similar to: precision of wait dialplan application

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

Hi, I need to create something like this: <?xml version="1.0" encoding="UTF-8"?> <Container> <id>aQlfVHX+qPM</id> <lifetime>2009-09-19T08:14:55Z</lifetime> </Container> The response should contain the next headers: Content-Type=`application/vnd.3gpp+xml`

Hi, I've got a problem with my asterisk set up which has been going on for a while (months). I'm currently running 1.2.7.1 on a gentoo box with the topology below: +-----+ PSTN ---------+ * +------------- Service Provider (wctdm400p) +-+-+-+ IAX | | | | FXS --+ +-- SIP (cisco 7940)

Folks, I've been trying to handle the problem where blocked callerids appear as coming from asterisk <asterisk> on the email notification, and the message envelope simply doesn't say anything (does not actually play the vm-unknown message). So, following the tip provided by several previous posters, I tried putting this in my extensions.conf (the xx's are my DID,

Recently, I got the following error messages in CLI periodically. Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002 handle_request_subscribe: Got SUBSCRIBE for extension XXXXXXXXXXX@from-int from 192.168.0.123, but there is no hint for that extension I have no idea what the error message tell me. I am sure I haven't that account XXXXXXXXXXX in my database and there is no hint extensions in dial

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Below is a copy of my log, zapata.conf & extensions.conf that relate to the ZAP lines. Basically when we dial out it takes on 10-12 seconds before the ZAP line actaully picks up. I'm hoping to find out what the cause is for this as it's causing user grief with extremely long connect times, and I believe it may be causing issues of cross lines (an outgoing call gets mixed with an

Has anyone configured ast_fax (sending faxes via asterisk) with sendmail? The creation of rules to trap all numbers <mailto:XXXXXXXXXXX@domain.com> XXXXXXXXXXX@domain.com seems too complicated. Does anyone have setup details to share? (I don't want to switch MTA's). As a workaround, I could launch the app automatically from sendmail using an alias like: fax: "|

Hello. I'm in need of some help to shed some light on a strange problem. The box is running Samba 4.1.7 on FreeBSD 9.2/amd64 and I suspect (though I'm not sure) all troubles started after the upgrade from 4.1.6. The setup is quite simple: no AD, no LDAP, only one server. A command as simple as "smbclient -U user //SERVER/SHARE", if fired up several times in a row on the

All, I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all* of my incoming calls are coming up as FAXes. I had to disable my fax extension because every call to my POTS line was getting redirected to my FAX machine. After removing the FAX extension, if I call my POTS line from my cell phone, I get the following: *CLI> -- Starting simple switch on 'Zap/1-1'

Hi, there: When plot ann predicted results I need to indicate numbers of inputs for each column. V1 V2 V3 V4 1 86.2700 49.9380 30.7630 0.1327 2 89.5127 55.9707 33.7683 0.1186 3 91.1833 58.4670 34.5610 0.1134 matplot(t, pch = 1:4, type = "o", col = rainbow(ncol(t)),xlab = "No. of inputs ", ylab = "Mean of 6 Datasets",

Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxxxxxxxxxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something

Hi, I'm facing an issue with at least two Samba servers in production both showing important number of log entries like: smbd[27355]: PANIC: assert failed at smbd/oplock.c(149): exclusive_oplocks_open>=0 Looking at the source code, this panic is triggered when the number of exclusive oplocks for a given file becomes negative. Once triggered for one smbd process, this process can

Hi, I have a IAX2 trunk between two sites (connected with an high bandwidth link) but sometime/often I get: chan_iax2.c: Auto-congesting call due to slow response and call is dropped (and routed on a PSTN link). In iax.conf, I have: [iax-out] username=iax-in type=peer trunk=yes secret=xxxxxxxxxxx qualify=yes host=xxx.yyy.zzz.32 auth=md5 Any idea? Perpaphs is due to

I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxxxxxxxxxx type=peer username=0406082250 Regards Anders Svensson -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello ! Are there any plans at Digium to include also german voice prompts ? Thanks regards Hans

Hi, environment: new Centos 6 installation out of the box. somebody has an idea what causes this error mail to be sent? ---8<--- Subject: Cron <root at fileserver1> /usr/local/asc/bin/ascservicemonitor 2>&1 >/dev/null Content-Type: text/plain; charset=UTF-8 Auto-Submitted: auto-generated X-Cron-Env: <SHELL=/bin/sh> X-Cron-Env: <HOME=/root> X-Cron-Env:

Hi, Recently we upgraded to Dovecot 2.3.11.3 and configured an example metric like this: metric imap_command { event_name = imap_command_finished group_by = cmd_name tagged_reply_state user remote_ip } And enabled the OpenMetrics listener like this: service stats { inet_listener http { port = 5000 } } While the result is great, I noticed that some metrics are being truncated

I've got a CentOS 6 machine that's slated to go into production providing some web and development-repository services. Part of the environment is gitweb, which works as expected with one glitch: SELinux doesn't allow gitweb.cgi to query sssd to display who owns the repositories. The audit log entries are pretty straightforward, e.g., type=AVC msg=audit(XXXXXXXXXXXX): avc: