similar to: IAX and calls on hold

Hello - I recently offloaded some of the SIP traffic on to a separate Asterisk box and interconnected our main Asterisk system with the new system via IAX. The SIP clients are running 7960's. When a call is put on hold, often times when the call is pulled off hold, there seems to be no RTP in at least one direction. There seems to only be voice in one direction. There is usually at least a

Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently

Hi all, What does (T) mean on the output of "iax2 show peers"? The following my output. darkstar*CLI> iax2 show peers Name/Username Host Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata 201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A & B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know

Here's our config: cisco 7960's running 6.3 sip code latest cvs of * t100p zaptel card adit 600 channel bank 7 pots lines and 2 fax machines on the adit 600 dialing out from the cisco phones gets sent out via the zap channels, but I'm having some serious echo problems. I currently have the adit set to +3 rxgain and -6 txgain, with my zapata.conf containing: echocancel=128

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Hi, I've had a hard time understanding what was going on in a new * setup. The deployment has a * box running on dual xeon RH9 stock 2.4.20-8 and some different versions of asterisk (1.2.10/1.2.16) + libpri + zaptel + wanpipe. Short version: audio from iaxclient clients is fine from windows but poor from linux when going to zaptel. E.g. Iaxcomm running on windows works fine, but the same

Hi, I just moved from 1.08 to 1.10 and now have one example failing, which, under 1.08, passed. Is the due to a change in behaviour? Here''s my spec (removed some passing examples) require File.dirname(__FILE__) + ''/../spec_helper'' describe "A user" do before(:each) do @user = User.new @valid_user = User.new( :email =>

To Whom It May Concern: I have an ActiveRecord-based application (non-Rails). Life was grand until I upgraded ActiveRecord yesterday, after which point I was getting TONNES of these errors from my app and in PostgreSQL''s logs: FATAL: terminating connection due to administrator command According to Google searching, this happens when an external process sends PostgreSQL SIGINT or

Dear Samba-Team. First, I don't know if this is the right place for my request. If a bug report is more suitable, I can file one. My request is interconnected with CUPS, but I was told to ask you instead as - CUPS avoids to document software-specific stuff, because there are too many changes in the programs interconnected with CUPS to keep up with and the SMB backend is - as a part of Samba

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

Hi, I''m looking to create a ruby script that loads a bunch of records and manipulates these. Where do I start? I got this to begin with: #!/usr/bin/env ruby require File.dirname(__FILE__) + ''/../config/boot'' players = Player.find :all But it complains about a DB connection. How do I set this up? Jeroen

Hi, I am not sure if this is the right mailinglist (since I have an ogg problem not a vorbis)..... I am looking for some pointers on how to packetize an ogg stream. My idea is to use udp packets for transport, so I would need a way that would minimalize the impact of lost packets. Are there any code examples out there for this? Thanks, Jeroen p.s. Please cc me, I am not subscribed to this

Hi, I am not sure if this is the right mailinglist (since I have an ogg problem not a vorbis)..... I am looking for some pointers on how to packetize an ogg stream. My idea is to use udp packets for transport, so I would need a way that would minimalize the impact of lost packets. Are there any code examples out there for this? Thanks, Jeroen p.s. Please cc me, I am not subscribed to this

Hi Everybody, I''m playing around with the engines created by James Adam and I bumped into the following. I would like the rake bootstrap command to use the testing database and I just know there''s an easy way to do so. I could have just copied the dev db''s to the test db''s but I think I''m missing something fundamental here. Hints are very welcome.

Hi, Is there an easy way of querying an active record for a maximum column value? I need to do queries like: SELECT MAX(salary) FROM employees TIA, Jeroen

> From: Eric Christopher > Subject: Re: [LLVMdev] How to associate extra comments to a > MachineInstruction ? > > Should be spelled like this yes? > > Asm->OutStreamer.AddComment("foo") > Asm->EmitFoo(); > > -eric That should work at the moment that you are emitting the instructions. But what would you do when you are manipulating a

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to