similar to: Fwd: Call Wainting dysfunctions

Dear all, Anyone has experience using group and group_count to limit outgoing calls in AGI/CAGI? SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP EXEC Gotoif $[${GROUP_COUNT(OUTBOUND_GROUP@${CALLERIDNUM})} > 1]?BLOCK SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP But it doesn't work as it should. Tried in extensions.conf and it works. Any idea. Thanks Ray

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

Hello users! Recently I read that AgentCallbackLogin is going to be deprecated soon. Wanting to set up a few callback type queues, I set them up as suggested in queues-with-callback-members.txt. I was able to set the queues up completely this way, however, I'm trying to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login status. FOP monitors their status if I call

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

AgentCallBackLogin is going to be deprecated, so I've decided to emulate chan agent using AQM and RQM funcions and Local channel. I use asterisk 1.2.13 and latest 1.2.x. zapata. I used example 2 from http://www.voip-info.org/wiki/view/Agents+without+agent+channel and example from queues-with-callback-members.txt from asterisk 1.4 doc directory. My dialplan is very similar to Digium's

Hi, I need to port OSCAR Cluster and SystemImager softwares to centos8, but I miss a lot of package that seems to be built for centos-8. For example, I cant find docbook-utils and docbook-utls-pdf while I see them here: https://koji.mbox.centos.org/koji/buildinfo?buildID=651 I?ve installed epel-release and elrepo-release and centos-release-stream. Do I miss something or is it a matter of time

Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK except for the DTMF for number 4, because if the caller from mobile dial 1004 or 1014 extensions -which

Package: xen-utils-3.2-1 Version: 3.2.1-2 Severity: grave I tried to start an HVM guest to setup Windows 2003 64 bits entreprise R2, and I spent a long time figuring out that there is no /usr/lib/xen folder by default, and this is where Xen is searching for it's file when using HVM, VNC and all. Doing a simple symlink with ln -s /usr/lib/xen-3.2.1 /usr/lib/xen made it working, if it's not

Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, it?s just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : <vendor> ----SIP--? <Asterisk> ?----IAX2---? <Asterisk with Dhadi channels> Here: -- Executing [966505103150 at from-internal:1]

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Sorry if this hit the list twice.. but I didn't see it come through: Hey guys, Just for archival purposes, I have setup the Intellitouch ITC-3002 (2006) SIP phones for use with asterisk (1.2.7.1). After a few "gotcha's", I was able to do transfer's, moh's, push a button to check voicemail, callerID, etc.. One big gotacha was the dial timeout (by default!) is set to

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

On Thu, 26 Sep 2019 at 05:24, LAHAYE Olivier <olivier.lahaye at cea.fr> wrote: > > Hi, > > I need to port OSCAR Cluster and SystemImager softwares to centos8, but I miss a lot of package that seems to be built for centos-8. > For example, I cant find docbook-utils and docbook-utls-pdf while I see them here: https://koji.mbox.centos.org/koji/buildinfo?buildID=651 > So what

http://dovecot.org/releases/1.1/rc/dovecot-1.1.rc12.tar.gz http://dovecot.org/releases/1.1/rc/dovecot-1.1.rc12.tar.gz.sig Sorry, one more RC because of the mbox+quota bug. v1.1.0 still planned for tomorrow. - mbox: Don't give "Can't find next message offset" warnings when plugin (e.g. quota) accesses the message being saved. - deliver: Settings inside protocol imap {}

http://dovecot.org/releases/1.1/rc/dovecot-1.1.rc12.tar.gz http://dovecot.org/releases/1.1/rc/dovecot-1.1.rc12.tar.gz.sig Sorry, one more RC because of the mbox+quota bug. v1.1.0 still planned for tomorrow. - mbox: Don't give "Can't find next message offset" warnings when plugin (e.g. quota) accesses the message being saved. - deliver: Settings inside protocol imap {}

Hi list! I have this configuration for sending T38 faxes to my T38 fax termination provider: T38modem --> hylafax --> Asterisk-SIP-Extension --> T38 termination provider --> T.30 termination to PSTN We are experiencing 2 problems with this (if you want configuration files, it won't be a problem, just tell me): 1. T38 termination provider receives faxes at 2400 bpps from our

Hi all, Recently, I have upgraded the asterisk as following. asterisk-1.4.13 asterisk-addon-1.4.4 libpri-1.4.1 zaptel-1.4.5.1 Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh After upgrade, the server get segfault randomly and asterisk crash and restart itself. I got 2 core dumps of the segfault. Based on the core dump, we can't figure out the root cause to

I''ve recently installed Debian Lenny witch Xen (xen-utils, xen.tools and xen-linux-system-2.6.26-2-xen-amd64). I want to virtualize some windows servers, but I''m unable to connect to localhost VNC when I start the vm (xm create) to proceed with windows instalation. The VM config is: import os, re arch = os.uname()[4] if re.search(''64'', arch):

thank you for your response below the asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack --

Hi, I've built customs kernels from the SRPMs with the following: rpmbuild -bp --target=i686 /path/to/spec cd /usr/src ln -s /usr/src/redhat/BUILD/kernel-x.y.z/linux-x.y.z linux cd linux make oldconfig make menuconfig make all make modules make modules_install And it works just fine. The problem now is that I think installing that way isn't