similar to: How to play audio to callee when a fax is detected ?

So starting up asterisk-11.5.1-2.fc19.x86_64.rpm I get: [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_mgcp.so': /usr/lib64/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_iax2.so':

2013/8/13 Administrator TOOTAI <admin at tootai.net> > Le 13/08/2013 16:41, Olivier a ?crit : > >> Hello, >> > > Hi > > >> [...] >> >> >> How can I work around this ? >> Suggestions ? >> > > Answer the call, wait few seconds and then ring Bobs extension. If > asterisk detect fax it already sended to fax extension so

Hi, I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these warnings: loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister loader.c: Error loading module

Hi All, This question is probably more for the LDAP experienced users/developers as I'm sure it would work fine if I weren't using LDAP (but I am, and I'm almost there with the setup!!!). I've setup an extension with the following: AstExtension: 210 AstApplication: Macro AstApplicationData: call-ext,SIP/testuser&IAX2/testuser,210 When I dial this extension, I see the

Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the "U" options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent destination for gosub: (Context:screen, Extension:s, Priority:1) Here are the segments

On 9/12/18 1:22 PM, Joshua Colp wrote: > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: >> I understand that HangUp() hangs up the calling channel. I want to >> hangup the called channel. >> >> SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. >> >> I send SIP/.... to listen to a long, very long, file. > > Define "send". How are you

The Asterisk Development Team has announced the release of Asterisk 1.8.28.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.28.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

I've have a simple caller id lookup on incoming: [teliax-in] .......... exten =>s,n,GoSub(set-callerid-name,0${CALLERID(num)},1) ................ [set-callerid-name] exten => 0,1,NoOp( no CALLERID num set) exten => 02135590993,1,Set(CALLERID(name)=Matthew ) ............................................... exten => _0!,n,NoOp(CALLERID: ${CALLERID(name)}) exten => _0!,n,Return()

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 =>

The Asterisk Development Team has announced the release of Asterisk 1.8.28.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.28.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.10.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.10.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hello using Asterisk 1.8.32. I notice that there is a spontaneous reboot of the Asterisk system from time to time. When I look in the logs (verbose file) I noticed that every time this occurs it's at a moment that there is a MySQL action, be it a lookup or an insert/update/delete. I must say I do have some MySQL queries that occur in my dialplan when a call comes in, to look up

Hello, I'm having problems with LIMIT_PLAYAUDIO_CALLEE in the Dial application. I want to play the limit file to both caller and callee at the same time, but it plays the limit file first to the caller and then to the callee. I searched the list and found someone with the same problem back in '06, but couldn't find any solution for the problem :( Anyone knows? Thanks, Best regards,

Hi, Is there a way to "announce" a call to the callee? For instance, I've answered an incoming call, collected some info and now want to ring an extension, and make an accouncement to that extension before connecting the caller through. Thanks, Steve

Hi all, I am setting up * for the first time, every thing is working fine, but I would like to implement an additional feature: Thus we have multilingual caller menu - I would like to play a little sound file to the callees to let them know in which language they should answer the incoming call, before I pass the caller through (caller should hear the normal ringing meanwhile). Is that possible?

try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To:

Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

yes, robert, but how do i "join" the two legs inside a call file or in the dialplan? i have experienced that call files can do a call to a channel and if this call is answered it can either be connected to an extension inside a context or call an application with parameters. roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Thank you, Robert! The announcementfile plays well, but at Dial-option "m" i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a