search for: vgw1

I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can

When I place an outbound call from asterisk 1.6.1.12 to a FXO port on my Cisco 1760V 12.4, the channel changes - seemingly incrementing: e.g., in the first call, below, the channel name is "SIP/vgw1-00000075" -- the second call (on the same FXO port after a soft hangup on the CLI) is "SIP/vgw1-00000077" How can I extract this information in the dialplan so that I can use the SoftHangup app in asterisk to disrupt an existing call ? pbx1*CLI> soft hangup SIP/vgw1-00000075...

I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active channels 1 active call 194 calls processed pbx1*CLI> in my dialplan, i have: exten => s,1,Set(CHAN=${SHELL(asterisk -rx "core show channe...

using asterisk 11.6.0-rc1 i just converted my "nat=yes" to "nat=auto_force_rport,auto_comedia" I have 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-con...

...evening. All my custom modules (including swift <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. Any...

...60V 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./