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Displaying 20 results from an estimated 30000 matches similar to: "No subject"

2014 Mar 21
3
Need more meetme users -- hitting some limit
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call
2011 Sep 02
0
No subject
is typing his number, though there is a 15 seconds timeout, and even if = I type the number very fast it still may happen to me.<o:p></o:p></p><p = class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>It has = been my casual observation that the speed at which I enter digits on my = phone is unrelated to the speed at which my
2009 Aug 29
0
asterisk-users Digest, Vol 61, Issue 84
On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'
2005 Aug 23
2
SIP powercycle not hanging up
I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"
2007 Jul 12
0
No subject
* The exit behavior of the AGI applications has changed. Previously, when a connection to an AGI server failed, the application would cause the channel to immediately stop dialplan execution and hangup. Now, the only time that the AGI applications will cause the channel to stop dialplan execution is when the channel itself requests hangup. The AGI applications now set an AGISTATUS
2011 Apr 12
0
No subject
lob1 [lob] Show IPA verb, lobbed, lob=C2=B7bing, noun =E2=80=93verb (used with object) 1. Tennis . to hit (a ball) in a high arc to the back of the opponent's=20 court. 2. to fire (a missile, as a shell) in a high trajectory so that it drops ont= o=20 a target. 3. Cricket . to bowl (the ball) with a slow underhand motion. Who do suggest we should be lobbing our fax machines at? --=20 Thanks
2011 Apr 12
0
No subject
system() to execute this script since it is (obviously) not really an AGI. I'm guessing that system() would be slightly more efficient than agi(). Both require a process creation, but agi() requires (slightly) more Asterisk resources in setting up the AGI environment. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards
2009 Jul 20
0
No subject
echo test | mail -s test thomas.perron at gmail.com If that doesn't work and you don't get any useful clues from the command output, start digging where your syslogd logs messages. Next, from a shell command line, try echo test | mail -s test 5555551212 at txt.att.net Note that this recipient is specific to this carrier. If that works, it should work in Asterisk assuming you
2009 Jul 20
0
No subject
depends on where you are in the world. Generally speaking, somewhere around 8 to 12. There are many advantages to PRI over POTS: ) "Instantaneous" call setup. ) Higher reliability. ) Less cabling behind the server. ) Better support from your provider. ) More features. ) Better audio quality. IMO, you should always use a PRI unless you can't afford it. -- Thanks in advance,
2009 Jul 20
0
No subject
channels'," "asterisk -r -x 'sip show channels'," or "asterisk -r -x 'zap show channels'." Note that these are the "1.2" commands. You are probably using a more current version. If the count exceeds your threshold, create a call file to call you and play an appropriate message. -- Thanks in advance,
2011 Sep 02
0
No subject
core show function SIP<TAB> I use: set(PEERIP=${SIPCHANINFO(peerip)}) in one of my dialplans. For AGI, whatever function in your library that executes 'GET FULL VARIABLE' should do the trick. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline
2007 Apr 09
1
Re: asterisk-users Digest, Vol 33, Issue 35
We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >
2015 Apr 27
2
adding area code
On Mon, 27 Apr 2015, Bryant Zimmerman wrote: > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
2015 Jun 26
2
Asterisk 13 logging to two places
On Fri, 26 Jun 2015, Dale Noll wrote: > I added a filter to the /etc/rsyslog.conf file > > :syslogtag, contains, "asterisk" stop > > Syslog is still receiving the messages, but is discarding them. Nice to learn a new (to me) feature of rsyslog. What does 'logger show channels' show? -- Thanks in advance,
2010 Jan 03
0
asterisk-users Digest, Vol 66, Issue 4
"asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> wrote: Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to
2009 May 06
2
Where are 2 letter language values defined?
I've googled for way too long, where are the 2 letter language values defined? I know: en = English es = Spanish fr = French but what about Croatian, Russian, Serbian, Vulcan, etc? Is there a list documented for Asterisk or is it "just use the 2 letter country code Internet TLD?" Thanks in advance, ------------------------------------------------------------------------ Steve
2011 May 17
1
OT, free software for SIP ladder diagrams?
I was debugging a turnup with Global Crossing the other day and they presented me with a web page that displayed a 'ladder diagram' of a call including a ton of detail all neatly organized in tabs and links so you could drill down to any level of detail needed. The copyright notice says 'Copyright? 2008 Empirix.' Is there any free software available to analyze a pcap or
2008 Mar 29
2
Finding iaxy's (iaxies?)
According to http://kb.digium.com/entry/12/ The Iaxy will respond to pings on port 9999. You can ping your broadcast IP on your network and listen with tcpdump on your network on port 9999 which will show the Iaxy responding and what IP address it is coming from. Ex. ping 192.168.1.255 tcpdump -i eth0 udp port 9999" Before I get my karma whacked again, does this work for
2014 Nov 18
2
AGI and AMI in PHP -- What's current?
I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? -- Thanks in advance,
2008 Nov 13
1
Asterisk and Zaptel version numbers -- how close is close enough?
I'm doing a new install for an old customer. The customer is running a custom version of Asterisk based on version 1.2.7.1. It works for them -- aside from a memory leak requiring a restart once every couple of months... I think the "corresponding" version of Zaptel is 1.2.5, but I'd like to run a bit more modern like Zaptel 1.2.27. Am I just asking for trouble? Thanks in