similar to: Queues and Timeouts.

I have been seeing this problem for a long time and it occurs in 1.4.0b2 (as well as 1.2.0-1.2.12.1). If the internet connection is lost and I have SIP services that require me to register, any SIP devices attached to the system stop working. I have an IAX phone connected to one of my servers that I've been having this problem with which will work fine (and filover to the PSTN) the

Hi, I've just built my first asterisk server. Current information: OS Version: Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10 06:50:22 EDT 2007 i686 i686 i386 GNU/Linux Asterisk Build: Asterisk 1.4.11 Asterisk GUI-version Revision: 1479 $ Server Date & TimeZone: Thu Sep 6 02:37:11 EST 2007 I've used the Asterisk GUI for setup with two IP

Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/<number>@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc).

I have SIP handsets 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

Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the

I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13. Is anyone else getting the following error in the asterisk console: [Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short every couple of seconds when a handset is in a call? I didn't notice this happening when I was using an older GXP2000 with the same firmware (doesn't mean that it

I know I must be being daft, but is there a way to set which context the queuing system uses when it dials the operators/agents? By default it appears to use the default context. I've looked through voip-info.org and can't find anything, someone please put me out of my misery.

Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been

Hi, I have a question about the definition of agents. The agents.conf file looks like this: [general] persistentagents=yes [agents] maxlogintries=5 ackcall=no wrapuptime=500 musiconhold => default group = 1 agent => 1311,1311,Tom agent => 1531,1531,Tim and here is the queues.conf: [general] persistentmembers = yes [queue1] musiconhold = default strategy = rrmemory servicelevel = 60

Hi All I have a problem with queues that I have been trying to solve for many months - the customer has now picked back up onto this and wanting a solution - any guidance, ideas or solutions welcome. This is the situation :- We have a number of agents in a ringall group, a call joins the queue and all handsets ring - great. An agent answers and handles the call - great. The call

Hello, I just upgraded some of my el6xen boxes to latest CentOS 6.7, Xen 4.4.3 and dom0 3.18.17 rpms, and noticed these new problems: 1) virt-manager fails to start with an error: "D-Bus library appears to be incorrectly set up; failed to read machine uuid: Failed to open "/var/lib/dbus/machine-id": No such file or directory" And indeed.. there was no such file. Running

Hello, I have a partially working Realtime Queue using unixODBC and Firebird. I followed the steps on asterisk-guru and the voip-info asterisk message boards but have been having no success. I say that my queue is partially working because callers can join the queue. However the calls do not go to the agents even if the agent is not tied up and is available for calls. So the caller is stuck

Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (haven't tried SIP), there is no ring. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/a4f29a97/attachment.htm

Hello, My question is this, I have a service queue that members follow the service interval (wrapuptime = 30). However, sometimes these members need to call the customer back, thus making an active call. Occurs when this member disconnects the call shortly following section in the queue already sends a new call for him not understanding the range (wrapuptime = 30). My question is if I can

Hi, I'm trying to use Round Robin Memory with my queues. It seems to work fine... that being I call in.. first time agent 1 will get a call, second time agent 2, and so on. However, it seems if a period of time has gone by since agent 1 got a call (I don't know how much but say 10 minutes or so) when another call comes in.. agent 1 gets the call again. Can anyone confirm this? Is

Hi all, I have configured the queue below, but when I go into the queue, asterisk does not announce hold time: [support] musiconhold=>default strategy=ringall context=check_time timeout=20 wrapuptime=1 maxlen=3 announce-frequency=5 announce-holdtime=yes joinempty=no leavewhenempty=yes reportholdtime=yes I've tried changing timeout, announce-frequency, but still the same; queue works

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

Hi, I have a virtualization host running KVM and libvirt 0.9.12 (on Debian unstable). I would like to connect with virt-viewer 0.5.4 from a different host to a domain that has Display Spice configured as graphics interface. I can connect to the KVM host using virsh --connect=qemu+ssh://mh at fan.local/system and start m KVM domain from there, so I guess that authentication is just fine.

- perform dns lookup on hostname, - randomize local tunnel port - simple autobuild script - bump rpm spec version --- autobuild.sh | 41 +++++++++++++++++++++++++++++++++++++++++ main.c | 14 +++++--------- ovirt-viewer.spec | 9 ++++++++- tunnel.c | 29 ++++++++++++++++++++--------- 4 files changed, 74 insertions(+), 19 deletions(-) create mode 100755

- perform dns lookup on hostname, - randomize local tunnel port - bump rpm spec version updated to include danpb's feedback regarding getaddrinfo --- main.c | 36 ++++++++++++++++++------------------ ovirt-viewer.spec | 9 ++++++++- tunnel.c | 29 ++++++++++++++++++++--------- 3 files changed, 46 insertions(+), 28 deletions(-) diff --git a/main.c b/main.c index