similar to: Native Bridging and Polycom 600 Solved

Hi, A brief 6-step guide on how to hardcode a change in the Asterisk source that will allow it to work from behind a nat device. I know it?s messy, but it may prove useful to some people. 1. First punch a whole in your nat device. I just forwarded the port 5060 (for sip) and all ports between 10000 to 10020 (for rtp) to my asterisk gateway. 2. Now make sure your /etc/asterisk/rtp.conf correctly

Hi, Try as I might, I can't get hangups detected on a Zap channel with loop start lines. So, after someone leaves a voicemail and then hangs up, Asterisk doesn't know it, exits VoicemailMain2, and loops back to the corporate greeting, tying up the line even though the outside caller has hung up. Therefore, I've added the following hideous hack - er, code - to voicemail2.c. It

Hello, I've got a deployment with the SBC in between the clients and Asterisk (11.17.1 version). When the UAC tries to subscribe for "dialog" event package, the NOTIFY request sent by Asterisk fails. The SBC uses a different Contact (user part) for the 1st and the 2nd SUBSCRIBE (with Auth.). The issue is that Asterisk sends the NOTIFY with R-URI of the first SUBSCRIBE's Contact,

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

hello, I read http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large anybody could tell me more about this ? Is it available with ARA ? Regards Harry Method 3 Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone? A: In sip.conf create a section pointing at your

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

Revision 77616 Modified Sat Jul 28 07:44:16 2007 UTC (3 months ago) by rizzo File length: 681368 byte(s) Diff to previous 77538 make use of received= and rport= fields in sip replies. In a nutshell, these fields are used to tell a sip entity the address and port its request came from, and are extremely useful in the presence of NATs, especially with symmetric NATs where STUN is totally

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------

Hello, I've got the following configuration: 2 X101Ps Asterisk built with BUSYDETECT_MARTIN busydetect=yes busycount=10 callprogress=yes signalling = fxs_ks With this setup, the best I can do is get voicemail with 17 to 19 seconds of silence tacked on at the end. Ideally, I'd like at most 2-5 seconds. Has anyone had any success with this? It seems that hangups are indeed detected,

Hi, I cannot get two Polycom 600 phones to bridge natively. My sip.conf has canreinvite=yes for both phones. They connect, and I can talk as usual, but sniffing shows the RTP stream is routed through Asterisk. The exact spot where the attempt to natively bridge fails is in rtp.c, line 1281 (CVS from October 8, 2003): f = ast_read(who); if (!f || ((f->frametype == AST_FRAME_DTMF)