similar to: Musiconhold priority

Hi list, I'm stuck for the moment @ the following : My Query (in a macro) : exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\ vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\ SIPACCOUNT="${ARG1}") Asterisk CLI : [Aug 30 14:07:42] -- Executing [s at macro-vakantie:2] MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set

Hello list, this works : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Dial(SIP/${SIPACCOUNT}) The phone auto-answers the call... this does not work : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Page(SIP/${SIPACCOUNT}) The phone rings and does not auto-answer the call... Can you tell me

This is the core trace (gdb) bt #0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6 #1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828 "default", interpclass=0x0) at res_musiconhold.c:646 #2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 <Address 0x64 out of bounds>, interpclass=0x88 <Address 0x88 out of bounds>) at channel.c:4614 #3

Hello list, this is the dialplan : <snip> exten => s,n,Dial(SIP/test1&SIP/test2,,t) <snip> exten => 10,1,Dial(SIP/test1) exten => 20,1,Dial(SIP/test2) So there is an incoming call that rings SIPaccounts test1 and test2. Account test1 answers and wants to transfer the call to test2. Transfer is : #20 This is what the CLI shows : [Jul 2 10:55:30] -- Executing [20

Hello list, Can I do something like this for BLF functionality : [test-blf] exten => _XX,hint,Macro(GetSIPaccount,${EXTEN}) exten => _XX,hint,SIP/${SIPACCOUNT} GetSIPaccount is a macro that looks at a MySQL-database for the realtime table sip_buddies where the SIPusername is taken from. It works great for internal calls... but how about BLF-functionality ?? Feedback is appreciated !

---------------------------------------- From: "Joshua Colp" <jcolp at digium.com> Sent: Sunday, October 4, 2015 12:12 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:09 PM, Bryant Zimmerman wrote: > -- > Joshua > Thanks for your reply. It thought the same thing, but when I

Similar to commit 0bf8bf50eddc ("module: Remove const attribute from alias for MODULE_DEVICE_TABLE") Fixes many -Wduplicate-decl-specifier warnings due to the combination of const typeof() of already const variables. Signed-off-by: Nick Desaulniers <ndesaulniers at google.com> --- drivers/gpu/drm/nouveau/include/nvif/object.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

Hello list, I try to pick up a ringing extension but nothing works. To be clear, I'm trying to pick up extension 10. [Jun 14 17:37:34] -- Executing [**10 at from-TESTCORP:4] Pickup("SIP/testcorp3-00000041", "10 at 123456") in new stack [Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No target channel found for 10. [Jun 14 17:37:34] --

Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 Thanks -- -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf* [incoming] exten => 6000,1,Answer exten =>

Hi all, I'm seeking feedback (good, bad or indifferent) in regards to developing (further) a new class system for R, that uses encapsulated class definitions (i.e. the method definitions are literally inside the body of the class definition). A working (however very rough and poorly tested) system is available in my R package "oosp" with documentation in the vignette

I am having a strange problem with MOH. Say I have two users, A and B. I can set MOH in the extension for B and if A calls B and B hits hold, A will hear B's hold music. If however A hits hold, it goes to the default music. If I pull the setmusiconhold from extensions.conf and use musicclass in sip.conf under the peer A, I get the same thing. Peer A has musicclass set and A calls B and B

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( But when I used Vanilla Asterisk then All things are working.... Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf*

Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within

Hello, I've defined some new musiconhold classes in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [908001] mode=files directory=/var/lib/asterisk/moh/908001 random=yes ; [101001-1] mode=files directory=/var/lib/asterisk/moh/101001/1 random=yes ; [101001-2] mode=files directory=/var/lib/asterisk/moh/101001/2 random=yes But the new classes never show up

Hi List, I am trying to locate some detailed documentation and sample configs. I downloaded and compiled Asterisk, and I haven't been able to find much detailed docs on the config files. The distribution I compiled and installed doesn't have any config files, and the handbook is good but doesn't cover all of the configs. Here's my specific problem, when launching Asterisk for the

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

? Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want?to design a system where customers?can call my

Hello list, I have defined a new MoH-class in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; *[106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes* In sip.conf I have this commented out : ;mohinterpret=default ;mohsuggest=default Asterisk sees these moh-classes and files : vps2301*CLI> moh show classes Class: default Mode: files

Hi, I am writing a backend and having issues with properly lowering the result of getElementPtr ( specifically the add node that it generates). If we take this IR: %struct.rectangle = type { i24, i24 } ; Function Attrs: nounwind readonly define i24 @area(%struct.rectangle* nocapture readonly %r) #0 { entry: %width = getelementptr inbounds %struct.rectangle* %r, i16 0, i32 0 %0 = load i24*