similar to: callforward with asterisk-gui.problem with stdexten

Hi gang, Since I'm getting no joy from device_Status or SIPPEER in 1.4.26-rc1, I thought I would do an AGI to read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet;

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten =>

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the

I currently have in extensions.conf: exten => 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,n,Monitor(wav,${CALLFILENAME},m) exten => 106,hint,SIP/106 exten => 106,Macro(stdexten,106,${HINT}) When I called x106 this was logged: -- Executing [106 at voicemenu-custom-4:1] Set("DAHDI/7-1", "CALLFILENAME=_xxxxxxx") in new stack --

Hi! Could someone give me a hand? If I dial 200 for echo testing it works... Everytime I dial an extension ex. 505 get the error below.... In this example it was from 508>505 a Xlite Pro to a TA. I believe it has something to do with the way i'm executing the command dial but I use the "standart" that comes in the samples from asterisk. *CLI> -- Executing

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

I am having a problem with an Asterisk server, in that when it is receiving a call from another Asterisk server using an IAX2 trunk the phone rings for 10 ms and then there is a hungup from asterisk and then the phone rings again before another hangup. The funny thing is that after I really hang up on the calling phone it repeats this as if I am still trying to call. Any Ideas?

Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

with the below defined in logger.conf on 11.6 cert 6 I am not getting any log message other than notice and warning in any files when doing module reload logger - queue log is the only one that says it restarts *CLI> module reload logger == Parsing '/etc/asterisk/logger.conf': Found Asterisk Queue Logger restarted built fresh box with make samples - added 2 stations, dialing from

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If