similar to: Realtime: Should I say or should I go (now) ?

Displaying 20 results from an estimated 10000 matches similar to: "Realtime: Should I say or should I go (now) ?"

2007 Oct 25
3
Realtime on Asterisk 1.2.24
Does realtime work reliably on Asterisk 1.2.24? Are there any definitive guides, I can only find bits and pieces here and there. Any accurate howtos would be of great help. I am missing func_realtime.so. Where does this file come from? Asterisk or asterisk-addons? I saw in one of the howtos that it is needed. Is it needed for 1.2.X or 1.4.X. Also, what about the switch lines in the
2007 Sep 04
11
stop log/debug messages into /var/log/messages
Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console=> notice,error ;messages => notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential
2007 Dec 17
3
VoIP service providers/PSTN termination points
Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax
2007 Aug 20
1
1.4.4. caller ID not working ?
Hello All, Is CALLERID() setting broken in 1.4.4? My small dialplan : [testclid] exten => _0.,1,Set(CALLERID(all)=Ben Jacob <988077>) exten => _0.,n,Dial(SIP/${EXTEN}) Correct me if I am wrong, Set(CALLERID(all) above supposed to change the display name as above(Ben Jacob) and change the From URI to 988077 at myip?? As of now, only the _display name_ is being replaced, but not the
2008 Jul 01
3
music on hold realtime
Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie
2007 Nov 22
1
common/shared voicemail box
Hello All, I am using ODBC storage for voicemail on my asterisk box. I want to have a common voicemail box for different extensions. I know how to do that, but the question troubling me is how and where do I store the the extension name for which a particular voicemail was left. e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 55555. Now, when someone calls 1000, and leaves a
2009 Feb 25
1
Realtime database function help
Hello Everyone! According to voip-info.org the correcy syntax for the realtime function is: REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read REALTIME(family|fieldmatch|value|field) on write It seems from the syntax that it is only possible to retrieve a full row according to the value of only of column. This translates in SQL language as Select * from family where fieldmath =
2008 Jan 10
1
Asterisk Realtime unixODBC timeout?
How does one get asterisk to timeout realtime request via res_odbc to unixODBC? I've set timeouts as appropriate for freetds (which unixODBC is using.) However, it doesn't seem to work. It takes over 3 minutes to timeout a connection and queries never seem to timeout, so a channel waiting on a query never terminates. I did notice that res_odbc.c never sets a timeout on the query
2008 Mar 08
3
replace astdb with a cluster-capable sql database engine
I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a
2007 Aug 01
2
multiple pbxes, multiple domains, same user ids?
Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk
2007 Nov 28
2
Billing/Call Control engine : AGI scripts/ AstMan API
Hello ppl, Have implemented a really nice Billing engine using AGI scripts. So far it works fine, tho haven't yet put it in the torture cell. The AGI scripts have been written in PHP, using MySQL for the billing and profile information. The major disadvantages I see using AGI scripts : 1. A new process(invocation of PHP scripts) on every new call. 2. MySQL connections on every instance of
2008 Oct 08
1
make func_realtime work like app_realtime (1.6)
Yell at me if you will, but I hate func_realtime - it's not very usable nor is it change-friendly (update your database and your dialplan completely breaks). I'm getting a new 1.6 box built out and working, and wanted to emulate the functionality of APP_realtime somehow, so I started digging around in the func_realtime source - here's what I came up with: For 1.6.0, look at line 86
2007 Sep 18
4
Linux limits
Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in
2008 Sep 08
2
Pointers to replace astdb
Hi listers, We want to implement one call center with asterisk. The idea is it should be scalable, with openser as an dispatcher and bunch of asterisk servers to do ACD, Queues, Agents things... Easy to say :( Look closely to the current asterisk, we do see some problem: - SIP registrations was stored in astdb. - And queue members also was stored in astdb. - ... asterisk was built as
2009 Mar 21
2
1.6.2 beta 1 crash
Hi, I'm starting testing 1.6.2 beta. CentOs 5.2 I found my first crash, first I have [Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql: Attempted to update column 'useragent' in table 'sip', but column does not exist! [Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL RealTime: Updating on column 'lastms', but
2018 Dec 06
3
how to use a database
On 12/05/2018 05:00 PM, Antony Stone wrote: > On Wednesday 05 December 2018 at 15:31:38, hw wrote: >> I don't see a table for that. > > You need to create that for yourself. > > Asterisk can write to database tables, but doesn't help you set them up, for > reasons I can't comment on. How do I know what the schema needs to be? Does anybody have a scheme for
2011 May 11
4
concurrent call tracking
Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to start looking
2015 Jan 25
2
Wiki (pjsip+realtime) says don't put the transports into realtime. Still true?
Hi, The asterisk wiki page says: "Sorcery.conf allows you to try to configure other PJSIP objects such as transport using realtime and it currently won't stop you from doing so. However, some of these object types should not be used with realtime and this can lead to errant behavior." Which objects and is this still true in 1.13.1 ? Thanks, Antonio. PS:
2011 May 23
1
[Fwd: FW: extconfig.conf]
Hi Andrew, OK, (the simple fact that those machines are not connected to internet makes that i have to go to those machines and copy them on a usb-stick, so it causes some delay each time...) -------- Forwarded Message -------- Sorry - I meant extconfig.conf - not cdr_mysql.conf (my mistake). I use (and done for a long time) mySQL for realtime storage - and it's never let me down (touch
2015 Jan 29
2
any valid up-to-date info about Kamailio-Asterisk integration ?
Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks about a (completely new, for me at least) approach to build scalable telephony systems, using N instances of Kamailio and N instances of Asterisk Are there any