similar to: go on in current context after destination channels hung up ?

We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it.

Asterisk Project Security Advisory - AST-2014-014 Product Asterisk Summary High call load may result in hung channels in ConfBridge. Nature of Advisory Denial of Service Susceptibility Remote

Asterisk Project Security Advisory - AST-2014-014 Product Asterisk Summary High call load may result in hung channels in ConfBridge. Nature of Advisory Denial of Service Susceptibility Remote

I have recently posted a message regarding hung SIP channels when using the Monitor() command. Well, I was wrong.   The channel hanging wasn't caused by the Monitor command. They also hang when they aren't monitored. The cause seems to be our PSTN gateway provider. When for some reason their PSTN gateway crashes or reboots (OK, this should happen, but anyway...) and RTP/SIP data stops

Could some out there with a PRI check and see if this problem shows up on your system? The test is to dial a number routed to * via a PRI where there is no match in the dial plan for the dialed number. Asterisk will reject the call, but "show zap channels" still shows the channel assigned to the number that was dialed under the extensions column. The channel WILL answer another call,

Can someone tell me how to clear hung SIP channels in asterisk without restarting? Currently I have 62 channels and only show 10 in use.. this is some of the sip show channels output.. xxx.xxx.xxx.xxx 00xxx24xxx 04240xxxxxx 00103/00001 UNKN (d) ?How can I remove these? from * without rebooting? ? .o-------------------------------------------------------o. Brian Fertig Network

I'm having many troubles with x101p (orginal from Digium, wcfxo kernel module) on analog line simulated by a Cisco Router I'm experiency random hung up while zaptel doesn't recognize call progress (Italy signalling) Signalling is simple ignored as if someone hangs up on the other end of the analog line, on the asterisk end is it possible to listen the busy tone as generated from

When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and

/Hello Users, Good Morning, In Conferemcing How to Disconnect the phone while in between the Conference ..... When *I press the ' # ' key for Disconnecting the Conference.......... Below the Following to shows some Warning, ( in Red Color ) from-sip en *CLI> -- Executing Playback("SIP/9002-08f9feb8", "conf-hasentered") in new stack

Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten

don't quote me on this but I believe the earlier assumtion is correction. I think you need to have RTP going bothways otherwise the call will disconnect. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Shaw Sent: Friday, August 13, 2004 12:40 PM To: asterisk-users@lists.digium.com Subject: Re:

F.Y.I >>> "Tian, Kevin" <kevin.tian@intel.com> 08.11.27. 11:50 >>>Sorry for a typo. I did mean domU instead of dom0. :-) The point here is that time_resume will sync to new system time and wall clock at restore, and thus pv guest should be able to continue... Xen system time is not wallclock time which just counts up from power up. As Keir points out, only its

Hi, I met a problem when using libguestfs-1.28.1 (centos 7.1) on a virtual machine. Execute virt-inspector command to get OS information and the command hangs . ....................... Launching appliance, timeout set to 600 seconds libguestfs: launch: program=virt-inspcetor libguestfs: launch: version=1.28.1rhel=7,release=1.18.el7,libvirt libguestfs: launch: backend

Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue

I have a PAP2 with 2 phone ports. When I call them everything works fine until I hung up the call. There is about 30-40 seconds until I can call to that extension again. Before that it gives me busy messages. Extension config: exten => 199,1,Dial(SIP/199,30) exten => 199,102,Hangup Any suggestions? Thanks

Hi all, how can I catch the event generated when a parked call is hung up? In my dialplan, when arrives a call to a specific number, Asterisk parks the call and announces the parking slot to a number. But if the user hangs up the parked call, I don't know how to catch the event, from dialplan, that removes the call from the parking slot. I want to know if there is a method for do this.

I have a problem where my Asterisk server stops answering new TCP requests and begins to use 99.9% of the CPU on my box. The server is a 64bit Xeon with 2GB of ram. I haven't been able to recreate the problem but it occurs sometimes when there is a call coming from my provider (via IAX) to a customer (via IAX). The customer and the provider systems are running asterisk also. The call will

The FXO lines on my TDM400 are connected to PSTN lines in Israel. As far as I know, the lines around here have disconnect supervision (I've seen some other Israelis on this list, anyone know for sure?), because it's worked on Dialogic cards, which reported hangup, not busy detect (while when I connect a Dialogic card to a PBX, I have to measure the busy signal's frequency/cadence or

When it accesses a real floppy disk, it seems to do Hung when the boot disk of MS-DOS (Win95/98/Me) was able to be started by using ISOLinux. example) Not inserted Floppydisk A>dir b: error message A/R/F ? A> inserted Floppydisk A>dir b: ....hung.... It seems to do Hung in Real PC though the phenomenon like place tested with VMware was not Hung. There was no problem in a usual

I was playing around with conferencing tonight. I was able to place a bunch of SIP phones and a couple of my Zap FXS phones into a conference. So I thought, "Let's see what it's like when people come in from outside." So I called a friend and had him call in on one of my Zap channels, WHICH IS CONNECTED TO MY POTS LINE THAT DOESN'T DO DISCONNECT SUPERVISION. When he