similar to: Using Goto with Asterisk Realtime configuration

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 =

We are using the following to record conversations. exten => _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP}) exten => _1XXX.,2,Monitor(wav,${CALLFILENAME},m) exten => _1XXX.,3,Dial(IAX2/4506:zj5S3A5a@nl.voipgate.nl/${EXTEN:1}) exten => _1XXX.,4,Congestion exten => _1XXX.,104,Congestion This was working previously to record both sides of the conversation but now

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 =

Hello all, Our Asterisk unofficial online forums (http://asterisk.xvoip.com) are moving forward and are alive. Thanks to all Asterisk guru's who helped to achieve it. Everyday more and more new members are joining this community, however not everyone is using mailing list or knows how to use IRC, this is why they are coming to web-based forums. We believe that together with Digiums

Hello all, Last couple weeks we had a lot of business discussions on mailing list, however some people don't like it, some people don't needed it, etc. I had couple discussions with Asterisk community members, who is interested to have business discussions about Asterisk, including but not limited to : business implementations, reselling , Asterisk commercial packages, IP phones,

well, asterisk unofficial forums are online since Nov 2003. http://asterisk.xvoip.com We have 500 registered members as of today and forum is alive. About 1000 uniqe visitors are hiting forum everyday. We will post link to your forums to provide users with more info. http://asterisk.xvoip.com I agree, the Wiki is Step #1 for any Asterisk user. The Wiki is a great reference, but you

Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000>

Did anyone here use the * forums over at asterisk.xvoip.com? I've been unable to connect for a few days now and was wondering if anyone knew if they're down for good. It'd be a shame if they are since * newbs like me need every resource we can find. Joel Moore

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

Why mailing list info was changed and no information about Senders email address in available anymore? Is this fight with spammers? It means from this moment, if we want to reply someone off-topic/off-list we can't do it, because we don't see Sender's email address, only name of person and we only possible way to do it , via mailing list. Thanks, Alexander E-Mail:

Please disregard previous posting. My mail client is nuts .. Thanks, Alexander -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Asterisk online forums Sent: Thursday, November 20, 2003 7:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Mailing list email masquerading. Why mailing

Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten => 806,n(light),noop(light) exten => 806,n,hangup exten

Hello, I just checked out the latest CVS and compiled and now get the following error: [res_config_mysql.so] => (MySQL RealTime Configuration Driver) Jan 26 13:03:51 WARNING[27081]: config_old.c:27 ast_load: ast_load is deprecated, use ast_config_load instead! == Parsing '/etc/asterisk/res_mysql.conf': Found Jan 26 13:03:51 WARNING[27081]: res_config_mysql.c:561 parse_config: MySQL

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

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

Hello everybody, I have setup asterisk real time extensions and its working pretty well. But the problem is when I am jumping between the contexts using the Goto statement in the database. I am getting a error = Parsing '/etc/asterisk/sip_notify.conf': Found -- SIP Seeding peers from Astdb: 'ezzibpo4' at ezzibpo4@210.211.246.47:5061 for 60

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

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 >

Hi. i am running asterisk 11 and i would like to have features access codes in my system such as call waiting(all types) (enable/disable), call forward (enable/disable) and DND. my dialplan is pretty simple and it is the following [DefaultPlan]exten => _XXXXXXXXXX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten => _XXXXXXXXXX,1,Busy() exten => _4XX,2,Answer()exten =>

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 =>