similar to: Trouble compiling asterisk 1.2.14

Hi Guys, I hope this is the correct mailing list for this question. I have a dual 1.6 Ghz Itanium with 4 Gb of memory. Yes, a lot of power for Asterisk. I am running SuSE Enterprise Server with the 2.6.5-7.97-default kernel. I have just started to look into Asterisk and I am in the building stage. Now building something on Itanium is almost always a bit of a challange. This is unfortunately

ralniv wrote: > Below are my instructions for getting a libgsm friendly version of Wine compiled on SnowLeopard. I assume that you already have wine-devel installed and configured to your liking. I also assume that you use MacPorts for package management. > > [1] Uninstall wine-devel > > Code: > sudo port uninstall wine-devel > > > > [2] Edit the portfile for

Below are my instructions for getting a libgsm friendly version of Wine compiled on SnowLeopard. I assume that you already have wine-devel installed and configured to your liking. I also assume that you use MacPorts for package management. [1] Uninstall wine-devel Code: sudo port uninstall wine-devel [2] Edit the portfile for wine-devel. It is located at:

Hi all, I'm trying to compile asterisk on the mainframe (s390 / s390x) and I am running into issues. I was wondering if somebody could give a hand? I'm thinking that I should be able to do this. I have noticed that Debian even has binary RPM's out for Asterisk now. I'm trying to do this on SuSE SLES8 (with the 2.4 kernel). What I see is, an issue that arch=s390 isn't

I'm getting this messages trying to call with one sip trunk: Feb 3 16:43:09 DEBUG[3389] channel.c: Avoiding initial deadlock for 'SIP/usa-e2ea' Feb 3 16:43:09 VERBOSE[3491] logger.c: -- SIP/usa-e2ea answered Zap/1-1 Feb 3 16:43:09 WARNING[3491] channel.c: No path to translate from Zap/1-1(68) to SIP/usa-e2ea(256) Feb 3 16:43:09 WARNING[3491] app_dial.c: Had to drop call

1. My configuration: OS: ubuntu11.04, CPU: Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz llvm: llvm-2.8 2. My running environment: ~#llvm-gcc -DCOMPDATE="today" -DCFLAGS="" -DHOSTNAME="thishost" -DNeedFunctionPrototypes=1 -DSASR -DPROBES=8 -O3 -emit-llvm /home/qali/Develop/Benchmark/MultipleSource/FreeBench/distray/distray.c -o

Hi, Anyone is using LG ip phone LIP-100 with Asterisk. I've two of this phones but seems to work only with net2phone, in the product page http://isupport.lge.co.kr/html/ibu_lgic_modelView.jsp?jgrcode=D2_IPTP&modelid=M_IP100C the features are showing SIP and H.323 support. Can be used with my asterisk box? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq.

Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday).

Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")

Hi, I using asterisk 1.2.4 on a CentOS with Linux 2.6.9-22.0.2.ELsmp kernel. I've two x100p cards connected, only one card is reconigzed by asterisk. 02:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 02:02.0 Ethernet controller: Davicom Semiconductor, Inc. 21x4x DEC-Tulip compatible 10/100 Ethernet (rev 31) 02:03.0 Communication controller: Tiger Jet

From: "Guillermo Salas M." <gsalas at manta.telconet.net> > http://www.gizmo5.com/opensky Free calls are available up to 5 > minutes. If you need longer calls there's a commercial service you can > purchase. > Can be used to receive calls from skype? Yes it can. For example anyone who calls me now on Skype at michaelGizmo5 it will ring the IP phone connected to

Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone. Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone. <-- SIP read from xxx.yyy.142.234:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP

I downloaded and installed the none commercial g729 codec very often now I only disable HT on my systems I think * doesn't like this One of the guys @ digium advised me to turn it of, since they haven't written * to be multi treading any way The codec I download is the http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium4.so It should work fine. Wouldn't know what it

Hi, I have been making an RPM of FLAC to bundle with GStreamer. In order to get it working I had to make some rather hackish solutions in the SPEC file. The flac Makefile does to build into the correct directories while creating an RPM for some reason. I have attached the SPEC file I ended up with if it is of interest. Of course it didn't help me much cause it turned up we had a bug in the

I've uploaded Xapian 1.2.14 (including Search::Xapian 1.2.14.0), which you can download from: http://xapian.org/download You can see a summary of the most notable changes on the wiki: http://trac.xapian.org/wiki/ReleaseOverview/1.2.14 As always, if you encounter problems, please report them here, or to the bug-tracker: http://xapian.org/bugs Here are the SHA1 checksums of the released

Hi All, How to install bristuff on asterisk 1.2.14? install scripts are trying to download and compile those versions: asterisk-1.0.10 zaptel-1.0.10 libpri-1.0.9 and I'm running: asterisk-1.2.14 zaptel-1.2.12 libpri-1.2.4 I only need Pickup application from bristuff to be able to pickup channel independent calls e.g. when I have incoming call from PSTN and I would like to answer

We have an Ubuntu Dapper with 2.6.14 kernel, asterisk 1.2.14 debs from http://pkg-voip.buildserver.net When misdn stuff (misdn-init start) is not started, everything is fine, our 8 FXO (Channel 1-8) 4 FXS (21-24) are working well. If we start the misdn stuff (one card, port 1,2,3,4 in misdn-init.conf and no TELCO cable plugged in the card), the dialtone disappear on TDM lines :-( Does

I upgraded my Asterisk system to version 1.2.14 to check if the sound quality issues I was having with Chanspy in 1.2.7 remained. I'm still getting them, and I'm honestly out of ideas except from RTFS. The called party sounds normally fine, but it's impossible to hear the caller. Sometimes, when the called party is talking, the caller can also be heard. The conversation sounds broken,