similar to: MOH over IAX2 - NOT working

The "error" I was talking about was in your log: "...== Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-6364'..." The call terminated here in a error which prevented the dialplan from continuing. Something there is broken, my recommendation is to check you registrations first inside asterisk: > sip show peers Something wasn't

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

> On Jan 2, 2024, at 23:17, thelma at sys-concept.com wrote: > > On 1/2/24 15:13, asterisk at phreaknet.org wrote: >>> On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: >>> I'm using asterisk-16.30.1 >>> >>> When I try to call another asterisk server over IAX I get a busy signal, >>> >>> chan_iax2.c:4739 __auto_congest:

Hi, I started looking into asterisk cause we're looking for a real-world solution. (when I say we I talk about a 50+ HQ and a 10+ branch office). We currently use a Panasonic analog PBX, with home-made IVR and PSTN lines. We'd like to deploy most of Asterisk's capabilities through out our organization - to save on long distance and international calls. I've been

I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c1222950155 Anybody

Is there a procedure for suggesting changes to HELP files of the core R distribution? If yes, what is it? If it would be considered a friendly gesture, I could find the relevant *.Rd file and submit a suggested modification to it someplace. Alternatively, I could just send suggestions someplace if they would receive appropriate consideration. On many occasions, I think of

Hi, I believe the issues raised by this message are the same as mine, more on a commercial sense than for self use, but mostly the same. I've seen posts where real-life installations are mentioned, but not a reference to how Asterisk is working on production (and productive) environments. Any experiences would be very welcome I believe, not only on pure technical, but wider, sense. Thanks

Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756]

Ok, I am running NUT in dummy mode. I have added a new ups in ups.conf [dummy] driver = dummy-ups port = upsc.dev desc = "dummy ups for testing purposes" upsc.dev has been generated by exporting the Elit configuration. I have added the monitor line in upsmon.conf and the "exec" flags (all the events) MONITOR dummy at artu 1 user pass master SHUTDOWNCMD

Hello. Is it possible to "mix" musiconhold music and playback voices? What i want to do is something like this: A person calls a number, gets a playback voice while in background music is playing. The configuration i use at the moment don't do what i want. Someone knows how to do it? Thanks in advance. exten => 18,1,Answer exten => 18,n,Background() exten =>

Hello, I have some Aastra 9112i's in production that almost function flawlessly. The problem I'm having is when a caller is put on hold they do not hear hold music. If they are on hold for too long (~ a minute?) they are hung up on. All other phones including Aastra 480i and Sipura/Linksys ATAs all seem to be working fine. Is this a quirk anyone else has experienced? Any

others have found this problem ? this possible bug can be used by user in that way that one password login can use 2 maildirs in filesystem effitively give them all space qoutas and lost of other goodies so to speak: foo at example.com with a password can login with fOO at example.com and fOo at example.com add more chars to get more mailbox :/ confirms ? i found the problem when i had

The ext3-2.4-0.9.17-2418p3 patch applies with 0 fuzz against both 2.4.18 pre 4 and 2.4.18 pre3-ac2. Will it be included into 2.4.18 pre 5? -- Ralf Hildebrandt (Im Auftrag des Referat V A) Ralf.Hildebrandt@charite.de Charite Campus Virchow-Klinikum Tel. +49 (0)30-450 570-155 Referat V A - Kommunikationsnetze - Fax. +49 (0)30-450 570-916 "Debugging is

$200,000 Refinance Home Loan for only $917/month at http://KFC.djk38.com Bad Credit OK! ============================================== place left. First we go this way." Redrick waved sharply in the direction of the nearest hill a hundred steps from the rocks. "Got it? Let's go." life you've led me by the nose, and I thought and bragged that I was living

FROM: COL. KENNETH ABELANGE. DEMOCRATIC REPUBLIC OF CONGO. Tel No: Your country Intl. access code +873762692483 Fax No: your country Intl. Access code +873762692485 kennethabelange@africamail.com Dear Sir/Madam <p>SEEKING YOUR IMMEDIATE ASSISTANCE. Please permit me to make your acquaintance in so informal a manner. This is necessitated by my urgent need to reach a dependable and

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)? I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new and not fully available in all distros. -- #Joseph GPG KeyID: ED0E1FB7

I've tried 2 builds of dovecot, based on a amd64 rebuild of the experimental package on http://xi.rename-it.nl/debian/ I've tried to include as much useful info as possible, sorry if I've missed out anything of use. Let me know as I'm more then happy to assist in any way possible. $ telnet host 110 Escape character is '^]'. +OK Dovecot ready. USER ian at domain.com +OK

Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS

When I type: asterisk -rx "core show channels" I usually get Channel Location State Application(Data) SIP/pstn-4444-000003 7807574622 at internal: Up Dial(SIP/77807574622 at pstn-9998 SIP/pstn-9998-000003 (None) Up AppDial((Outgoing Line)) Is there a way to pull information about time the channel started? -- Joseph