Archives : October-2016
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I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp.I have another SIP trunk thats wants to run on port 5068 (long story). I have enabled tcpenable=yes in sip.conf and defined portP68 in my trunk definition. It does not seem that anyth..
This is a multipart message in MIME format.——=_NextPart_000_002D_01D22734.3A533540Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: 7bitHow can I lock a device state so it can only publish AVAILABE, BUSY, or RINGNING? (Eg, if ..
Are these incoming calls copper or VOIP?If you only accept copper calls, make sure Asterisk is only listening to 127.0.0.1 and enforce this policy with another layer dropping any incoming SIP packets at the firewall.If you only intend to accept ca..
I have a host 192.168.1.3 that wants to run SIP on 5068 (long story). My host is 192.168.10.201. My host needs to stay on 5060 because of all the other devices I have connected.I tried putting portP68 in my SIP extension definition but that did not work..
all, I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall related, but Im unsure.A registration to Sipgate is established successfully: =========================================================================================pjsip_sipgate/sip:sipgate.de:5..
Apparently Verizon is blocking or changing packets on port 5060 so my softphone from my hotspot will not work.How do I set asterisk (11.23.0) to run default 5060 for all other devices Ihave – BUT for my software run on a different port like 5070?..
What part of Asterisk 14.0.2 opens the random, high numbered (33094 currently) UDP port? This port is opened even without any channel drivers loaded.$ sudo netstat -ltunp | grep asterisk udp00 0.0.0.0:51488 0.0.0.0:* 13830/asterisk udp00 0.0.0.0:5..
fresh install of Asterisk 13.11.2, client unable to register.For now I have IPtables disabled, also selinux is disabled [1006]type=friendusername06secret=mysecretcontext=sip-phonecall-limit=1callerid=iuser disallow=allhost=dynamicallow=all any ideas?Than..
i recently purchased a Wildcard AEX800 digium card. Ive installed asterisk 13 and all prerequistses on ubuntu serv14.04 LTS. Dais the driver am using; ive configured all but when i call from PSTN through fxo port an not getting anything in logs or..
list, I have Asterisk running well inside our network. I did some experiments exposing it to internet but had some issues: 1. NAT issues (voice one way, etc). From what I understand double-NAT users will always have something like this 2. Immediat..