similar to: using ast_request("zap", format, "pseudo")?

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

Hello to the list, I have a question about Asterisk/app_rpt: When a signal is *not* being received and or transmitted by the radio system attached to Asterisk/app_rpt via its interface, is the incoming and or outgoing data suppressed (silence suppression)? In other words! At idle but still connected to some other device, what data will be transferred? I currently don't have access to a

Hy, I have a network with WIFI communication and VHF/ HF channels. I have integrated asterisk in the network using SIP, ZAP and IAX2 channels for WIFI communications, but I don't Know How I could integrate the VHF/ HF channels. I have heard speaking about app_rpt project, but I don't Know very much about this. Could I integrate VHF/ HF channels with this application? if the answer is

"app_rpt is an application which comes bundled with Asterisk, however, a later version may be available on our source repository. All you need to do is go to asterisk.org, download asterisk, configure the asterisk to build app_rpt by modifying the Makefile in the asterisk/apps directory, and then compile and install it. You can get the latest version of app_rpt.c along with the sound files,

On the various technical issues raised here (OK, we posted the rules, we won't discuss the legality anymore) I think there is only one main obstacle to using FRS radios for extensions on *. They are simplex (push-to-talk, release-to-listen). The protocol for dealing with voice-activated-switching (VOX) has been used in ham and public safety simplex autopatches but it's really tricky

Hi all, It seems that using exten => _*0. is not possible in extensions.conf. I changed disconnect => *0 in features.conf to something else. From what I can tell with the little C knowledge I have is that it's caused by a hardcoded *0 value chan_zap.c. Line 5730 of chan_zap.c (svn rev 1077) shows: } else if (!strcmp(exten, "*0")) { struct ast_channel *nbridge =

Hello, The next Asterisk Users Group meeting has been scheduled for this Saturday May 13th at 11:30am. Meetings are held monthly on the second Saturday of each month, excluding July and December. Meetings are held at Sound Choice Communications LLC... http://maps.google.com/maps?oi=map&q=7839%2012th%20Ave%20S%2055425 Sound Choice Communications is located in Bloomington Minnesota, just

Pete VK2YX writes > I've been involved with IRLP for about 5 years and am one of the original > install team. I've gone through the emmotions of allowing "other" networks > connect to IRLP and I know its caused some lots of headache. > > As far as a closed network goes, yes there is LOTS of passion to keep it HAM > only and I'd have to support that notion.

Greetings, Where can I get the chan_echolink channel driver from? I've seen reference to it, but have yet to find a place to download/compile it. It is part of the app_rpt.so module... I am told, but do not see the source with app_rpt. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I need to setup call using particular timeslot on one of E1's. I've looked into http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and it says that: exten => TestTrakt,1,Dial(ZAP/1-2/517255333) exten => TestTrakt,2,hangup should work and force call setup via span 1 (port 1) but when I try setup call rasterisk says: -- Executing [TestTrakt at

Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help

Dear All, I am not sure if this is the right place to ask my question but I can't find a newsgroup or support for this app_RPT concept so I hope if some one in this community who have tried it out could help me out. I studied this application requirments and saw the hardware needed they describe a radio quad which uses RJ 45 but I can't see where the RJ goes in order to be able to

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

You don't need to reinvent anything to tie radios to *. Ham systems like IRLP, Echolink, eqso etc all have fairly tight controls to keep from being abused (although with a little Linux knowledge, the IRLP package can easily be used to set up your own network using their protocol). Jim Dixon seems to have done the work to integrate radios with * already. See

I'm tring to compile fax applications on Debian system. the spandsp library compiles ok, and when i try to patch the make file in apps directory as is said in the instructions it returns errors. I'm using cvs version of asterisk . -------------------------- voipgw:/usr/src/asterisk/apps# patch < Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of

In article <mfbt6f$9rt$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that > is behind a network device to which I don't have ready access, which is > performing NAT with possibly some kind of SIP ALG, and an Asterisk 11 > system on a public IP. > > My question is

In article <n009u2$85v$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > Apologies, this is slightly off-topic being to do with an EPEL package, > although it's running on CentOS6, so I thought others here might have come > across this issue. > > I have five CentOS 6 systems running fail2ban from EPEL, and this > package was updated

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)