similar to: app_intercept

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

i would like to carry some oversea pstn-destinations via sip to providers like stanaphone, however, in case of a network-failure or if the provider is not available, i want to fallback to the zap-channels so the call is carried out to the pstn directly. the usual approach would be to check the dialstatus(e.g.NOANSWER). however, asterisk tries >60seconds to reach that peer(even when the ip

Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

I know in v1.6 its part of logger.c but I noticed this: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=11625 However, it doesn't seem to ever been applied to any version of 1.4.x branch.. Nor can I figure out what it was applied to? This is over 3 years old, you would of figured it would have been applied to 1.4 at some point in time.. Any ideas?

On 05/31/2016 09:17 AM, Denniston, Todd A CIV NAVSURFWARCENDIV Crane wrote: > Curiously my intel 845-G has had gl applications (freecad and wine > based windows games) STARTED working correctly with this update. > > I see in a latter email that you have a) found a path issue for users > vs root, and b) you are using nvidia. This reminded me of an issue I > was having with some

Hi ! I just wrote cdr_sqlite.c, see http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001986 This little app creates automatically the sqlite database file in /var/log/asterisk/cdr.db, creates a table 'cdr' inside it and inserts all CDR records into this table. Please comment. I'll use this in my project DESTAR (http://www.holgerschurig.de/destar.html) to show these

All, I am posting this here to announce I have finally managed to get my Cisco IAD2421 to speak MGCP with Asterisk. Due to an acute lack of reading on the subject as searched on Google, I'm putting this out with the hope that it helps whomever should need to do this in the future. This should also apply to the IAD2420 and the other models in the line, but as I do not have access to those,

Hello All, I posted a bug on the 14th of this month, and haven't heard anything back. However, I've since discovered that the problem is not in chan_iax.c as I originally thought, it's actually app_mixmonitor.c. Basically when I use 1.4.26.2 with an ilbc codec between two asterisk servers trunked via IAX, with mixmonitor Asterisk crashes on me. Here's a link to the post:

Hi everyone, Does someone know why the solution for bug 13115 (https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=13115) was made only for trunk? Having that this bug went solved more than a year ago, it means that all the 1.6.X.X branches have it applied already? Can this be backported to the 1.4 branch? This could be another good reason to upgrade to 1.6.0.16 after I do some

Hello, I recently completed a fresh install of Asterisk SVN-group-srtp-r183146M-/trunk , and I'm running into an issue getting the voicemail application module to load. Output from debug shows: ------------------------------- [Aug 11 22:00:01] NOTICE[20173]: loader.c:875 load_modules: 1 modules will be loaded. [Aug 11 22:00:01] WARNING[20173]: loader.c:376 load_dynamic_module: Error loading

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to

August 5th, 2005: Craig Southeren announced today that OPAL (http://www.voxgratia.org) now provides support for the IAX2 protocol(Written by Derek Smithies and released under the MPL). This support allows you to use chan_woomera (http://www.pbxfreeware.org) driver developed by Anthony Minessale II to interconnect your asterisk systems and use the IAX2, SIP, and H.323 protocols. I would

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

Dear listmates, I've recently moved a server from RHEL5 to CentOS 5.3 after it was decided not to renew the subscripton. Everything works beautifully except for Bugzilla, which throws MySQL errors. For example, the sanitycheck.cgi page says: # # # Bugzilla ? Sanity Check * Home * | New * | Search * | * | Reports * * | My Votes * | Preferences

I submitted a bug to the tracker *(*bug<http://bugs.digium.com/view.php?id=6947>) regarding the 256 character limit when copying a voicemail to a list of mail boxes. The bug was closed with this note: "Fixed in 1.2 branch, merged to trunk." Could someone explain to me what that means... in English? I searched the release notes of the newest asterisk version to see if this bug

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and

The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files