similar to: asterisk13: no voicemail prompt in German

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's

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 Asterisk11.14.0 it

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 inside macros was deprecated? Thanks in advance. Atenciosamente,

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

We have a couple of parallel ring settings (and this has worked well for eons). Either in the form of same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..) Or via a subroutine (below) that has a bit of extra logic: FOO = 1010 & 1019 & 1017 & 1033 ... same => n,gosub(sub-callout,s,1,(${FOO},”Ringing all class FOO telefons")) Now I have two types of phones

Hello there, I appreciate any help about this problem that I can't figure out... I need to record all my calls: this is pretty easy using Monitor() before the Dial(). eg: exten => 425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes

Is there a recommended way to find out the cause of DIALSTATUS = CONGESTION for PRI/BRI channels? Currently I am evaluating the DIALSTATUS variable and I also count the active ISDN channels for the ISDN trunk in question. Counting the active ISDN channels seems somewhat clumsy as the mapping to a specific trunk must be done by hand (or write even more code). I have a setup where outgoing calls

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

Hi List, Merlin Magix hardware v02 I'm trying to get asterisk to act as a voicemail server for a lucent merlin magix PBX that we purchased used. We have 4 FXO channels between the two PBXs on a Sangoma A200 card. The 770 dialgroup is working properly, in that calls to 770 are answered by Asterisk. The magix is sending mode codes in the format #XX#XXX#, where the 2nd block of digits

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called

Hi, i have a new Samba Fileserver. The problem is with a folder permissions the folder permissions are 1596981296 4 drwxr-x--- 6 oliver.werner it_secuirity 4096 Jan 3 2014 Security In this folder are some other folders and files I can open this folder but looks empty in explorer on windows or finder on macOS. when i change the

Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4, I can't make asterisk try to send a register. I have configured my pjsip.conf similar to https://wiki.asterisk.org/wiki/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

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

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

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 (important parts) [vr1a882] ... nat=force_rport,comedia

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