similar to: IAX2 Trunking - CVS-Head

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

Hi I get an error popping up when the call ends as follows: DBD::mysql::db do failed: Unknown column 'callstart' in 'field list' at /var/lib/asterisk/agi-bin/astcc.agi line 90, <STDIN> line 32. Does anyone else get this same thing? Looks as if my database table is wrong, or something else is up...not sure Thanks Clive

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new

I think this is the fourth attempt to send this blessed message, so let's hope this gets through without any 'unprocessed' or 'ignored' in-lines on auto-reply. I wish to report to you some strange problems I'm experiencing with installing packages directly into my -R- 2.15.1 (there is an indirect solution, which I note below). First, here's some essential information:

Two * servers: *a and *b. Outside call comes in *b, and is automatically routed to *a. Someone on a sip phone connected to *a then decides to transfer the call to someone on a sip phone connected to *b. The transfer works. At this point, is *a still in the converstation? Or is * smart enough to see where the data stream is going/coming from? Thanks for any help in advanced, and sorry if

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive Email :

Hi Having my first go at compiling Asterisk from cvs source. Compiled and installed zaptel ok Running make asterisk returns the following error message /usr/bin/ld cannot find -lssl collect2: ld returned 1 exit status The last part of the compile messages on screen are- editline/libedit.a db1.ast/libdb1.a stdtime/libtime.a -ldl -lncurses -lm -lresolv -lssl There is obviously something I have

The locfit library is now available through CRAN, in the Contributed R Code directory. Locfit fits local regression, likelihood and density estimation models, in the spirit of loess but with many additional features. To install, unpack the locfit_19980309.tar.gz file, and R INSTALL locfit Most of the functionality and examples on my home page http://cm.bell-labs.com/stat/project/locfit/ should

The locfit library is now available through CRAN, in the Contributed R Code directory. Locfit fits local regression, likelihood and density estimation models, in the spirit of loess but with many additional features. To install, unpack the locfit_19980309.tar.gz file, and R INSTALL locfit Most of the functionality and examples on my home page http://cm.bell-labs.com/stat/project/locfit/ should

Hi, My question is in refernece to the posting by Jeremy McNamara here.. http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html He states that in order for "trunking" to work the type has to be peer.. When I set mine up I did so using type=friend just to make it simple.. So when I read the above posting I thought well maybe my "trunking" has not been

Anybody know of a UK supplier of "Voip Telephony with Asterisk" " by Paul Mahler ? I've searched on the web, and the only suppliers I can find are US based, and the postal charge is as much as the book. cheers -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856261

Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ........... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to

Hi, I am trying to install the Asterisk-addons-1.4.2, and when I make install it prompt me such error messages make[1]: Entering directory `/usr/src/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory

Hi all, Just want to check from the list experienced personal about the Digium HPEC, where I had purchased the HPEC and wish to run with TDM card Sangoma A200. I can't install HPEC to run with Sangama A200 card, even I had changed my hpec file from i686 to i386. The error that I had as bellow; > Found key 'HPEC-XXXXXXXXXX' for 2 channels. > Found valid HPEC licenses for 2

IAX2 now has support for a "trunk" mode ("trunk=yes" in the appropriate friend section). Trunk mode allows IAX2 to use bandwidth extremely effectively. The original impetice (and strategy) was a result of a mistake in which it was claimed that Asterisk with a T100P could send 96 simultaneous calls over a single T1 using VoIP. Thanks to a suggestion by the customer, combined