similar to: How to tell Asterisk to to send Ringing signals as into RTP

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem

Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been

Hi all, i use asterisk 11.14.0 and I suspect that the voicemail application counts the time wrong. In my voicemail.conf: [general] minsecs=3 maxsilence=5 format=wav maxsecs=180 silencethreshold=140 [...cut..] In the asterisk-cli: [Jan 26 15:23:49] -- Executing [s at macro-voicemail:77]VoiceMail("SIP/XY-0005175a", "aNumber,su") in new stack [Jan 26 15:24:04] --

Hi All, I've been looking for information on how to use asterisk and early media to allow for a video-preview of the caller at the callee's phone for days... but I haven't been too successful :( I found that there seems to be a company "2N Helios IP" which claims (youtube-video) that "their" SIP server is able to provide early video (using a Grandstream 3157v2

Greetings All, Regarding this archived post. http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html Did anyone ever find an solution to this? I've got a new box running 13.3.0 with the exact same issue. For those that don't read the link. I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, These are loaded into asterisk without

hi: our current connection is below: sip phone<--->asterisk<---->alcatel PBX<---->PSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3.

Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip reload" or "iax2 reload" does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working

On 26-10-16 15:03, Dan Jenkins wrote: > > > On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > > I keep getting the following error when trying to connect to the > Asterisk server using AMI : > > $socket = fsockopen("tls://11.22.33.44 >

And I think they're both small. Solved: tcpdump showed no packets coming in, so I went to my DID provider's Website to discover to my intense embarrassment that the DID number had been set up forwarded to their voicemail. I got egg on my face for this one. I changed that setting to SIP/IAX and packets now arrive and go where they should. Two problems remain. 1. Still can't

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

Hello, I wonder if anybody is using PJSIP realtime in production environment? I've started to play with it and encountered many problems. Here's my config: sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints extconfig.conf: [settings] ps_endpoints => pgsql,users,pjsip_endpoints_v pjsip_endpoints_v is postgresql view. 1. The biggest problem: if I have small number of endpoints

Hi list! On an Asterisk-Server I have some users. Just two of them have a Mailbox. I want to write a little Web interface to manage many things and I'd like to have a menu point for the voicemail, but just if the user has a Mailbox. I found the AMI-Command MailboxStatus, but it does not return what I need, since it returns 0 if the user has a Mailbox but no messages and if the user has no

I'm using BLACKLIST() to check numbers, which does not like leading + signs. I want to test if there is a plus sign, and then remove it. I tried: ; strip leading plus sign same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} ) same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) = ${CALLERID(num):1})

Well, what immediately stands out is: "FILE * open failed!" Have you triple checked that the full filepath is correct and that the user that Asterisk is running as has full permissions to access your valid certificate file? I have it working with microsip and a free TLS cert from LetsEncrypt. When I get to the PC with that on, I can write up what settings I've got if that helps?

Hello, Years ago, I used Monit to monitor Asterisk and restart it whenever it failed. Now, I wonder which tool I should pick for an Debian 8 (current) or CentOS 7 (future) environment. The main reason I'm looking for this tool is to avoid as much as possible, current 5 minutes delay between Asterisk's stop and first cutomers complains. 1. I always install Asterisk from source but

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

Hello Although not stictly a asterisk issue, any help would be apreciated. Firstly a few notes on the snom 360, which I have had on a test bed for the last week. Its a great phone, with a good user interface, both physically and its web based one. At its lastest firmware it does have a few quirks, with regards to the way it handles usernames and passwords on the physical interface. These have

Hi, can someone sched a light what exactly mean the read write permissions in manager.conf? [public] secret = private deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.255.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Lets say I want some users to use dial through manager interface. But don't want to allow them to run asterisk commands?

All, Using SIP with progressinband=yes I get Asterisk to generate the ringing sound for callers. However, I was wondering if it is possible to configure what is 'played back' to the calling party? i.e. instead of just 'ring ring' could I potentially play back a song from an MP3, WAV or GSM file? I'm thinking it would be quite cool to offer a customised 'ring'