search for: varsiti

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2015 Jul 23
2
Cisco 7940 and PJSIP registration
Thank you. I read that last yesterday afternoon, and I could've sworn I tried that but I will look into it again (I've tried so many different things it was getting cloudy what I've tried and what worked etc, combined that the extension config gets messed up after playing with it so much so I'm often recreating it as well). I also found a bug report in the FreePBX bug tracker
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
David, I should also note; 246 is my extension, it has IP 172.22.3.238. 172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway. The trunk is called ?testing? at the moment. The route that selects this trunk uses a 9 prefix. This system is in semi-production, so there might be fluff in the log from other active calls. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
Hello, So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly); exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}) If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing.. Here's a paste of a few things out of the two files that I
2015 Aug 18
2
Asterisk 13 chan_sip trunk appending @string to dialled number
Yes indeed. Do you have the dialplan, eg from /etc/asterisk/extensions.conf? Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 172.22.4.12. On 18 August 2015 at 16:21, Brendan Ord <bord at staff.onthenet.com.au> wrote: > Starting to make sense when I saw this line: > > > > [2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785 >
2015 Aug 18
5
Asterisk 13 chan_sip trunk appending @string to dialled number
Hello, I'm having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends '@CUBE' onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456 at CUBE chan_sip.c: Got SIP response 500 "Internal Server
2015 Jul 22
2
Cisco 7940 and PJSIP registration
I?ve gotten to the bottom of this; Seems that the pjsip.endpoint_custom.conf isn?t getting included properly, or my syntax is wrong. If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I?m using FreePBX, so it owns this file and my changes won?t persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F
2015 Jul 22
2
Cisco 7940 and PJSIP registration
Hi list, I've been googling this issue and found some good resources however I am still running into problems with the following combo ... Here's my story; - Asterisk 13.4 with FreePBX 12. - Migrating from Asterisk 11 / FreePBX 2.11 - Mix of Cisco 79xx handsets, mostly 7940G's. My problems started with (the very common) issue of the 7940 not replying to 401
2015 Jun 04
0
Differences between blind or attended transfer and impact on CDR entries
Hello, Sorry for a bit of a newbie post but we all had to start somewhere right .. I'm wondering if someone can briefly explain the difference between blind and attended transfers and why they would generate two very different CDR entries.? From my own research, it seems that transfers are both ultimately a SIP REFER and thus seeing two different CDR entries just confuses me further.
2009 Jul 04
0
soccer
How can I get good at soccer in a short time period? I'm 15 years old and I'm going to be a sophomore. I'm thinking of trying out for the soccer team. I'm not trying to get on varsity or junior varsity, but I don't want to look like an idiot on the field. I just want to know how to get better at soccer over this summer. I have a soccer goal in my backyard. Just some passing
2015 Mar 10
1
DND on a Polycom IP450
Only slightly asterisk related I suppose, but hoping someone has attempted this... I have an old installation with a bunch of IP501s, and one died. I replaced it with an IP450, and the user sorely misses his DND button. I hated those DND buttons anyway, as I couldn't control them centrally. I'd *like* to program one of his softkeys to send a *XX sequence to do DND on the
2016 Apr 01
2
Asterisk 11.22.0 Now Available
Kilburn Abrahams wrote: > Hi > > AU 1.5 core sounds are missing. > > ake[1]: Entering directory '/usr/src/asterisk/sounds' > --2016-04-01 07:59:09-- > http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-core-sounds-en_AU-alaw-1.5.tar.gz > Resolving downloads.asterisk.org... 76.164.171.238, 2001:470:e0d4::ee > Connecting to
2015 Mar 25
5
Call Quality Measuring
Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality? I?ve been playing around with ?sip show channelstats? but can?t other than measuring the packet loss I don?t really know what I?m supposed to be looking for
2015 Jun 22
5
Product CDR/Queue/Meetme
Gentleman, Moderators, i don't know if this topic if OFF-Topic, if yes, please tell me. I had some difficult looking for a Asterisk software that provide me some functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So i decided to develop than. In a few weeks i'll deploy a Beta version of this software and i'd like to know if is somebody available to try this