[asterisk-users] UPDATE instead of RE-INVITE

) Regards, -- Efficiency is doing things right; effectiveness is doing the right things (Peter Drucker) Miguel Oyarzo DevOps VoIP Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia

Re: [asterisk-users] Has anyone succeeded in making a WebRTC call from Mozilla Nightly to Asterisk?

You have to load the module res_srtp (secure media) in Asterisk. module load res_srtp.so (this is a requirement to talk websocket) If you don't have it, must build it and install it. Cheers, -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com

Re: [asterisk-users] sipgate outgoing calls

It looks like the challenge response after INVITE is not been accepted. Provide more detail. $ sip set debug peer sipgate -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia

Re: [asterisk-users] sipgate outgoing calls

Challenge authentication look good. --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Are you sure this number format 01179553708 is accepted in that SIP trunk? Some VOIP providers only accept international format. -- == Miguel Oyarzo DevOps Engineer

Re: [asterisk-users] sipgate outgoing calls

What you don't have mentioned yet is whether your outbound call reaches the destination. -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/19/2013 8:26 PM, gpxctawjc

Re: [asterisk-users] asterisk 1.8 sends SIP/2.0 481 Call/Transaction Does Not Exist to INVITE

. It looks like the other end is never closing the previous dialog?.. is Asterisk sending a proper 200 OK after receiving a BYE? NAT problem? regards, -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org

Re: [asterisk-users] Sending SMS from asterisk

and SMS to 0415340999 by dongle0. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/14/2013 9:15 AM, Asghar Mohammad wrote: HI bilal, I don't

Re: [asterisk-users] Sending SMS from asterisk

alteration). Anyway, 160-characters limit should be enough for most users. Cheers! -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/10/2013 2:10

Re: [asterisk-users] Sending SMS from asterisk

SMS (the recipient receives a single SMS only) chan_dongle works very well. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/9/2013 1:09 PM

Re: [asterisk-users] dahdi firmware for centos 6

-- ==** Miguel Oyarzo Senior [Network Systems Design] Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia

Re: [asterisk-users] Asterisk 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly

, -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 11/11/2012 9:46 PM, Eric Kuhnke wrote: Hi all, I'm trying to troubleshoot an issue with my SIP service. All outgoing calls

Re: [asterisk-users] Asterisk 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly

qualify=yes yes, correct. In addition, He might being notifying to the calling part incorrectly about the called part is behind a nat :) http://pastebin.com/1Mgn72vN (line 71: nat=no) -- ==** Miguel Oyarzo Senior Systems Design Engineer Linux User: # 483188