similar to: 1.6.2 No "soft hangup"?

On Tue, Apr 20, 2010 at 10:49 AM, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Tue, 20 Apr 2010, Tilghman Lesher wrote: > >> On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote: >>> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI> >>> prompt, and found references on using the command "soft hangup >>>

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit

I'm having trouble with both my Polycom IP600 and IP500 disconnecting calls to the PSTN after about 1 hour. The below log is of a phone call that lasted 1hr 39mins which is my record so far. I cannot figure out what is causing the call to terminate and I am hoping somone on this list can help me. In this example both the phone and the asterisk server have public IP addresses so NAT shoul not

Hi, What does this line of output mean? Bridge stops because we're zombie or need a soft hangup: I'm seeing this sometimes... I've looked in channel.c, but the code is not much more revealing than the debug line... -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45

Dear all, Anyone can point me how to soft hangup all channels using single command ? I am using Asterisk 1.4.15. thanks Salaque

How can I "force" soft hangup (if that makes sense)? "show channels" reveals a stale sip channel. It's of an analog phone behind a Grandstream ATA which was communicating with another SIP softphone. The latter crashed. A soft hangup of the softphone seems to have worked but it doesn't for the analog/ATA phone. "show hints" also shows that it's InUse. But

Hi,

Hello List, our setup: Callcenter IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular providers on the xircom analog port, ~60 agents Debian 5.0.1 (Lenny) Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue segfault fix Zaptel 1.4.11 Debian Package My Problem is I have two channels (Zap/9-1 and Zap/6-1) which have a duration of over 4 hours. I am

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

How can I monitor channel that "hangup"? I'm using asterisk 1.8.15.1 and there are many times that nobody is using the line but when I run: asterisk -rx "core show channels" it show: Channel Location State Application(Data) SIP/pstn-4444-000000 (None) Up AppDial((Outgoing Line)) SIP/pstn-9998-000000

Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u" "E2U+SIP" "!^\\+1866(.*)$!sip:1866\\1 at tollfree.sip-happens.com!" . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u"

Hi all, I'm writing my own FastAGI server. I noticed when I send agi Hangup command with and without the channel, asterisk does not hangup the channel and waits for the AGI server to close the connection. Is this how it's supposed to be? Thanks - Abeed -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

Dear R users, I want to remove zero's or NA values after this model. year value1 value2 1854 0 12 1855 0 13 1866 12 16 1877 11 24 year value1 value2 1 12 12 2 11 13 3 16 4 24 Thank you! -- --- Catalin-Constantin ROIBU Forestry engineer, PhD Forestry Faculty of Suceava Str. Universitatii no. 13, Suceava, 720229, Romania office phone +4 0230 52 29 78, ext. 531 mobile phone +4 0745 53

https://bugzilla.mindrot.org/show_bug.cgi?id=1866 Summary: ssh-keyscan should read .ssh/config Product: Portable OpenSSH Version: 5.8p1 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P2 Component: ssh AssignedTo: unassigned-bugs at mindrot.org ReportedBy:

I have a samba 3.023c server with winbind joined to a windows 2003 AD domain. The issue I'm having is from the windows computers, I can't connect to shares on the samba server using the administrator account. It works just fine with normal domain users. When I try to connect with the admin account, I the smbd process that forked to handle the request core dumps. The same thing

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context

Hello, Asterisk party, If block the call before dialing (Hangup()), CDR's don't write to MySQL or CSV. Otherwise, if Hangup() is after Dial(), CDRs write normally. Here is the dialplan: ; we skipped dial, because the number is "blocked" exten => _X.,n(Finish),Hangup() exten => h,1,NoOP("hangup") exten => h,2,ResetCDR(w) exten => h,n,NoCDR() exten =>

Good day, All, Is there any way to maintain the number of decimal places in the type of situation below? I would like to maintain the number of decimal places in 0.667, despite the fact that its column-mates have a fourth decimal place. Thank you for your time. Jim dat.txt contents: MARKER ALLELES FREQ1 RSQR EFFECT2 STDERR CHISQ PVALUE rs6599753 C,T

I have a hosted server. I have TWO different locations what have phones. Chicago and Indiana If I send audio direct from server to Chicago I hear it - same with indiana. But if indiana calls chicago - NO AUDIO. I see this in the CLI -- Channel SIP/63009-00000013 joined 'simple_bridge' basic-bridge <475050e7-9d99-43f0-a9bf-7aa581a97fd9> -- Channel SIP/63000-00000012 joined