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Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic

Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---> Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send

Hi Guys! This is strange issue with 1.8 I have restarted my asterisk and it destroy all registered SIP peers now only solution is i manually reboot all phones to get them register back. I have never seen issue like this before. Any idea what would be the issue ? Thanks S -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

Hi I need to set the number of incoming calls to one, but the outgoing calls should be unlimited. I think the busylevel parameter is for it(incoming calls), but not works. My config is: cat sip.conf [general] [template](!) qualify=yes cc_agent_policy=generic cc_monitor_policy=generic call-limit=2 busylevel=1 callcounter=yes subscribecontext = hint allowsubscribe=yes [100](template)

Am 03.03.2015 um 18:16 schrieb James B. Byrne: > CentOS-6.5 (FreePBX-2.6) > Asterisk-11.14.2 (FreePBX) > snom870-SIP 8.7.3.25.5 > > I am having a very difficult time attempting to get TLS and SRTP > working with Asterisk and anything else. At the moment I am trying to > get TLS functioning with our Snom870 desk-sets. And I am not having > much luck. > > Since this

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

*I'm trying to build an application that provides statistics of calls*>* and call recording. Someone told me this could be done out of band*>* with a SPAN (?) port that would replicate SIP and media packets to a*>* separate NIC without having to actually pass the real-calls thru*>* asterisk. It was explained that this SPAN port would in the SBC*>* would replicate data