Re: [asterisk-users] Detecting Called party Ring indication (and act on it)
Hello Ketema: I found this, i hope you serve http://les.net/asterisk/pddpatch/ 2009/8/15 Ketema Harris ket...@midnightoilconsulting.com is there a way to have asterisk short circuit the dial timeout parameter based on called party sending ring progress ? History: I have multiple routes that a call can take. Some routes are not so good and take a long time. Currently I use the Dial time out parameter, but it times out whether or not the called party is ringing or not (basically has to answer) I think the term is PDD (Post Dial Delay)? I wish to move on to the next route if the progress is longer than X sec, but if I get indication of ringing then stay on that route. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Hello Joseph I recommend that you use *The Mediatrix 4100 Series* are very good. Juan M. 2010/3/28 Joseph Begumisa j.begum...@gmail.com Hi, Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? I have a couple of analog handsets that I want to hookup to my asterisk server? Any tested and tried product recommendations are welcome. Thanks. Best Regards, Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with SIP realtime
Hello Jonas: Change this parameter, if you are using Mysql. [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 *dbsock = /var/lib/mysql/mysql.sock* cheersss... 2010/2/22 jonas kellens jonas.kell...@telenet.be The problem was that I had a different value for 'name' and 'username'. How can I have the 'name' different from the 'username' ??? Why do these 2 need to be the same ?? Jonas. On Mon, 2010-02-22 at 20:36 +0100, jonas kellens wrote: Little fault in my mailing : The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/Asterisk/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/Asterisk/sip_buddies My database-name is just 'Asterisk', my bad. So... what am I missing for this realtime SIP to work ?? Jonas On Mon, 2010-02-22 at 20:13 +0100, jonas kellens wrote: I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from 'sip:test...@192.168.1.150sip%3atest...@192.168.1.150;transport=UDP' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/Asterisk/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/Asterisk/sip_buddies I have the following in extconfig.conf : sipusers = mysql,Asterisk,sip_buddies sippeers = mysql,Asterisk,sip_buddies I have the following in res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 dbsock = /tmp/mysql.sock Something I'm missing ?? Need extra configuration ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Verifone-Agi
Hi. I had developed an IVR solution which is able to configure all type of incoming campaigns showing the IVR structure in a tree mode. Also its actions allow to execute external queries (SQL), Shell applications, Web service invocation, all in order to validate IDs, get phrases to play or execute transactional operations and also simple functional-call-actions. This IVR is a Windows-based windows service and can be configured remotely by a client app. This week in another project I have to develop an application using the VeriFone vx510 device and I read this device needed or could use a PPPoE connection in order to validate and send all information collected from the end user. My question is if I can use the asterisk and the IVR I built to interact with the VeriFone. I mean, VeriFone-E1 or FXs-o-Asterisk-MyAgiIVR. Obviously I have to adapt my IVR to interpret the incoming information from the VeriFone and can I return information to the VeriFone device in real-time?. Greetings ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] International Carriers
Hello Facundo, i have an entreprise in Peru, if you want i can give you a best price for call in peru. My traffic is on net. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Facundo Ameal Enviado el: Viernes, 26 de Enero de 2007 10:55 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] International Carriers Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --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
[asterisk-users] ZT_SPANCONFIG failed on span 1: No such device or address (6)
Help me please.. ZT_SPANCONFIG failed on span 1: No such device or address (6) how can i fixed this problem. Thank you. JmiguelY ___ --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] Asterisk In ternal sip calls I can´t send/recive
Hola Omar:
solo cambia tu extension.conf
[entrada]
exten = s,1,Wait,11
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Dial(SIP/200,60,Ttr)
exten = s,5,Dial(SIP/201,60,Ttr)
exten = s,6,Dial(SIP/202,60,Ttr)
exten = s,7,Dial(SIP/203,60,Ttr)
Saludos.
- Original Message -
From: Omar Lopez Limonta [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 12:46 PM
Subject: [Asterisk-Users] Asterisk Internal sip calls I can´t send/recive
When i made internal call into my LAN using x-lite sip phone client I
retrive in askterisk CLI :
---
ERROR
--
Verbosity is at least 6
-- Remote UNIX connection
-- Executing Dial(SIP/201-979d, SIP/201|60|t) in new stack
-- Called 201
May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
== No one is available to answer at this time
-- Executing VoiceMail(SIP/201-979d, 201) in new stack
-- Playing 'vm-intro' (language 'es')
== Spawn extension (anurix, 201, 2) exited non-zero on 'SIP/201-979d'
May 29 18:09:34 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 52991
(Non-critical Response)
-
(192.168.1.44 is the Asterisk HOST)
I can do outgoing calls with Zap interface without problems, only i
__can´T__ do calls into my lan with SIP phone/protocol , i can listen
voicemail because is the second action on extesion.
These are my configuration files:
sip.conf
-
[203]
type=friend
qualify=yes
username=203
secret=203
host=dynamic
callerid=\JuanI\ 203
canreinvite=no
reinvite=no
context = anurix
transfer=yes
mailbox=203
callgroup=1
pickupgroup=1
nat=never
--
extensions.conf
--
[exterior]
exten = _0.,1,Dial(Zap/1/${EXTEN:1},60,r)
exten = _0.,2,Hangup
;Contestar llamada
[entrada]
exten = s,1,Wait,11
exten = s,2,Answer
exten = s,3,Wait,1
exten = 1,1,Dial(SIP/200,60,Ttr)
exten = 2,1,Dial(SIP/201,60,Ttr)
exten = 3,1,Dial(SIP/202,60,Ttr)
exten = 4,1,Dial(SIP/203,60,Ttr)
;BUZONES DE VOZ DESAHABILITADOS
[anurix]
include = exterior
exten = 200,1,Dial(SIP/200,60,t)
exten = 200,2,Voicemail(200)
exten = 200,3,Hangup
exten = 201,1,Dial(SIP/201,60,t)
exten = 201,2,Voicemail(201)
exten = 201,3,Hangup
exten = 202,1,Dial(SIP/202,60,t)
exten = 202,2,Voicemail(202)
exten = 202,3,Hangup
exten = 203,1,Dial(SIP/203,60,t)
exten = 203,2,Voicemail(203)
exten = 203,3,Hangup
--
http://www.tuactualidad.com
IM: pollo.es.pollo en gmail.com
Te lo traigo fresco.
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Re: Re: [Asterisk-Users] doubts about asterisk configuration fromdatabase
Hello: Do you need install Mysql-devel. Best Regards - Original Message - From: 吴应芳 [EMAIL PROTECTED] To: Asterisk Users Mailing List - No asterisk-users@lists.digium.com Sent: Monday, May 29, 2006 12:04 AM Subject: Re: Re: [Asterisk-Users] doubts about asterisk configuration fromdatabase another questions！ According asterisk realtime sip webpage，I had done following steps： (1) Make, make install asterisk-addons then copy res_mysql.conf.sample to res_mysql.conf and edit the res_mysql.conf with my databases parameter (2) Edit extconfig.conf ---add sip.conf = mysql,asterisk,sipfriends (3) Create a sipfriends table in asterisk database and register some sip phone into the table (4) then restart asterisk... but ... *CLI May 29 12:50:08 WARNING[22360]: config.c:920 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available ask: MYSQL given database driver would be ? thanks~~ hello,, Yes, asterisk can use realtime mode On 5/29/06, 吴应芳 [EMAIL PROTECTED] wrote: hi, I want to complete asterisk configuration from database(MYSQL),now I come across some doubts: 1. http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+databasediff2=3 says that Dynamic 'friends' (Asterisk v1.0.*) and the number of options supported by this 'MySQL_Friends' system is currently very limited,at the same time I find asterisk-1.2.* don't provide this functions,why? for other factors? 2. If I want to do it with asterisk 1.2.*, do I need to add only chan_sip.c and Makefile files in channels directory? 3.Is there any other way to complete asterisk configuration from database? thanks ! BEST REGARDS! Sharon ___ --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 -- Jeffery `∧ ∧�� ミ^r^ミ灬)~ iaxtel Num: 1-700-576-1311 fwdnet Num: 728150 http://www.diaip.com ___ --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 ___ --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
