search for: asteriska

Hi, I'd like to setup oneway connection - so asteriskB can place calls on asteriskA and be safely authenticated with rsa keys. I just don't get any response on asteriskA. I've generated pair of keys: name.key, name.pub and put them on both servers - is it right to only have name.key on asteriskA and name.pub on asteriskB ? I get everybody is busy ... on asteriskB, a...

Hello, I'm traying to link 2 * servers using SIP and the following errors was show: "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13 22:47:07...

hello, i'm testing sending ISDN cause codes to customer pbx (test scenario for unallocated number) topology: PSTN-E1-AsteriskA-AsteriskB-SOMEPBX INVITE from SOMEPBX to PSTN AsteriskA sends to AsteriskB Status-Line: SIP/2.0 503 Service Unavailable X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 how can i resend HangupCauseCode from AsteriskB to SOMEPBX? i'm tried this on Ast...

Here is the setup: Phone -SIP G729-> AsteriskA -IAX G729-> AsteriskB -SIP G729-> Carrier The call completes but AsteriskA prints on the screen a ton of those "Dropping Frame of G729" messages starting about 5 seconds into the call: Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729...

Hi, Does anybody know how to get the SIP call ID of a "Dial" command? Thanks in advance. Ray -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071008/7f27548e/attachment.htm

...lerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no callerid=asreceived group=1 signalling=pri_cpe ; pri_cpe - kad se povezujem s provjaderom, pri_net kad sam ja provider (dva asteriska) channel => 1-15,17-31 When I call out this is what I see on CLI -- Executing NoOp("SIP/148-74a3", "Lama") in new stack -- Executing NoOp("SIP/148-74a3", "888") in new stack -- Executing GotoIf("SIP/148-74a3", "1?4:6") in new stack...