search for: asterisk13

Dears, I need to save some information on userfield when calls end in Asterisk13, but I have two error cases: 1. With endbeforehexten=no in cdr.conf, I have a registry in cdr, but userfield is not set. 2. With endbeforehexten=yes, I have two lines in cdr, one with duration, src e dst correct, and a second line with userfield setting and dst h. I am using cdr_odbc.conf, with A...

Hi, We are try new Asterisk13 and was noted it don't execute h exten priorities inside macros. We have a macro where we make all our call processing, and we use h exten inside it for billing (updating CDR(vars)). If context where that macro is called have some h extens, asterisk execute them. So, I wonder, h exten insid...

According to the instructions given at https://www.asterisksounds.org/de I converted and installed German prompts successfully and for numbers, I can successfully listen to a German female voice counting or telling the date/time. But unlikily, somehow the voicemail prompt is still English, although my general language settings are "de". I use pjsip.conf, not sip.conf. In

Scenario: Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisk's wiki, the transport section of pjsip.conf is configured as follows: ; Transport via UDP [transport-nat-udp] type= transport

...display/AST/res_pjsip+Configuration+Examples#res_pjsipConfigurationExamples-ASIPtrunktoyourserviceprovider,includingoutboundregistration my pjsip.conf: http://pastebin.com/raw.php?i=EA0PEcrb using tcpdump, I never even see a packet sent from asterisk trying to register. on the asterisk console: asterisk13*CLI> pjsip show registrations No objects found. asterisk13*CLI> pjsip show contacts Contact: <Aor/ContactUri...................................> <Status....> <RTT(ms)..> ========================================================================================= C...

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a normal two party call then MOH would no longer function? ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com...

Hi, I'm maintaining the FreeBSD ports for asterisk(With madpilot at FreeBSD.org as identity). Here's a link to the asterisk13 port for your reference: http://www.freshports.org/net/asterisk13/ I performed some tests with RC1 and am doing some final tests with the final release before committing the update. Up to now the ports forced using gcc, version 4.8 lately, to compile it. And for this update I'll keep things...

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (i...

Hello, does some people here use https://voicetrading.com which is a Dellmont service from Netherlands. At the high begining they were also known as Finarea (CH and DE mixed Co) Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our callerID is no more seen by them. We use Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or equal to CALLERID(num). We tried replacing + with 00, same problem. There support said they don't receive any callerID and we know that it's not...

Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 <?>: tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 <?>: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)!

...ples#res_pjsipConfigurationExamples-ASIPtrunktoyourserviceprovider,includingoutboundregistration > > my pjsip.conf: http://pastebin.com/raw.php?i=EA0PEcrb > > using tcpdump, I never even see a packet sent from asterisk trying to > register. > > on the asterisk console: > asterisk13*CLI> pjsip show registrations > No objects found. > > asterisk13*CLI> pjsip show contacts > > Contact: <Aor/ContactUri...................................> > <Status....> <RTT(ms)..> > > =========================================================...

http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ -- Dave Topping e: info at dntopping.uk t: 03445 888 888 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170901/ae060564/attachment.html>

...les-ASIPtrunktoyourserviceprovider,includingoutboundregistration >> >> my pjsip.conf: http://pastebin.com/raw.php?i=EA0PEcrb >> >> using tcpdump, I never even see a packet sent from asterisk trying to >> register. >> >> on the asterisk console: >> asterisk13*CLI> pjsip show registrations >> No objects found. >> >> asterisk13*CLI> pjsip show contacts >> >> Contact: <Aor/ContactUri...................................> >> <Status....> <RTT(ms)..> >> >> ==========================...

...hanks Richard. This is exactly the answer I was looking for. > > > I'm now assuming that Asterisk 11 was using it's equivalent > "bridge_simple" but I was getting confused because the only bridge module I > saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that > would have been the only bridge getting loaded at first. > > > Is it expected that if bridge_softmix handled a normal two party call > then MOH would no longer function? > That is correct. bridge_softmix is optimized for multi-party conferencing where passing control fra...

Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was