Archives : March-2014
Home » 2014 (paged 43)
Am looking for a service provider who can provide enterprise SIP trunk with100 channels concurrent sessions.I see some like Inphonex, Broadvoice… and etc….Is there any suggestions for the service provide..
everyone,I would like to seek your advice regarding a build (or rather configure)problem I am running into. For reference, tests are all relative to a build from a 1.8.26.0 tarball, on Debian Wheezy.I would like to understand if it is possible, and..
externhost is monitoring for ip changes with an interval of externrefresh, so far so good.Wouldnt it be handy if asterisk would do an sip reregister if it detects an ip change?My SIP provider has sometimes very high intervals of 1 hour and if ip chang..
I was successful in compiling asterisk in raspbien except for the following error If I enable the gsm codec. It appears there is something in the Makefile n this directory that needs to be changed. Probably involving optimization. Not sure why it d..
Im tinkering with Asterisk for * for about 12 years now and since about10 years, its my home PBX. I was off the list for something like 7years – had other things to do. But… I remember, then, sometimes came over 1000 mails in 24h. Now its hardly..
Hi I have been trying for several days get 3 Cisco spa508g phones (firmware 7.5.5) to work with asterisk 11.6 cert1 and sla. I can get the phones to all ring when an incoming call arrives, and I see the slatrunk working. However the blf funct..
Everyone,We are looking to transition our 23 channels from testing/lab into production. During testing we used the free open source g729 license using the instructions found here:http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asteris..
Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E cant hear, is there a way to suspend D and E for a while (whilst t..
Hi I am trying out Asterisk 12 between two servers in order to test the 100rel (PRACK) capability. When I set the called server to send Ringing (180) and wait before answering the call it is not sending 100rel Require in the Ringing message (non provisio..
HiRe raising this issue as its still affecting me.Where is the asterisk server getting port 0 from? We use ARA and port 0 is neither in the full contact not in the port field of the sip table. Nor is port 0 in the realtime cache for any peer register..