search for: schreiter

Hi all, Does somebody know what's new with SS7 and * ? I'm very interested. Is it ready ? I'm prepared to pay if necessary. Thanks, Angel.

Hi, I'm trying to get H323 support using asterisk 0.4.0 Unfortunately the pwlib and openh323 versions mentioned in the asterisk-oh323 readme file are no more available, and I had to use newer ones. Now I installed all libraries, but got a segemntion fault when starting asterisk while reading the chan_oh323.conf file. When I declare more than 9 gwprefix I get first a error "out of

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

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, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find

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

Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

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 found some contradicting infos about pass through of T.38 data. Are there any experiences of just passing T.38 via SIP from one T.38 application or gateway trough asterisk to another T.38 application or gateway? Would asterisk maybe even pass T.38 from chan_oh323 to chan_sip (without understanding the content)? Please tell me, if you have knowledges or experiences on this topic!

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, one month ago, I announced, that I will look at the openss7 project in order to use it together with asterisk. It took a while for me to check the capabilities of and around the project. Since the openss7 project consists of only one person, and when there was just silence for a long time in the project's mailing list, I doubted, whether it was a good idea to consider using openss7. In

Hi, I intend to change the cdr_mysql-field "uniqueid", which seems not to be used so far, to an (not unique) indexed field and use it later for my own hints and infos. I don't have very much traffic so far, and I wonder, if there will appear problems when asterisk is under high load (100 simultanious calls) and the log table contains 1.000.000 log lines. This would mean, that

Hi, I just bought the E100P from digium. It has both keys: 3.3V and 5V, so it would fit both, in a 5V-PCI slot and in a 3.3V PCI slot. Is it true, that I can plug it without destroying it in an ordenary 5V PCI slot? Roger.

Hi, there are still some questions to be answered by OpenSS7.com in order to decide, whether E400P-SS7 is a good choice for the asterisk SS7 support. In the meanwhile I'm also in negotiations with another manufacturer (whose name I currently may not tell due to a NDA) of SS7 hardware, who gets likely persuaded to offer a cheap SS7-PCI-card which would be suitable for asterisk. Asterisk users

Hi, I'm using asterisk with chan_h323 together with gnugk. chan_h323 and gnugk were recently compiled with pwlib-1.5.2 and openh323-1.12.2 as advised. When connecting asterisk directly by ohphone (without gatekeeper), everthing is fine. When using gnugk for usage control in routed mode, I find a funny situation in asterisk's H.323 debug: == New H.323 Connection created. --

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, > 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.

...26 +0000 (UTC) > From: tony@softins.clara.co.uk (Tony Mountifield) > Subject: [Asterisk-Users] Re: how to fetch a call? > To: asterisk-users@lists.digium.com > Message-ID: <cgnfpm$56k$1@softins.clara.co.uk> > > In article <412F4122.6070401@planinternet.de>, > Roger Schreiter <roger@planinternet.de> wrote: > > 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". &g...