similar to: Asterisk stopped answering the calls

Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten => 997,1,Answer() exten => 997,2,Playback(tt-weasels) exten => 997,3,Hangup() exten => 999,1,Playback(tt-weasels|noanswer) exten => 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not

I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Answer ; Answer the

All, I''ve upgraded to 2.0.2, and I can''t get my login screen (the first POST request in the application) to work. When I post this form, I see the "InvalidAuthenticityToken" error. I have protect_from_forgery :secret => ''my_secret'' set in application.rb and I am using an active_record session store based on this line in environment.rb:

Hi, I running into the following problem on my Asterisk setup: --snip-- [Sep 3 01:40:59] NOTICE[9170] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from >

I have installed the CVS Head as of 12/12/04, as well as the asterisk-addons to ensure that /usr/lib/asterisk/modules/res_config_mysql.so exists. I have configured the following (after building a new DB with the appropriate SQL examples, with mods to drop the invalid keys, on the Wiki): - /etc/asterisk/res_mysql.conf [general] dbhost = 127.0.0.1 dbname = my_db dbuser = my_uname dbpass =

I am using a combo of static files and Asterisk Realtime configuration. This section works fine when a static file: --------------------------- [from_pstn] ;Voipgate exten => 4507,1,Goto(from_pstn,s,1) exten => s,1,Macro(dial-ext) exten => s,2,Hangup --------------------------- But, when I drop it in the database and try it in Realtime mode I get this error: ---------------------------

My porblem is incoing PSTN calls are being forwarde to the * box, the phone rings, but when the phone is picked up, the call is not taken - it continues to ring. I am forwarding the call to (<S0:6666>) in my dial plan Can anyone assist? This is driving my crazy! Extract from the * console Executing Dial("SIP/3001-047c", "SIP/2004") in new stack -- Called 2004 --

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address

Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial

Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

I have installed the CVS Head as of 12/12/04, as well as the asterisk-addons to ensure that /usr/lib/asterisk/modules/res_config_mysql.so exists. I have configured the following (after building a new DB with the appropriate SQL examples, with mods to drop the invalid keys, on the Wiki): - /etc/asterisk/res_mysql.conf [general] dbhost = 127.0.0.1 dbname = my_db dbuser = my_uname dbpass =

This has probably been answered somewhere, but I'm stumped. I have two Zap channels (FXS and FXO), both working fine. I can call from Zap/1 to Zap/2 and reverse. I've also configured SIP channels, both inside and outside of my firewall. Inside can call outside, and outside can call inside. Also, both inside and outside can make and receive calls to/from Zap/1 & Zap/2. What

Hi, all I'd like to configure Asterisk to receiving call from PSTN. After PSTN phone call in, Asterisk will prompt user to enter a number, then Asterisk will transfer the call to a SIP phone by this number. Please help me check the following extensions, is that OK? thanks! [from_pstn] exten => _.,1,Answer() exten => _.,2,GoTo(Xfer,s,1) [Xfer] exten =>