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...localprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Dial("Zap/1-1", "ZAP/g0/817XXXXXX") in new stack -- Called g0/817XXXXXX -- Hungup 'Zap/4-1' -- Executing Macro("SIP/302-ffef", "dialout-trunk|1|817XXXXXX") in new stack -- Executing Macro("SIP/302-ffef", "record-on|302") in new stack -- Executing AGI("SIP/302-ffef", "set-timestamp.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin...

...n't changed any firewall rules during the changeover, and is forwarding unique ports for each phone. Furthermore, the SIP configuration for these phones send a qualification message every 60 seconds to keep any NAT translation alive. Anyway, here's the dialplan for the IVR (only extensions 302 and 303 are included for brevity): [ivr-XXXXXX] exten => s,1,Answer exten => s,n,Playback(silence/1) exten => s,n,Background(XXXXXX/greeting) exten => s,n,WaitExten(4) exten => 302,1,GotoIf(${DB_EXISTS(CFIM/302)}?dialfw:dial) exten => 302,n(dialfw),Set(extension=${DB(CFIM/302)})...

...f the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show hints with no calls: -= Registered Asterisk Dial Plan Hints =- _3XX at hints : Custom:${EXTEN} State:Idle Watchers 0 302 at hints : Custom:302 State:Idle Watchers 2 303 at hints : Custom:303 State:Idle Watchers 2 301 at hints : Custom:301 State:Idle Watchers 2 And wit...

Hi list, I got a question: When I try to ChanSpy a SIP channel I only listen one channel, for example, I call from 302 extension and I have two active channels: SIP/r1-voip-1b7b (None) Up Bridged Call(SIP/302-f1f1) SIP/302-f1f1 09143213452@prueba-sip- Up Dial(SIP/09143213452@r1-voip|4 When I try to spy this call from another extension: 1.SIP/301-fecc 4302@prueba-si...

Dear R-users, I found a strange problem working with products of two matrices, say: a <- A[i, ] ; crossprod(a) where i is a set of integers selecting rows. When i is empty the result is in a sense random. After some trials the right answer (a matrix of zeros) appears. --------------- Illustration -------------------- R : Copyright 2003, The R Development Core Team Version 1.8.0

...ared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not work. This error would occur: Spawn extension (companyname-default, 304, 1) exited non-zero on 'SIP/302-0824f618' In the case above, phone extension 304 and 302 were talking, and 304 pressed 'hold'. 302 gets dropped, as indicated above. I enabled sip debuggint (fully below) and notice that I get: SIP/2.0 488 Not Acceptable Here Warning: 399 SDP Not Acceptable Lots of info about the 4...

...m unclear how to acheive this given that there is a nested structure- the value varies between wavelengths within individuals. Many thanks Nevil Amos data is structured thus: > ANWC_NO Wavelength Repeat value > 1 00239 300 r1 0.079501 > 2 00239 302 r1 0.084113 > 3 00239 304 r1 0.087697 > 202 11157 300 r1 0.008449 > 203 11157 302 r1 0.009489 > 204 11157 304 r1 0.010142 > 403 11158 300 r1 0.026999 > 404 11158 302 r1 0....

I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8. Setting forwarding for "Always" works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location. Setting forwarding to "No Answer" means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a 302 SIP message issued. I can't find anything in the phone s...

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in publ...

...R users, Wonder if somebody could give me help on how to reshape this type of data: ----------------------------------------------------------------------------------------------------------------------- Date:10.09.19 Time:21:39:05 Lat:N62.37.18 Long:E018.07.32 0000-0010 | 28| 28 0010-0020| 302| 302 0020-0030| 42| 42 0030-0040| 2| 2 0040-0050| 1| 1 0060-0070| 1| 1 _ Date:10.09.19 Time:21:44:52 Lat:N62.38.00 Long:E018.09.07 0000-0010| 32| 32 0010-0020| 334| 334 0020-0030| 27| 27 0030-0040| 2| 2 0070-0080| 1| 1 0080-0090| 1| 1 0090-0100|...

...ndsets which allow the user to forward a call to another number after a specified interval of ringing time. On the SwissVoice this is refered to as CFNR (Call Forward on No Response). What actually happens is that after a specified period of time (default 15 seconds), the handset sends back a "302 Moved Temporarily" response to Asterisk. The problem is that when Asterisk receives the 302 message, it doesn't reset the ringing timer in the Dial command. Let's say I've issued a Dial command such as: exten => _34XX,1,Dial(SIP/fred|20) exten => _34XX,n,Voicemail(fr...

I have some Cisco 7905 phones with the SIP load 1.02.00(040406A). When the phone is off-hook but no call has been placed, or when the Do Not Disturb is activated, the phone returns a 302 "Moved Temporarily" message back to asterisk as follows: ----------- -- Executing Dial("SIP/5060-0811bb00", "SIP/9871234|20|Ttr") in new stack -- Called 9871234 -- Got SIP response 302 "Moved Temporarily" back from 24.xx.xxx.6 -- Now forwarding SIP/5060-081...

Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get: Got SIP response 302 "Moved Temporarily" When forwarding the call to other SIP server. This is a "bug": http://lists.digium.com/pipermail/asterisk-users/2004-May/045774.html --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

Hello, I have a question . dose asterisk supports "302 Redirecting..." ? I have SIP Server "Not Asterisk" and my Asterisk is registering as a client for this device . when i try to call another client registered to the same SIP server i got Busy Tone and here is the asterisk CLI output ----------------- -- Got SIP response 302 &quo...

This testscript creates reflinks to files on different subvolumes, overwrites the original files and reflinks, and moves reflinked files between subvolumes. Originally submitted as testcase 302, changes are made based on comments from Eric: http://oss.sgi.com/archives/xfs/2013-03/msg00231.html Two new common/rc functions used in this script (_require_cp_reflink and _verify_reflink) have been submitted recently: http://oss.sgi.com/archives/xfs/2013-05/msg00745.html Thanks to Eric Sandee...

...hich obviously leads to a failure. Any comments, do I have a configuration issue on my side that I missed? Cheers, Philipp -- Executing Dial("SIP/philipp-bd5f", "SIP/992365264680@nikotel- out|90") in new stack -- Called 99xxxxxxxxxx@nikotel-out -- Got SIP response 302 "Moved Temporarily "new sip:joesmith@63.214.186.6"" back from 63.214.186.6 -- Now forwarding SIP/philipp-bd5f to 'joesmith@from-sip' (thanks to SIP/nikotel-out-c286) May 7 14:20:54 NOTICE[18450]: chan_local.c:362 local_alloc: No such extension/context from-sip@jo...

I''m having trouble catching a 302 with link_to_remote. The redirect is issued if the user is not logged in with the before filter. In the lighty log i see the 302 scroll by and then it renders the login screen with a 200. That 200, I can catch and issue a redirect (just for testing). Any ideas on where I go wrong? <%= link...

...mple both the phone and the asterisk server have public IP addresses so NAT shoul not be an issue whatsoever. Any ideas or help would be greatly appreciated. -DevilFish Asterisk Version: Asterisk CVS-v1-0-01/13/05 Call Start: Jan 15 12:46:59 VERBOSE[4290]: -- Executing SetGroup("SIP/302-928e", "302") in new stack Jan 15 12:46:59 VERBOSE[4290]: -- Executing Dial("SIP/302-928e", "SIP/12699264242@sip_proxy-out|30") in new stack Jan 15 12:46:59 VERBOSE[4290]: -- SIP/sip_proxy-out-f201 is making progress passing it to SIP/302-928e Jan 15 12:47...

...ension 7002 is a SIP hard phone currently configured to forward incoming calls to extension 7003, when a call is unanswered within a 10s time frame - when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20) statement and no one answers, then : - after 10s, Asterisk receives "SIP 302 Moved temporarily" message and enters its dialplan to call 7003, as required, - 10s later (or 20s from the very start), call from 7001 to 7003 is cut and the next statement after Dial(SIP/7002,20) is run. The behaviour I would ideally implement is : - whenever a "SIP 302 Moved temporaril...

...ke a call. Everything is identical right up to the point where the HT503 gets a congestion instruction from the Asterisk server. Here is the debug output just at the point where it happens. -- AGI Script dialparties.agi completed, returning 0 -- Executing [s at macro-dial:7] Dial("SIP/302-08221a38", "SIP/301||tr") in new stack -- Called 301 Home*CLI> <--- Transmitting (NAT) to 192.168.0.100:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.100:5060 ;branch=z9hG4bK443855200;received=192.168.0.100;rport=5060 From: <sip:302 at 192.168.0.1>;tag=1...