[asterisk-users] agi and transfer
hi all, I wouldlike use an agi script in order to send some information at an othe server, so use an agi. But I wouldlike use this agi after or just front blind transfer or attended transfer. it is possible to execute an agi after or front a transfer via features.conf or an other way? thanks for your answer -- sylvain Garcia ___ --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] agi and transfer
Paul a écrit : In some situations you could execute agi by just adding it to an extension on the other server that gets the transferred call. The associated information could be passed by various means. That decision would be based on criteria like the frequency and volume of these transfers. A simple prototype setup could be done using nfs. The agi on server A writes things to a file and then transfers the call to an extension on server B. That extension executes agi which reads the file and further call processing is based on the file contents. Sylvain Garcia wrote: hi all, I wouldlike use an agi script in order to send some information at an othe server, so use an agi. But I wouldlike use this agi after or just front blind transfer or attended transfer. it is possible to execute an agi after or front a transfer via features.conf or an other way? thanks for your answer -- sylvain Garcia ___ --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 sorry, i think that my english is very bad. But I don't want ttansfer to another asterisk server. I have one asterisk server and for exemple when user make blind transfer or attended transfer I wouldlike with a script AGI in perl, send Callerid Number on tcp socket to another server ( not asterisk but simple perl script which listen on tcp port). So in fact, i would like excute an AGI when user use blind or attended transfer define in features.conf thanks for your answer. ___ --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] cdr_mysql does not write to mysql db
Matthew Boehm a écrit : sylvain garcia wrote: Kib Eki a écrit : Asterisk don't use directly mysql database for cdr, astersisk use odbc and odbc connect to mysql. So you must configure odbc corectly wiyt libmyodbc (on debian) the config file are here: Wrong. Asterisk can and does connect to MySQL directly. (Where the hell are these people getting this wrong info?) -Matthew Sorry Mea Culpa ___ 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
[Asterisk-Users] echo capi AVM fritz card
Hi All, I'm running asterisk 1.0.7 on debian sarge, and hardware-wise an AVM Fritz!PCI Card, together with chan_capi 0.3.5. The problem is that any inbound/outbound calls on analogue line result in echo on MY end (the asterisk end). I've played with the echo settings in capi.conf (mainly turning on echocancel and echosquelch, also tried playing with rxgain/txgain) to no avail. The only setting that has helped (somewhat) so far is enabling echosquelch. The echo disappears but a new problem arises. When the person on the other end starts to talk, the first bit is chopped off, and the last bit (before they go quiet) so it almost sounds as though it's doing voice detection and transmitting only when it detects voice. Also, if the other person is talking and i start to talk, they get cut off immediately so this isn't a practical workaround. Any help will be muchly appreciated. ___ 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] cdr_mysql does not write to mysql db
Kib Eki a crit: Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ 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 Asterisk don't use directly mysql database for cdr, astersisk use odbc and odbc connect to mysql. So you must configure odbc corectly wiyt libmyodbc (on debian) the config file are here: /etc/odbcinst.ini : [MySQL] Description = MySQL driver Driver = /usr/lib/odbc/libmyodbc.so Setup = /usr/lib/odbc/libodbcmyS.so CPTimeout = CPReuse = FileUsage = 1 /etc/odbc.ini : [MySQL-asterisk] Description = MySQL Asterisk Database Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = @ipofMysqlddatabase (not domain name) User = Password = Database = asterisk Option = 3 #Port = /etc/asterisk/cdr_odbc.conf : [global] dsn=MySQL-asterisk username=database_username password=database_password loguniqueid=yes ___ 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] attended transfert
David Romero a écrit : attended transfer are implemented on some cases on the phone side, if you need attended transfers on dial plan you need use asterisk CVS HEAD, i are using asterisk CVS HEAD and attended transfer work very well. just install asterisk CVS HEAD and configure features.conf file, on voip-info.org http://voip-info.org have good example of features.conf On 7/21/05, *sylvain garcia* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David RomeroROMDAV ## ___ 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 tx ___ 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] attended transfert
David Romero a crit: attended transfer are implemented on some cases on the phone side, if you need attended transfers on dial plan you need use asterisk CVS HEAD, i are using asterisk CVS HEAD and attended transfer work very well. just install asterisk CVS HEAD and configure features.conf file, on voip-info.org have good example of features.conf On 7/21/05, sylvain garcia [EMAIL PROTECTED] wrote: hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks ___ 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 -- David Romero ROMDAV ## ___ 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 if i use asterisk 1.0.5 on debian attend transfert is present in feature.conf, but doesn't work? it's also for CVS head? ___ 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] attended transfert
sylvain garcia a crit: David Romero a crit : attended transfer are implemented on some cases on the phone side, if you need attended transfers on dial plan you need use asterisk CVS HEAD, i are using asterisk CVS HEAD and attended transfer work very well. just install asterisk CVS HEAD and configure features.conf file, on voip-info.org http://voip-info.org have good example of features.conf On 7/21/05, *sylvain garcia* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David RomeroROMDAV ## ___ 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 tx ___ 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 sorry i have 1.0.7 version it's possible of attended transfer? ___ 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
[Asterisk-Users] attended transfert
hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks ___ 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] System Jsut hangs Up
Tim King a crit: I took care of my earlier problem. But now if I call in it just says goodbye, And on my extension no matter what I do it seems to just hang up on me immediately. Its a slackware 10.1 box with Digium 22b card. I am running AMP so its mysql driven. Im not seeing any errors. It just hangs up. Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___ 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 Could you describe your problem with your extensions.conf And send me your email please sorry for my english i'm french ___ 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
[Asterisk-Users] Fax DETECTION with CAPI
hi, I have dabian sarge with asterisk 1.0.7 and chan_capi 0.3.5 with AVM fritz card. I would like use detecion of fax, but it don't work. So, i would like know if it's possible to work fax detection with this card? And if it's possible how?? Thanks you for your help ___ 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
[Asterisk-Users] AVM CAPI INSTALLATION
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.
capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.101-02 (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
But if I call to asterisk even if asterisk isn't start, it's busy,I dont
unsderstand Help me
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Re: [Asterisk-Users] AVM CAPI INSTALLATION
Armin Schindler a crit:
On Tue, 28 Jun 2005, sylvain garcia wrote:
I use AVM fritz card A1, I have install with instruction of
Voip-info.or, but it don' work.
Which version of Asterisk and chan_capi do you use?
How does your capi.conf look like?
Armin
capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.101-02 (49.18)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
But if I call to asterisk even if asterisk isn't start, it's busy,I dont
unsderstand Help me
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I use Asterisk 1.0.7 on Dedian Sarge with chan_capi 0.3.5.11
I don't configuure my capi.conf it is like orig;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
; msn=50
; incomingmsn=*
;controller=1
softdtmf=1
accountcode=
context=incomingtest
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
;devices=2
;PointToPoint (55512-0)
;for outgoing calls use example 5551212
;and in dialplan you can use callerid like
;exten = _0XXX.,1,StripMSD,1
;exten = _XXX.,2,Dial,CAPI/55512${CALLERIDNUM}:bBYEXTENSION
;
;mode=immediate
;isdnmode=ptp
inal:
my configuration of my dial plan of my internal LAN is correct and
works fine, so I would like test my connection capi in order to receive
incomingcall first and outgoing call in second.
What is the modification to extensions.conf and capi.conf.
Thanks you for your help
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Re: [Asterisk-Users] AVM CAPI INSTALLATION
Armin Schindler a crit: On Tue, 28 Jun 2005, sylvain garcia wrote: Armin Schindler a crit : On Tue, 28 Jun 2005, sylvain garcia wrote: I use AVM fritz card A1, I have install with instruction of Voip-info.or, but it don' work. Which version of Asterisk and chan_capi do you use? How does your capi.conf look like? I use Asterisk 1.0.7 on Dedian Sarge with chan_capi 0.3.5.11 I don't configuure my capi.conf it is like orig; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ; msn=50 ; incomingmsn=* ;controller=1 softdtmf=1 accountcode= context=incomingtest ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 ;devices=2 This is just a template. You cannot expect this to work without changing the settings to your needs and enviroment. my configuration of my dial plan of my internal LAN is correct and works fine, so I would like test my connection capi in order to receive incomingcall first and outgoing call in second. What is the modification to extensions.conf and capi.conf. That depends on what you want to do and which numbers you use. Armin ok, thanks, I would like receive call to sip phone. In my extension.conf i have: [localnetwork] exten = 123,1,Dial(SIP/555,30,Ttr) I have one number for the external for my AVM CARD: 0572086964 What is the modification for my capi.conf and extension.conf in order to incoming call of 0572086964 are routed to sip channel 555. And for outgoinf call please. Thanks you very much ___ 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] Transfer
In your extension.conf 35,1Dial(SIP/33,Ttr) in order to transfert during a call #33 Victor Alvarez a crit: Hi, I'm afraid I don't know how to use thecommand Transfer. I have a couple of SIP users in the system and although exten = 35,1,Dial(SIP/33) works fine, exten = 35,1,Transfer(33) just don't work. All the description in the wiki is 'Transfer(exten)' without a single example. 35,1,Dial(SIP/33) would be a way to transfer the incoming call from 35 to 33, but what I want to do is to get 33 dialplan, not to dial 33. I mean, if 33 is 33,1,Voicemail that's what I would like to execute when calling 35. Could anybody help me? Thank you, Victor. ___ 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 ___ 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
[Asterisk-Users] webvmail debian package
Hi, I wouldlike use webvmail on my asterisk, I use debain Sarge with asterisk 1.0.7 package. I have installed package asterisk-web-vmail but when i go to http://MyAsteriskBOx, i have a page of presentation of Apache. Could you help me please? Thanks ___ 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] Debian Vs Fedora
You should choose your distrib if you konw Fedora or Debian. I use Debian because I prefer debian system package in order to update and security patch? Syed Akbar a écrit : Does anyone have any comments about using Debian stable release Vs Fedora for running Asterisk? Syed Akbar Alico Systems Inc www.alicosystems.com Tel: 562-436-1510 ___ 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 ___ 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] webvmail debian package
ok but i don't know where is the web space with cgi scripts. the package "asterisk-web-mail" don't install cgi script, i think Scott Kamp a crit: remove the default index.html from /var/www/htdocs and/or be sure the apache default DocumentRoot is pointing to the web space and not the default debian pages On Mon, 2005-06-20 at 09:25 +0200, sylvain garcia wrote: Hi, I wouldlike use webvmail on my asterisk, I use debain Sarge with asterisk 1.0.7 package. I have installed package "asterisk-web-vmail" but when i go to http://MyAsteriskBOx, i have a page of presentation of Apache. Could you help me please? Thanks ___ 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 ___ 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 ___ 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] VoiceMail
Use Voicemail with mail of your employees in order to send with attachment. Waldo Rubinstein a écrit : I installed Asterisk Voicemail in an office and now most of the employees are complaining that when they're listening to the messages, it takes forever to listen to their messages. The reason being is that before the message is played, the voicemail says the full date and time when the message arrived and that takes a long time. It's like: Friday . June20th. 2000...and...5... etc (you get the idea). Is there anyway to shorten that or even give users the option to not play that? Thanks, Waldo ___ 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 ___ 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
[Asterisk-Users] MeetMe ERROR Unable to dup channel
I would us Meetme for conferance SIP--SIP fist. my Meetme.conf: [rooms] conf = my extensions.conf: exten = ,1,MeetMe() But : == Parsing '/etc/asterisk/meetme.conf': Found Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Jun 16 10:33:22 ERROR[12100]: chan_zap.c:6969 chandup: Unable to dup channel: No such file or directory Jun 16 10:33:22 WARNING[12100]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Jun 16 10:33:22 WARNING[12100]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') I don't unederstand because i don't use zap channel. ___ 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
[Asterisk-Users] Features.conf Set Language
I use features.conf in order to park call, but I would like use french speaker. how set langage in features.conf? Thanks ___ 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
[Asterisk-Users] Request OPTION and 404 Sjphone Xlite
Hi, I have install asterisk and it works fine. But when I use Sjphone and I use Ethereal a Client send Request:OPTIONS sip:obelix.foo and Server answer Status: 404 Not found. But i can talk with two client and asterisk. When I use Xlite i don't have this request it's clean. I don't understand?? ___ 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
[Asterisk-Users] 404 not found
I use client Sjphone which work fine but i have Sniff a traffic.. - Sjphone send packet with OPTIONS to Asterisk - Asterisk ask with 404 not found This sequence come back often in my log. I don't understand why Sjphone Sens OPTION, and 404 not found.. Thanks for your help ___ 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
[Asterisk-Users] Fax + Fritz + Capi + detection
Hello I'm newbie in asterisk and i have a AVM Audiovisuelles MKTG Computer System GmbH Fritz!PCI v2.0 ISDN (rev 02) with CAPI Driver. I would like install fax detection, but i don't know if i should use NVBackground detect; or CapiAnswerFAx; or other. I don't understantd operation of fax. Tx ps: sorry for English i'm french ___ 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
