> I am facing an issue with Peer registration in my asterisk server . > > I am using asterisk version 220.127.116.11 and using SIP real-time > architecture.when i am doing registration it registered fine on asterisk > as peer is available in Database. > > ..
,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [22..
Im currently testing my FreePbx Box to work with our Avaya PBX to allow dialing outgoing international call and FreePBX extensions to avaya PBX Extensions calling. Unfortunately no luck to do it successfully. Any help would be much be appreciated, h..
Can anyone let me know how I can enable video (h.263) on SIP, but if a video call is passed over IAX, it will remove the video and pass the audio only. What I tried was: SIP – videosupport=yes – disallow=all – allow=alaw – allow=h263 IAX – disallow=..
These are the same for sip users and trunks disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 Who is asking to transmit frame type slin ? Nick On Thu, Mar 10, 2011 at 1:02 AM, Paul Belangerwrote: > On 11-03-09 02:26 PM, Nick Ustinov wrote: >> Us..
This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and havent been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on ..
Hi! My customer wants to allow calls to landlines in EU and US and disallow calls to cells in EU. Rest of countries are blocked. Country blocking is easy… Is there a service that allows checking phone number? Maybe some specific Enum? I ask for num..
Hi! I want my PBX to be reachable at my ekiga.net account. It seems I am registered: vajna2*CLI> sip show registry HostUsername Refresh StateReg.Time ekiga.net:5060magwas 585 Registered Sat, 13 Nov 2010 13:48:22 However when others try to call firstname.lastname@example.org..
All, I am running asterisk on Linux machine and trying to use confbridge application. Please have a look at Conf files. sip.conf ====== [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow = all allow=ulaw allow=a..