similar to: Reload problems with 1.8.27 and 11.9.0 - Someone else ?

Hello, I would like to use AMD on outgoing calls using analog line. I tested with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1 Other end is analog number behind another cisco/asterisk, also tested calling a mobile number with the same result. What I did: dial is done like exten => s,n,Dial(SIP/<IP gw>/<dialed number>,,M(myMacro)), which tell Asterisk to

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, I have patch failure trying to apply asterisk-1.8.26.1-patch.gz I took a look in the patch and found those lines at the begining: --- asterisk-1.8.16.0-summary.txt (.../1.8.16.0) (revision 410440) +++ asterisk-1.8.16.0-summary.txt (.../1.8.26.1) (revision 410440) @@ -1,221 +0,0 @@ - Release Summary - - asterisk-1.8.16.0 - -

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] --

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

I have used every asterisk 11.8.X version. Have not had an issue till 11.8.0 and 11.8.1 When I use ConfBridge I am attempting to put all participants in MUTE mode and just one talker... However, since 11.8.0 I am hearing feedback in the announcement like the channel is not really muted. I dropped back to 11.7.0 and I hear no feedback. Has something changed - or - am I not correctly setting up

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})

Le 08/07/2015 17:36, Richard Mudgett a ?crit : > > > On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > Hi list, > > we wanted to patch our servers with 11.18.0 patch against 11.17.0 > actual running version. Patch failed with > > zone-s:/usr/src/asterisk-11.18.0# patch

Hello, My target system is : PSTN <---> Sip Provider <---IP/ADSL---> Router with fw/NAT <--- SIP/IP/Eth --> Asterisk <--- SIP/IP/Eth --> SIP Phones Asterisk is configured to keep NAT connection with SIP provider open (with qualifyfreq) so I don't have any problem (yet) with either casual incoming or outgoing calls. To work around a possible No Audio when an incoming

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

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve

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

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

Good afternoon Best regard I'm having trouble with a CentOS server release 5.10, so that my users connect via VPN Intranet type, I could not find a solution, if I can collaborate appreciate them, attached logs when it worked and now. Log running: Mon Oct 10 13:50:02 2016 193.60.90.72:23683 Re-using SSL/TLS context Mon Oct 10 13:50:02 2016 193.60.90.72:23683 LZO compression initialized Mon

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

Hopefully someone can point me in the correct direction. I had a 1.4x system die on me yesterday, while I was prepping a new machine to replace it. Took the machine on site yesterday and spent the day and part of the evening getting things working. This morning, I finished up converting my dial plan, knowing there'd be calls of things that I missed. While testing, I've noted that all

Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number.