Ive MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension.Anything longer than 3-digits is cut off, example I dial extension 1000:[..
All;Im trying to install certified asterisk 11.6 cert16 on a Ubuntu 16server. However, when I try to compile it, Im getting hundreds and hundreds of errors. Here is a sample of the output. make: Leaving directory/usr/src/asterisk-certified-11.6-cert16/menusel..
Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone.Is there an better softphone?Or are there softphone solutions for PC desk..
Hey all,I have a setup with two analog lines coming into and Asterisk 13 box with a TDM400P and it takes a lot of rings before asterisk takes over. Ive traced this same box on two different analog providers so it probably isnt a problem with them.I..
I had meant to post a follow up to this, but just… didnt.Sorry.Anyway, I had made a silly change to my safe_asterisk script that caused it to start asterisk in the background, but also with a console.This caused asterisk to try to write to a non-exist..
I have connection with two networks (by VoIP provider setup)1 – 10.10.10.0/24 = SIP2 – 10.10.11.0/24 = VoiceHow to tell Asterisk send / receive voice traffic not on SIP network. When I look into dumps, I see Asterisk trying to use SIP net for voiceUnfortunate..
I just tried this in my extensions.confexten => **,1,Noop(Testing)exten => **,n,Playback(demo-congrats)Did a reload… and the above does not happen. I created as 12 instead of the ** and that works fine.Is there anyway to get the ** to work?I also..
One of my closest buddies, who happens to be a banker, let me in on a little tip earlier on.Theres this really awesome small bio technology firm which has discovered something ground breaking,and because of this unprecedented discovery theyre about..
Im trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures.For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesntS..
Trying to install asterisk 13 on CentOS 6.The ./configure tells me:configure: error: *** JSON support not found (this typically means the libjansson development package is missing)I dont really need JSON so I thought I would just disable it../config..