Re: [asterisk-users] B2BUA Webbased and Click 2 dial apps
Dinesh wrote: Hello, I have a requirement of bridging 2 sip connections via asterisk, which has to be web based. A person has to go to a webpage and enter his from sip uri(say sip1) and enter another sip uri(say sip2). Upon pressing the connect button, the webpage needs to send say a dial sip1 uri and dial dip uri 2 and bridge the call? Do I need any special sip api for this? Any ideas will be nice J . Does this webpage has to be on asterisk server running on the machine? Or can it be passed as a string to the server from the webserver? The easiest way to implement this is by placing a call-spool file in the proper directory - usually /var/spool/asterisk/outgoing, depends on your asterisk setup. More details here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out You can make this work locally (your cgi/jsp/whatever runs on the same box as the asterisk daemon) - or via nfs, ftp or a million other 'dirty-hack' paths; but there's also another, much cleaner approach using the Asterisk Manager Interface 'Originate' action: http://www.voip-info.org/wiki/index.php?page=Asterisk%20Manager%20API%20Action%20Originate Regards, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] WWW: www.imcb.a-star.edu.sg http://www.imcb.a-star.edu.sg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
Richard Cook wrote: We use PostgreSQL in house. It performs wonderfully and cross-platform drivers (ODBC, .NET) are way further along than MySQL. We switched from MySQL a couple of months ago and have never been happier. We use Postgres exclusively too (12 databses, several of them with several millions of records, both OLAP and OLTP roles). We switched from informix 4 years ago and we also subscribe to the never been happier point of view. begin:vcard fn;quoted-printable:=CE=91=CF=80=CF=8C=CE=BB=CE=BB=CF=89=CE=BD =CE=9A=CE=BF=CF=85=CF=84=CE=BB= =CE=AF=CE=B4=CE=B7=CF=82 n;quoted-printable;quoted-printable:=CE=9A=CE=BF=CF=85=CF=84=CE=BB=CE=AF=CE=B4=CE=B7=CF=82;=CE=91=CF=80=CF=8C=CE=BB=CE=BB=CF=89=CE=BD email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Install on Kernel 2.6.x
Shawn Parker wrote: i know asterisk itself will install on a linux kernel 2.6.x, but i've seen places say that the zaptel drivers wont? is this still true? is it possible to build asterisk/zaptel on a linux 2.6.x kernel? # uname -a Linux asterisk2 2.6.7 #1 Tue Aug 3 10:52:26 CET 2004 i686 unknown unknown GNU/Linux # asterisk -r Asterisk CVS-HEAD-08/19/04-14:25:20, Copyright (C) 1999-2004 Digium. # lsmod Module Size Used by wct4xxp82688 0 zaptel230756 139 wct4xxp ...I guess both work ;-) This machine switches more than 2000 callminutes a day on a production environment. -- Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No B-Channels. PRI. E100P. HELP!
Holger Schurig wrote: Just a wild guess (I never worked with this equipment): try bchan=1-15,17-31 dchan = 16 span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 That's my setup and it works fine with our Teles switch. By the way: what is on channel 0 ? timing. Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM AUDIOFiles
jeff quade wrote: I have read this doc-- but we need an answer which DOES NOT USE SOX: can't you acquire the audio files in .wav format? you can then: sox input.wav -r 8000 output.gsm polyphase ...on your linux box. We will be using PROTOOLS-Mix-Plus, or can use PROTOOLS-HD to produce our audio files-- But are looking for ANY Macintosh or PC Audio-Production software suggestion , (outside of using SOX) which will take an .aiff or .wav file and turn it into a GSM file. I still don't understand why you would so desperately need to do the conversion on a mac... anyway, I believe you might be able to compile sox on MacOS X Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM Audio Files
jeff quade wrote: The SOX resampled files work on our asterisk box-- but I gotta put someone else in the loop-- resampling the audio engineers .wav or .aiff files (hes a radio guy who works in .aiff at 44.1-32bit float) Im looking for a solution (software, and prefs) which will take the middle man (and SOX) out of the production loop-- ie the Audio Engineer simply masters and hands off GSM files which will work. I was hoping to find someone who has produced the correct gsm files without SOX, on either a MAC or PC. An alternative: export to WAV file, 8kHz 16bit integer. asterisk's quite happy with these, too (and you get a quality boost in no-compression channels). That's what I do, actually Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P Q.931
Simon wrote: Can anyone help i have * running on debian with a te410p , my telco tells me i need it to run in Q.931 anyone know how to make this happen ? That's the Layer 2 protocol, PRI signalling. You would obviously do CPE signalling (insert a line signalling=pri_cpe in /etc/asterisk/zaptel.conf) - first you need to get Layer 1 up, of course, and define which channel to be used for signalling in /etc/zapata.conf ( dchan=XX ). Apollon Koutildes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some (lack of) answers regarding the wakeup call application...
Rob Fugina wrote: It has occurred to me that the two AGI scripts could be rewritten as real compiled asterisk applications, but then it always hits me that without some kind of cron-line built-in scheduler, or changes to the outgoing call queueing that would allow a call to be scheduled for the future, there would still be that external cron-driven shell script. Ugly. Actually, there's no need for anything like that. Set the file's modification time to the value you require, and watch asterisk do all the dirty work for you. Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to pass call duration to an agi script
Philipp von Klitzing wrote: Not sure if also the csv version of the CDR records does include the unique id. It does, although not by default: cdr_csv.c: /* #define CSV_LOGUNIQUEID 1 */ /* #define CSV_LOGUSERFIELD 1 */ Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI/php script not working
Iain Stevenson wrote: 'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber followed by a valid string of arguments. Do a show application saynumber in *. In the meantime, you might as well try a show agi yourself :-) Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What has happened to my asterisk/PRI ?
Christoph Adomeit wrote:
May 14 09:16:21 VERBOSE[1180010432]: -- Extension '9149' in context
'dtagpri' from '02166458729' does not exist. Rejecting call on channel
10, span 1
I am sure all these callers called more Numbers than 9149, they
might have
called 9149-0 or 9149-xx but not the 9149 alone.
(...)
[dtagpri]
exten = _9149.,1,Dial,ZAP/g2/${EXTEN};
exten = _9149.,2,Hangup
Does somebody have an idea what has happened ?
Just a hunch: Try
overlapdial=yes
in your zapata.conf file.
Apollon Koutlides
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Re: [Asterisk-Users] 3 companies 1 card
Altus Snyman wrote: But who do I differentiate between the different number,how do I say: if a caller calls 1234(the destination) do: [company1] exten = s,1,Answer exten = s,1,Playback,company1-welcome ens. Normally this would be done by setting a context for each DNO in the device's configuration file. In modem.conf for an i4l device for example, it goes like: context=company1 incomingmsn=2108122444 context=company2 incomingmsn=2108122888 I don't know anything about this device you're using, so I can't be more specific... Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play a file
Dudlik wrote: than you and I have Wildcard TE410P in my * server What can I do when a client A call from another telecomunication operator over E1 to my IAX client ? Telecomunication operators usually use the unavailable messages and I thing they don't bill these calls between their customers. How do they do it ? I can tell you about the situation here in Greece, and can't tell that properly either since I don't have a thorough understanding of several telephony issues: What's done is that the B-Channel is opened while the call is still in progress, signalling-wise. Then a short message is delivered, the channel is closed and the call rejected with an appropriate cause value. Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Matching variable-length extensions with chan_zap in overlap dialling
Jeremy McNamara wrote:
Try exten = _0X. --- notice the period
[m807oth]
exten = _80780780.,1,StripMSD(7)
exten = _0.,1,SetVar,clidest=${EXTEN}
exten = _0.,2,Goto(cli,s,1)
...noticed mine? :-) I've tried a combo-wildcard (with an X, as in your
example) as well, with no results either. The code in chan_zap.c seems
to confirm that in overlap digit transmission the channel driver doesn't
check for multiple matches. The patch to check for
multiple/ambiguous/possibly incomplete matches is trivial, but
implementing the timeout is definitely not.
Apollon Koutlides
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[Asterisk-Users] Matching variable-length extensions with chan_zap in overlap dialling
I've been having trouble matching variable extensions on a zap channel
(an E1 line). Doing it the extensions.conf way:
[pri1]
; Match 8078078- calls
include = m807nat
include = m807mob
include = m807oth
[m807nat]
exten = _80780782X,1,StripMSD(7)
exten = _2X,1,SetVar,clidest=${EXTEN}
exten = _2X,2,Goto(cli,s,1)
[m807mob]
exten = _807807869,1,StripMSD(7)
exten = _69,1,SetVar,clidest=${EXTEN}
exten = _69,2,Goto(cli,s,1)
[m807oth]
exten = _80780780.,1,StripMSD(7)
exten = _0.,1,SetVar,clidest=${EXTEN}
exten = _0.,2,Goto(cli,s,1)
...when I dial, say, 00441565652244 * will match the first wildcard
digit immediately:
-- Accepting call from '2108126055' to '807807800' on channel 1, span 1
I've tried using an AGI to capture the rest of the digits, but that
didn't work either (wait for digit catches no digits), since the channel
is not answered yet (and I don't want to do that).
DigitTimeout in extensions.conf is of no consequence either, as long as
the call is not answered.
Looking in the bug archive I found this:
http://bugs.digium.com/bug_view_page.php?bug_id=0001422
which only perplexed me more... I tried to hack a bit of chan_zap (my
competence in C is far below adequate) and at least managed to avoid
matching immediately when there are more than one matches, but I got
stuck with the timeout issues :-)
Before trying (or rather paying somebody else with more programming
experience) to hack the chan_zap code to fit my needs, I thought I'd
consult the people on the list... any hints? Is there something I'm
missing here?
Apollon Koutlides
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