similar to: oh323 context for peers

Hello list, as I'm writing a little perl parser for queue_log analysis, I'd like to know *which* telephone answered a specific queue call. Unfortunately app_queue only logs the call id but does not log the call end point. This is okay for SIP endpoints, because their call id is something like SIP/endpointname-1234 so you can reasonably understand who was on answering, but for

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout

Hi, how can I disable event_log in order to reduce hard disk activity? I can't find any hints in conf files. Must I hack the source code or even use brutal methods like creating a dir called event_log in the log dir, in order to prevent asterisk from creating an event_log file? (Just chmod a-w event_log does not work, unfortunately.) Thanks for any hints! Roger.

Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to

Hello, everyone. No need to read this message. I'm posting for documentation for other poor, ignorant slobs like me who are struggling to pull together the many technologies to make converged networks happen. Hopefully, this will help save someone else the time I spent. I started the below email until I realized I had solved multiple parts of a compound problem but not all at the same time.

Hi, I have unpleasent short audio gaps when a perl based agi scripts starts. Thus, I now started to put all those things in C programmed daemons for fast-agi. Anyway I'm looking for another mean, which would help me more quickly. I noticed, that all agi scripts are running with system priority -11, like asterisk does. This is really waste of priority. I would like to have the AGI scripts

Hey, everybody--- Ignorance CAN be bliss, at least for a while, but, .... Just so you know... A week or two ago, some upgrades to the expression parser (you know, the expressions you put in $[ ... ] in your extensions.conf file) that I submitted, have been merged into the CVS HEAD of the source. Hopefully, for around 99.9% of you, it won't make any difference to you. The Makefile has also

PJISP cannot handle the From field when it does not contain a number. Can this be fixed? [Feb 25 12:35:43] ERROR[7143]: pjproject: <?>: sip_transport.c Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax error exception when parsing 'From' header on line 4 col 40: CANCEL sip:14408785990 at 162.255.138.102:5060 SIP/2.0 Via: SIP/2.0/UDP

Hi, some days ago we discused here the need for an analyzer for the 1000 Hz tone, as opposite application to Milliwatt. Here it is: Mwanalyze http://planinternet.net/download/voip/asterisk/app_mwanalyze.c It performs a Fourier analysis for a fixed frequency and tells the amplitude. The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple

Hi there, I was looking for cri-o 1.13 package to install on CentOS 7.6 but it seems that we only have 1.11 packaged. There is any alternative to 1.13 package to CenOS 7.6 or if I want to use it I should build from source? Thanks. -- Matheus Eduardo Bonif?cio Morais Analista de Infraestrutura de TI, Plataforma e Aplica??es Confedera??o Sicredi Centro Administrativo Sicredi ? Porto Alegre +55

Hi, I'm looking for a mean in chan_h323 to jump to a specific context dependent on the remote ip address. E.g. an argument, let's tell it "ignore_h323_name", in h323.conf users like this: [BillyBob] ignore_h323_name=yes type=user host=1.2.3.4 context=path1 in a way, every incoming call from ip 1.2.3.4 will fit this user, not only when the H323-name is BillyBob. Or a variable

Hi, Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could someone tell me the cable pinouts to make this conection? thanks Eduardo

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name |

Hi, I am working on a H.323 project which involves GnuGk and Asterisk My current goal is to provide IVR functionality for the H.323 users which register through GnuGk(eg. call credit information) I have successfully built a H.323 platform using GnuGk - it uses SQL accounting and authorisation. Now I am trying to integrate it with Asterisk in order to provide IVR functionality as I already

Hi, I have to develop a phone application using asterisk's chan_oss. When the phone is idle, i.e. the last command was a hangup, one hears a "toot, toot, toot, ..." But unforuntaly its use is in Germany, where one expects a continous "toooooooooooooooooooooooooooooooooo ..." before dialing. Is there anything to define the tone indicating "ready to dial"?

Hi, I compiled asterisk and chan_h323 on an Opteron in 64 bit mode. In the h323's Makefile I replaced in line 24 CFLAGS += -march=$(shell uname -m) by CFLAGS += -march=k8 and also tried CFLAGS += -m64 -march=k8 Both solutions do compile, but when starting asterisk, a load error occurs: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi When I grep

Hi all, I have one asterisk server with one ISDN BRI connection to PSTN, with h.323 support (oh323) I buy some voip phones, and I connect them to the same switch as asterisk server is; all is at the same TCP network. I need to route some extensions from my DDI (DID) line at asterisk to some voip phones, and also to do: outgoing calls from any voip phone to PSTN via asterisk, and intermediate

Hi, > chan_oh323.so: undefined > symbol: __use_ast_pthread_create_instead__ is not a bug, it's a hint: use "ast_pthread_create" instead [what your were using] and means: replace in asterisk-oh/asterisk-driver/chan_oh323.c at line 3764 "pthread_create" by "ast_pthread_create" Roger.

Hi List, does somebody know a SIP Provider which offers reseller possibilities? Moritz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040828/bddb5898/attachment.htm