Archives : May-2016
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I have an Asterisk 13.8.2, which is supposed to be only a client to an encrypted SIP service. All local phones are connected via UDP.Since I cant use PJSIP (see my mailing list post from yesterday), Itried configuring chan_sip to work that way. My setti..
2016-05-03 16:43 GMT+02:00 Matt Fredrickson :OKYes, I think issue must come from incorrect Audiocodes settings. Requiring T.38 settings within first INVITE seems very unusual.Thank you very much fo..
While I am executing a Macro on the called channel, right after the call connects, I need to execute an app on the master channel, from inside that macro, specifically, SendDTMF. If I execute it now, it send a text message to the Callee, when..
I posted this over in asterisk-dev, realized I probably should have put it here. there, We’ve been having a strange issue with a customer’s queues where a queued call will ring an available agent, agent answers, then a second or two later the ag..
Howdy everyone,Im writing a little click to dial type tool and Ive run into a snag where my Originate command needs to call a Sub routine to do a database lookup and some other stuff.I cant seem to get the syntax right to call Gosub with OriginateJ..
I am trying to do an install of asterisk by source… dahdi 2.11.1, asterisk 11.22.0 libpri-1.5.0 on CentOS 6.7(not my first time doing so, but this is a remote system someone else setup)All installed well, compiled fine but when I run asterisk it..
Im registering an Asterisk against my TLS capable service, using res_pjsip. My config looks like this:[devtrunk_reg]type=registration out..
I try to find informations concerning Mixmonitor command, but … without success. MixMonitor command take at last parameter command. This command can be a shell script.When record is over, and this command executed, asterisk wait for a return code..
all,im experiencing a really frustrating issue with my Asterisk 13.7.2 with realtime configuration on MySQL and Voicemail.Heres res_config_mysql.conf:/[default]////dbhost = 192.168.1.1////dbname = asterisk////dbuser = asterisk////dbpass = [xxxxx]////dbp..
Hello!I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2.As customer of German Telekom, I have three numbers and therefore three trunks – each number is bound to one trunk.After migration, some callers complained about missing ringback tone:t..