search for: kister

...<thanks darren!>) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip set debug peer vgw1 (vgw1 is my cisco 1760 ata) http://jeremy.kister.net/tmp/fax/console.txt http://jeremy.kister.net/tmp/fax/messages.txt http://jeremy.kister.net/tmp/fax/sip.txt I've tried using the packaged app_fax_spandsp and also Digium's app_fax_digum for 1.8.0-rc1 -- no difference in behavior. Anyone have any ideas how I can get this fixed? --...

...e playback to complete before dialing the DTMF, all works as i'd expect. Do you see a reason that this'd happen ? I wrote to the author of app_swift, but got no reply. Since the code is relatively short, can someone take a peek ? app_swift is [temporarily] available at: http://jeremy.kister.net/code/app_swift-1.6.2.tar.gz http://jeremy.kister.net/code/app_swift-1.6.2.patch -- Jeremy Kister http://jeremy.kister.net./

...3E at 10.9.1.9 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. WARNING[2492]: chan_sip.c:3805 retrans_pkt: Hanging up call CB674A02-B25C11DF-B6D5A08D-652FE73E at 10.9.1.9 - no reply to our critical packet (see doc/sip-retransmit.txt). I have a full sip debug at: http://jeremy.kister.net/tmp/sip_debug/ast.txt A running config of the c1760 is at http://jeremy.kister.net/tmp/sip_debug/c1760.txt Important parts of sip.conf are at http://jeremy.kister.net/tmp/sip_debug/most_of_sip.conf I have verified the same behavior with asterisk 1.6.1.12. Ideas? -- Jeremy Kister htt...

...=> s,n,Set(CONFNO=99${NUM}) exten => s,n,GotoIf(${DB_EXISTS(confbridge:${CONFNO})}?1) exten => s,n,Set(DB(confbridge/${CONFNO})=1) [foo] exten => s,1,Macro(confbridge-setup) exten => s,n,ConfBridge(${CONFNO}) exten => s,n,NoOp( ${DB_DELETE(confbridge/${CONFNO})} ) -- Jeremy Kister http://jeremy.kister.net./

After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? -- Jeremy Kister http://jeremy.kister.net./

...#39; /etc/asterisk/logger.conf [general] [logfiles] console => notice,warning,error,dtmf messages => notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? -- Jeremy Kister http://jeremy.kister.net./

...my asterisk box on the same subnet as a cisco 1760 (vgw1). a few times per day, Asterisk thinks vgw1 is dead (by qualify/options). A 'sip reload' always fixes the problem. i left 'sip set debug peer vgw1' on the console. but i dont see what's causing the issue.. http://kister.net/tmp/ast-sip.conf http://kister.net/tmp/ast-console.txt can anyone spot the issue? -- Jeremy Kister http://jeremy.kister.net./

Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone call to a specific number and make an announcement? I imagine the first part is the big question. joe a.

...ven though I don't see asterisk taking more than 3% cpu. Is this behavior indicative of a timing problem? loading res_timing_pthread.so makes things horribly worse. i don't believe any other software timer is available for Solaris/sparc, right ? other thoughts ? Thanks, -- Jeremy Kister http://jeremy.kister.net./

...at I had set up for monitoring on mantis going to be automatically monitored in jira ? similarly, are tickets that I reported in mantis going to show as me being the reporter in jira? or are the tickets going to stay in mantis until they are resolved and never make it into jira ? -- Jeremy Kister http://jeremy.kister.net./

...xited non-zero on 'SIP/143-00000043' however, my daemon listening on port 4573 never sees activity. so i set up a super-simple server* on port 4573 and saw that Asterisk is not attempting the connection. can someone replicate this behavior ? Or is this just my config ? * http://jeremy.kister.net/code/asterisk/simple_agid.pl -- Jeremy Kister http://jeremy.kister.net./

...ip:asterisk at 10.0.83.61>;tag=as7444eb08 To: <sip:10.0.138.226>;tag=metaswitch+1+0+e288612a Call-ID: 20afd7e40fb31362355eae245dae1fd6 at 10.0.83.61:5060 CSeq: 102 OPTIONS Server: DC-SIP/2.0 Organization: Content-Length: 0 -- Jeremy Kister http://jeremy.kister.net./

...got a Cisco 1760V with a VIC-2FXO-M1/VIC-4FXS and 5 Cisco sip phones. When I make a call from one of the FXS ports on the 1760, the call goes into asterisk's default context instead of where i think i'm directing it. Can someone tell me what I have misconfigured? 1760 config: http://kister.net/tmp/vgw1-confg extensions.conf: http://kister.net/tmp/extensions.conf.txt sip.conf: http://kister.net/sip.conf.txt -- Jeremy Kister http://jeremy.kister.net./

...Spawn extension (extensions, 09930267XXX0000, 1) exited non-zero on 'SIP/141-00000076' -- Executing [h at extensions:1] Hangup("SIP/141-00000076", "") in new stack == Spawn extension (extensions, h, 1) exited non-zero on 'SIP/141-00000076' -- Jeremy Kister http://jeremy.kister.net./

...7:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to establish the call anyway the calling side just hears ringing. i have plenty of debug info, but nothing too interesting. anyone else having this problem ? or is it time for bug report ? -- Jeremy Kister http://jeremy.kister.net./

...sk 1.8.15.0. imagining in extensions.conf: exten => 1,1,Dial(SIP/121) exten => 2,1,Dial(SIP/121&SIP/122) When a caller dials extension 2 /and/ I have trustrpid=yes generaterpid=yes sendrpid=yes in sip.conf and I use the pickup exten, the caller is disconnected. see: http://jeremy.kister.net/tmp/ast/group-with-rpid if i set the rpid generate/send = no for the cisco peer, the user is connected. see: http://jeremy.kister.net/tmp/ast/group-without-rpid calls to exten 1 work regardless of rpid settings. i have replication configs at http://jeremy.kister.net/tmp/ast/ Can someone h...

just thought this was cute enough to pass along, https://www.youtube.com/watch?feature=player_detailpage&v=GLwct15X_3g#t=135 -- Jeremy Kister http://jeremy.kister.net./

...1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by mnicholson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3 Thank you for your continued support of Asterisk!

...1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by mnicholson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3 Thank you for your continued support of Asterisk!

Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem. From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s In logs I see WARNING[8213] chan_sip.c: Retransmission timeout reached on