I did a cvs update this afternoon and since then asterisk doesn't seem to clean up the channels after they hangup. This has been working perfectly for quite some time previously... I do a show channels and it shows the channels still up. The only way out is to kill and restart asterisk.... I am frantically trying to would out how to get a non-current CVS copy of the source and get it back up and running :( If any debug output is needed, let me know what you would like. I am using SIP (ata186), ISDN (i4l), X101P and the USB FXS... Regards, Adam
On Mon, Apr 14, 2003 at 04:14:19PM +1000, Adam Goryachev wrote:> I did a cvs update this afternoon and since then asterisk doesn't seem to > clean up the channels after they hangup. This has been working perfectly for > quite some time previously... > > I do a show channels and it shows the channels still up. > > The only way out is to kill and restart asterisk.... > > I am frantically trying to would out how to get a non-current CVS copy of > the source and get it back up and running :(cvs update -D yesterday modulename from the man page: Examples of valid date specifications include: 1 month ago 2 hours ago 400000 seconds ago last year last Monday yesterday a fortnight ago 3/31/92 10:00:07 PST January 23, 1987 10:05pm 22:00 GMT> If any debug output is needed, let me know what you would like. I am using > SIP (ata186), ISDN (i4l), X101P and the USB FXS... > > Regards, > Adam > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- Scott Lambert KC5MLE Unix SysAdmin lambert@lambertfam.org
On Sunday, Apr 13, 2003, at 23:14 America/Los_Angeles, Adam Goryachev wrote:> The only way out is to kill and restart asterisk....I get that too. Using CVS as of Friday evening IIRC. -- http://www.askbjoernhansen.com/
Seems the Pingtel Expressa software phone does inband DTMF. Or at least the voicemail only works with that option active :) Maybe this line can be added to the sample sip.conf to avoid confusion. Keep up the good work ! Greetings, Tjardick
Hello
Thanks everyone for the tips so far. I've got the Cisco 7960's updated
and working much better. Inbound works and phone to phone works.
As far as I can tell outbound calls are not getting processed properly.
Here's the error.
Called g1/4261600
-- Channel 1, span 1 got hangup
== No one is available to answer at this time
-- Hungup 'Zap/1-1'
Here's my zapata.conf
[channels]
;
; Default language
context=default
switchtype=national
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;
; PRI config
switchtype = national
channel => 1-23
This is a snippet of extensions.conf, the rest of the file is standard.
; lines just contains extensions for local users. It is included from
; both "default" and "localusers".
[lines]
exten => 3155796503,1,Macro(stdexten,6503,SIP/6503)
exten => 3155796521,1,Macro(stdexten,6521,SIP/6521)
exten => 6503,1,Macro(stdexten,6503,SIP/6503)
exten => 6521,1,Macro(stdexten,6521,SIP/6521)
exten => 6565,1,Macro(stdexten,6565,SIP/6565)
exten => 666,1,Macro(stdexten,666,SIP/666)
; add additional extensions here, format for NUMBER would be:
; exten => NUMBER,1,Macro(stdexten,NUMBER,SIP/NUMBER)
exten => 8888,1,VoicemailMain
exten => 8888,2,Hangup
exten => 8889,1,Answer
exten => 8889,2,Echo
exten => 8889,3,Hangup
; localusers is the context that users on local phones will come into.
They
; can either dial local extensions or outside numbers.
[local]
include => lines
include => parkedcalls
include => international
exten => i,1,Answer
exten => i,2,Playback(invalid) ; "That's not valid, try
again"
exten => i,3,Hangup
exten => t,1,Hangup
; default is the context outside callers will come into. The "lines"
; context allows callers to connect to our local users. The "s"
extension
; makes an announcement when callers dial in.
[default]
include => lines
include => parkedcalls
include => international
; exten => 3155796503,1,Goto,10
; exten => 3155796521,1,Goto,10
exten => i,10,Answer
exten => i,20,Playback(invalid) ; "That's not valid, try
again"
exten => i,30,Hangup
exten => t,1,Hangup
Thanks
Ken McKittrick
Network Engineer
USADatanet
Can someone please help me. I am currently HEAD as of about 5 days ago (stable was giving me all sort of problems, upgraded per other users suggestions) on an Intel mainboard using a mix of Cisco 7960/40 SIP and 7910 SCCP. Can someone please explain what the following means? When this happens, I am about 1 minute from Asterisk going downhill. All of the SCCP phones quit, while the SIP phones can do calling to some degree. I get kicked out of any consoles and can't reconnect without restarting asterisk. Mark May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-0000001a' May 7 01:03:13 WARNING[28400] channel.c: Avoided deadlock for 'SCCP/118-0000001a', 10 retries!
hi !
i have the following in my extensions.conf
exten => 2000,1,Wait(60)
exten => 2000,2,Hangup
When i dial '2000' from my phone, I see 'Wait' being called.
After 60 secs,
I also se 'Hangup' being called. If I hangup the phone line before 60
secs
are over ('Wait' command is probably interrupted in this case), asterisk
crashes with segmentation fault.
Due to this problem, my 'campon' feature causes asterisk to crash often.
does anyone have an idea as to what this problem might be ?
tulika
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