similar to: Problem with chan_iax.cimplimentationcausesbadaudio?

All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4

We upgraded all five servers to 1.2.4. We tried trunking/notrunking. End users use an IAX2 softphone on their desktop PCs. Agents are VLANed and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from the agents to the local Asterisk server as IAX2/ulaw. Then they went over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well). Calls get to the PSTN from the

We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Which Pigeonhole Sieve extension should I use to process/test a message depending on whether the message is in a particular mailbox? (Is it the mboxmetadata extension by any chance? If so, I would appreciate if someone could tell me what the reason is for it not being enabled by default.) The reason I want to perform this test is that I have

Steffen Kaiser <skdovecot at smail.inf.fh-brs.de> writes: > On Tue, 10 May 2016, Jon Kristensen wrote: > >> Which Pigeonhole Sieve extension should I use to process/test a >> message >> depending on whether the message is in a particular mailbox? >> (Is it the >> mboxmetadata extension by any chance? If so, I would appreciate >> if >>

Hello all, I tried to compile Asterisk from CVS yesterday. I would like to try gnophone as IAX client on the same Asterisk server with a sound card. I didn't have any problem of compilation. Here are the config files in /etc/asterisk: /etc/asterisk/asterisk.conf /etc/asterisk/extensions.conf /etc/asterisk/iax.conf /etc/asterisk/indications.conf /etc/asterisk/logger.conf

I received an e-mail from a vendor who says: "We have recently become aware of an issue in the chan_iax2 implementation of IAX2. This issue leads to degraded audio quality. Due to this we are urging everyone to move to SIP." I don't want to discount what this person is talling me, but I'm curious to know why I would only be having issues connecting to his servers, and also what

Just looking for unsolicited thoughts on the Aheeva product? Anyone have anything to say? Thanks, Steve Totaro http://www.asteriskhelpdesk.com

Hi, For a customer, I am looking for a good and reliable Asterisk based system. Five servers will be installed at different locations and will be linked together with each other. This system will work as a call center as well. It has to be a stable and reliable. Customer also needs GUIs for system administration and agents call activities. He also wants video conferencing Please help me select

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

hello all, I have installed centos 5.5 ( linux text) and I have updated it with # yum install bison bison-devel================?ok # yum install ncurses ncurses-devel==========?ok # yum install zlib zlib-devel===============?ok # yum install openssl openssl-deve=======?ok # yum install gnutls-devel============ ==?ok # yum install gcc gcc-c++============?ok # yum install newt

Hi all, We are running Asterisk 1.0.0 with a TE410P. Very often we exerience calls dropping in the middle of the call. I enable the full logging and saw a couple of suspicious messages right before the hangup. Thos could happen on a Zap-IAX2 bridge as well as on a Zap-Agent bridge... I see Nov 23 09:08:36 DEBUG[-1274020944]: Bridge stops because we're zombie or need a soft hangup:

I have a system that is using Intel HDA as its output, and a Plantronics USB headset as the microphone input. I would like to have my wine configuration emulate this setup, I currently have been unable to confirm that it isnt... I just cant get Steam to choose the plantronics headset as the default voice device (this is the goal). Would appreciate any insight!

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

Anybody using Plantronics USB headsets? What softphone are you using and whats your overall experience? Any comments/suggestions?

RJ11 Plantronics 1 3 2 4 3 1 4 2 RJ11 Pin 1 is on the left when looking at the contact points. Plantronics Pin 1 is on the left when looking at the contacts (through the plastic sheild) My multimeter battery is low, so YMMV, but: Pin's 1,4 are connected with ~160 ohms Pin's 2,3 are connected with ~1400 ohms On my Plantronics head set. Chad -----Original Message----- From:

Not strictly asterisk related, however... Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?) Anyway, in a fit of idleness, I thought I'd see what codecs it supports, as I couldn't find it in the manual... And it supports: ilbc g729 ulaw/alaw No GSM! How odd is that, given

I am looking for SIP with power over ethernet desktop phones that will work with asterisk and Plantronics HL-10 Handset Lifter for Remote Answering. Any suggestions? I am considering buying about 150 of these desktop phones for a new call center.

Two asterisk, one in a 2.4.x and another one in a 2.6.x are connection ok using IAX, before i upgraded one of them (the one with 2.6.x) Trunk was working ok. Since i upgraded the one that its now 2.6.x, i m getting this message: chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) ..and no sound comming from the

Hi, I have just discovered (a year after it was implemented) a possibly undocumented incompatability between IAX in Asterisk 1.4 and any version of Asterisk pre-March 2008. It seems an AST_CONTROL_SRCUPDATE frame type was added (in March '08), but no mechanism to negotiate whether it can be sent to the remote end, so if a "new" IAX endpoint sends it, and the remote end ignores it,