the delay around(7-9 seconds). The purpose >

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

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

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

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'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 have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice,

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

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

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

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

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

Is there a way to detect if a call is a transfer in the dialplan? Here is my issue: I have an office with 2 extensions. Under normal circumstances any call that comes in should ring both extensions. I accomplish this through a queue. The problem is that if the call is answered on say extension 11 and the answerer wants to transfer the call to the other phone, extension 10, transferring

I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97

Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't

The Asterisk.org development team has announced the release of Zaptel versions 1.2.25 and 1.4.10. These releases contain many bug fixes as well as performance enhancements. A couple of the more major changes include: modifications to the wctdm24xxp and wcte12xp drivers to increase interrupt latency resilience, numerous bug fixes and updates to the xpp drivers, as well as some Makefile

The Asterisk.org development team has announced the release of Zaptel versions 1.2.25 and 1.4.10. These releases contain many bug fixes as well as performance enhancements. A couple of the more major changes include: modifications to the wctdm24xxp and wcte12xp drivers to increase interrupt latency resilience, numerous bug fixes and updates to the xpp drivers, as well as some Makefile

I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference

I've got a situation where I need to use a transfer to the parking lot as hold, but am not going to use BLF indicators on the phone to pick up the parked calls so I need to hear the 3-digit extension after the transfer. I'm using Snom 300 phones and have tried setting a programmable button to Key Event F_TRANSFER 700, which successfully does the transfer but cuts off audio so you