[Asterisk-Users] Asterisk Using Multiple Databases with ODBC?
Hello! Does anyone know how I can use two diferent databases with ODBC with the same erealtime family? Something like this: res_odbc.conf: ;;; odbc setup file [ast_cnf1] dsn = ORACLE username = asterisk password = asterisk pre-connect = yes [ast_cnf2] dsn = MySQL username = asterisk password = asterisk pre-connect = yes and the extconfig.conf: voicemail = odbc,ast_cnf1,voicemail_conf voicemail = odbc,ast_cnf2,voicemail_conf Thanks, jsalas. ___ --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] Asterisk didn't start with
Hello I Installed the Ceptral voicesand Iam trying tu use the swift module with asterisk. But when I start it show: [app_swift.so]May 15 17:53:09 WARNING[18876]: loader.c:325 __load_resource: libswift.so.4: cannot open shared object file: No such file or directoryMay 15 17:53:09 WARNING[18876]: loader.c:554 load_modules: Loading module app_swift.so failed! Il looked for that library (libswift.so.4) and I founded at /opt/swift/lib/. Where I must put this library? or maybe the Makefile is wrong? Thanks Jsalas ___ --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 didn't start with app_swift.so
Hello I Installed the Ceptral voicesand Iam trying tu use the swift module with asterisk. But when I start it show: [app_swift.so]May 15 17:53:09 WARNING[18876]: loader.c:325 __load_resource: libswift.so.4: cannot open shared object file: No such file or directoryMay 15 17:53:09 WARNING[18876]: loader.c:554 load_modules: Loading module app_swift.so failed! Il looked for that library (libswift.so.4) and I founded at /opt/swift/lib/. Where I must put this library? or maybe the Makefile is wrong? Thanks Jsalas ___ --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 Realtime with Oracle
Hello I use odbc and it works very well. For this I use the easysoft driver. regards. jsalas -Mensaje original- De: Kamran Ahmad [mailto:[EMAIL PROTECTED] Enviado el: Tuesday, May 09, 2006 5:25 AM Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Asterisk Realtime with Oracle HI all I want to connect Asterisk(using realtime) with Oracle. any one have any idea which one is the best method for this. ODBC/ or some other interface modules avaliable for directly connecting with oracle ? thanks Kamran Ahmad __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
RE: [Asterisk-Users] Two asterisk process in one hardware.
Hello I'm using voicemail with realitime. And I need use two diferent and separate databases. thanks. jsalas -Mensaje original- De: Mike Fedyk [mailto:[EMAIL PROTECTED] Enviado el: Monday, April 24, 2006 8:24 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Two asterisk process in one hardware. Juan Salas wrote: Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) What end outcome do you want? Maybe there is another way to do it... ___ --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] Two asterisk process in one hardware.
Hello. Has anybody knows how run two asterisk process in one hardware? (each one with its own configuration?) Thanks Juan Salas. ___ --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] oh323 signal update support
Hello We have detected that the newer Cisco IOS versions include a SignalUpdate message after each alphanumeric UserInputIndication. Did the oh323 asterisk module support SignalUpdate? Has anybody know something? Thanks Jsalas ___ --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] How to transmit Video
Look Eyebeamof Xterm. -Mensaje original-De: RAHEEL HASSAN [mailto:[EMAIL PROTECTED]Enviado el: Thursday, March 16, 2006 4:05 AMPara: asterisk-users@lists.digium.comAsunto: [Asterisk-Users] How to transmit Videoplease tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . Yahoo! MailUse Photomail to share photos without annoying attachments. ___ --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] G729, G729 annex A or G729 annex B?
Hello Some questions about codecs.. What's the difference between the this codecs? Which is used by asterisk? Thanks Juan Salas ___ --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 silence suppression?
I will try to test your adaptation. How I congfigureto enable VAD? Regards Jsalas -Mensaje original-De: Moises Silva [mailto:[EMAIL PROTECTED]Enviado el: Friday, February 17, 2006 11:26 AMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] asterisk silence suppression? The patch you saw is not for the stable branch. Salu2 Jsalas Right, but try using this, i adapted it, no guarantees, i have not made tests, just modified it to apply properly, it would be great if some one can test it:http://chewbacca.ivsol.net/asterisk-1.2.1-silence-suppression-4.patchRegardsOn 2/17/06, Rob Lith [EMAIL PROTECTED] wrote: That a phone setting you must set to not supress silence - i.e. in X-Lite/eyeBeam in the advanced settings/audio there is a silence setting.Same for the SNOMs, most phones should have it.RegardsRob On 2/15/06, Dan Elder [EMAIL PROTECTED] wrote: Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads? Thx as always___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org " ___ --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] a=fmtp:18 annexb=no
Hello Looking the SIP debug we see a change in the SETUP message from the Asterisk 1.0.x version to the 1.2.4. In the 1.2.4 we notice this line: a=fmtp:18 annexb=no This line cause problems in our plattform (We think our SIP - h323 gateway can't parse this line) Why this line its present in 1.2.4 version? Have anybody some clue? Regards JS. ___ --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 silence suppression?
The patch you saw is not for the stable branch. Salu2 Jsalas -Mensaje original-De: Moises Silva [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, February 15, 2006 2:28 PMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] asterisk silence suppression?Asterisk DOES NOT HAVE silence suppression (VAD) support for now. So it cannot be disabled or enabled. Simply does not exists. A couple of weeks ago i saw a patch to enable it. The link here:http://bugs.digium.com/view.php?id=5374so unless you have the previous patch, you should disable silence suppression in the clients.Regards On 2/15/06, Dan Elder [EMAIL PROTECTED] wrote: Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads?Thx as always___--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-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" ___ --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] Cisco 2620 as PRI gateway
If you
are using the 2620 like a SIP IP-PSTN gateway
your
voip dial-peer would be like this:
dial-peer voice
635099 voipdescription calls sent to Asteriskpreference
1destination-patternT
(or whatever youneed to match)session
targetsip-serverdtmf-relay h245-alphanumeric (or
whatever you need)
session-protocol sip
no vad
And you need a pots
dial-peer,
something like this
dial-peer voice 0 potsdestination-pattern
T (or whatever you need)port
0/0 0
And in sip-ua:
sip-ua sip-server asterisk server ip
address
This is the basic
Regards
Jsalas
-Mensaje original-De: Tim Reimers
[mailto:[EMAIL PROTECTED]Enviado el: Thursday,
February 09, 2006 10:04 AMPara: Asterisk Users Mailing List -
Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] Cisco 2620 as
PRI gateway
Yeah-- sorry...
"
dial-peer voice 635099 voipdescription calls
sent to Asteriskpreference 1destination-pattern
[635-9]..progress_ind setup enable 3session target
ipv4:10.10.1.28dtmf-relay h245-alphanumeric
"
I had been trying to do this with H.323 -- the Call
Manager uses H.323
There are some sip commands available in that dial-peer
ACS-GW(config-dial-peer)#voice-class sip ?
rel1xx Type of reliable provisional response
support transport Configure transport related
parameters url url type in
request line of outgoing INVITE
Not sure how I set those---
This:
voice-class codec 1voice-class h323
1
is what is in there for the Call Manager h.323 dial-peer
That's obviously NOT what I want for the Asterisk-SIP
connection...
but I don't know what Ineed to do regarding the
'sip url' or 'sip transport' or 'sip rel1xx' commands, if
anything...
How does one debug SIP activity? I see the debugs for
calls--- but I don't know the related debugs for actively
watching--
like you would 'debug isdn q931' -- that's the
outgoing side of the router--
what would be the debug for a SIP call 'arriving' at the
router??
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan
SalasSent: Wednesday, February 08, 2006 2:17 PMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject:
RE: [Asterisk-Users] Cisco 2620 as PRI gateway
Did
you create the dial-peers in the2651?
-Mensaje original-De: Tim Reimers
[mailto:[EMAIL PROTECTED]Enviado el: Wednesday,
February 08, 2006 1:41 PMPara: Asterisk Users Mailing List -
Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] Cisco 2620
as PRI gateway
sip-ua sip-server ipv4:asterisk server
ip address
OK -
So I added those lines to my 2651 with the IP of my
asterisk box...
How would I set this up as a SIP trunk in
Asterisk?
I have done this, in building a SIP trunk in
AMP.
host=10.12.1.252type=friend
I
don't know if/how to specify a username/password (as was the defaults in
there- the router didn't support having that
configured..)
So
I picked friend..
Then, in call routing, I picked my "Outbound
Routing"
the "9_outside" route of "9|."
Set that to use the new 'gw-rtr' I'd created...
no
go...
Debug ISDN q931 doesn't show anything going to the
router...
In
Asterisk-
" -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
back from 10.12.1.252"
snipped from below
The router doesn't show anything...
the below
shows up in Asterisk - mode
-- Executing Macro("SIP/6351-cc18",
"dialout-trunk|3|2439499|") in new stack -- Executing
GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto
(macro-dialout-trunk,s,3) -- Executing
Macro("SIP/6351-cc18", "user-callerid") in new stack
-- Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new
stack -- DBget: varname=AMPUSER, family=DEVICE,
key=6351/user -- DBget: set variable AMPUSER to
6351 -- Executing DBget("SIP/6351-cc18",
"AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack --
DBget: varname=AMPUSERCIDNAME, family=AMPUSER,
key=6351/cidname -- DBget: set variable AMPUSERCIDNAME
to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18",
"0?5") in new stack -- Executing
SetCallerID("SIP/6351-cc18", "Tim-Zyxel 6351") in new
stack -- Executing NoOp("SIP/6351-cc18", "Using
CallerID "Tim-Zyxel" 6351") in new stack --
Executing Macro("SIP/6351-cc18", "record-enable|6351|OUT") in new
stack -- Executing GotoIf("SIP/6351-cc18", "0
0?2:4") in new stack -- Goto
(macro-record-enable,s,4) -- Executing
AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new
stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060208-115748|1139417868.14: Outbound recording not
enabled -- AGI Script
RE: [Asterisk-Users] Cisco 2620 as PRI gateway
Did
you create the dial-peers in the2651?
-Mensaje original-De: Tim Reimers
[mailto:[EMAIL PROTECTED]Enviado el: Wednesday,
February 08, 2006 1:41 PMPara: Asterisk Users Mailing List -
Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] Cisco 2620 as
PRI gateway
sip-ua sip-server ipv4:asterisk server
ip address
OK -
So I added those lines to my 2651 with the IP of my
asterisk box...
How would I set this up as a SIP trunk in
Asterisk?
I have done this, in building a SIP trunk in
AMP.
host=10.12.1.252type=friend
I
don't know if/how to specify a username/password (as was the defaults in
there- the router didn't support having that configured..)
So I
picked friend..
Then, in call routing, I picked my "Outbound
Routing"
the
"9_outside" route of "9|."
Set
that to use the new 'gw-rtr' I'd created...
no
go...
Debug ISDN q931 doesn't show anything going to the
router...
In
Asterisk-
" -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
back from 10.12.1.252"
snipped from below
The
router doesn't show anything...
the below shows up in
Asterisk - mode
-- Executing Macro("SIP/6351-cc18",
"dialout-trunk|3|2439499|") in new stack -- Executing
GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto
(macro-dialout-trunk,s,3) -- Executing
Macro("SIP/6351-cc18", "user-callerid") in new stack --
Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new
stack -- DBget: varname=AMPUSER, family=DEVICE,
key=6351/user -- DBget: set variable AMPUSER to
6351 -- Executing DBget("SIP/6351-cc18",
"AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack --
DBget: varname=AMPUSERCIDNAME, family=AMPUSER,
key=6351/cidname -- DBget: set variable AMPUSERCIDNAME
to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", "0?5")
in new stack -- Executing SetCallerID("SIP/6351-cc18",
"Tim-Zyxel 6351") in new stack -- Executing
NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" 6351") in new
stack -- Executing Macro("SIP/6351-cc18",
"record-enable|6351|OUT") in new stack -- Executing
GotoIf("SIP/6351-cc18", "0 0?2:4") in new stack --
Goto (macro-record-enable,s,4) -- Executing
AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new
stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060208-115748|1139417868.14: Outbound recording not
enabled -- AGI Script recordingcheck completed,
returning 0 -- Executing NoOp("SIP/6351-cc18", "No
recording needed") in new stack -- Executing
Macro("SIP/6351-cc18", "outbound-callerid|3") in new
stack -- Executing GotoIf("SIP/6351-cc18", "1?3") in new
stack -- Goto
(macro-outbound-callerid,s,3) -- Executing
DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new
stack -- DBget: varname=USEROUTCID, family=AMPUSER,
key=6351/outboundcid -- DBget: set variable USEROUTCID
to 6351 -- Executing GotoIf("SIP/6351-cc18", "0?6") in
new stack -- Executing SetCallerID("SIP/6351-cc18",
"6351") in new stack -- Executing NoOp("SIP/6351-cc18",
"CallerID set to 6351") in new stack -- Executing
SetGroup("SIP/6351-cc18", "OUT_3") in new stack --
Executing CheckGroup("SIP/6351-cc18", "") in new stack
-- Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499") in new
stack -- Executing SetVar("SIP/6351-cc18",
"DIAL_TRUNK=3") in new stack -- Executing
AGI("SIP/6351-cc18", "fixlocalprefix") in new stack --
Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf -- AGI Script
fixlocalprefix completed, returning 0 -- Executing
SetVar("SIP/6351-cc18", "OUTNUM=2439499") in new stack
-- Executing Cut("SIP/6351-cc18", "custom=OUT_3|:|1") in new
stack -- Executing GotoIf("SIP/6351-cc18", "0?16") in
new stack -- Executing Dial("SIP/6351-cc18",
"SIP/acs-gw-rtr/2439499") in new stack -- Called
acs-gw-rtr/2439499 -- SIP/acs-gw-rtr-b33f is
circuit-busy == Everyone is busy/congested at this time
(1:0/1/0) -- Executing Goto("SIP/6351-cc18",
"s-CONGESTION|1") in new stack -- Goto
(macro-dialout-trunk,s-CONGESTION,1) -- Executing
NoOp("SIP/6351-cc18", "Dial failed due to CONGESTION") in new
stack -- Executing Macro("SIP/6351-cc18", "outisbusy")
in new stack -- Executing Playback("SIP/6351-cc18",
"allison7/all-circuits-busy-now") in new stack -- Got
SIP response 481 "Call Leg/Transaction Does Not Exist" back from
10.12.1.252 -- Playing 'allison7/all-circuits-busy-now'
(language 'en') -- Executing Playback("SIP/6351-cc18",
"allison7/pls-try-call-later") in new stack -- Playing
'allison7/pls-try-call-later' (language 'en') --
Executing Macro("SIP/6351-cc18", "hangupcall") in new
stack -- Executing ResetCDR("SIP/6351-cc18", "w") in new
stack -- Executing
[Asterisk-Users] oh323 and asterisk v1.2.2
Hello all. Has anybody work with asterisk version 1.2.2 and oh323 module? When I try to make it show: wrapendpoint.cxx:800: error: expected primary-expression before ')' token wrapendpoint.cxx:800: error: `PIsDescendant' undeclared (first use this function) wrapendpoint.cxx:800: error: (Each undeclared identifier is reported only once for each function it appears in.) wrapendpoint.cxx:801: error: expected primary-expression before ')' token make[1]: *** [wrapendpoint.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.3/wrapper' make: *** [subdirs_build] Error 1 any clue? Thanks. Jsalas ___ --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] NOTICE[3589]: frame.c:128 __ast_smoother_feed: Dropping extra fr ame of G.729 since we already have a VAD frame at the end
Hello. Im using Asterisk like IVR card application. It works very well in h323 and SIP, but when the IVR generate a call in SIP it show: Jan 4 15:39:32 NOTICE[3589]: frame.c:128 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end As I see, this is a problem originated by a equipment with VAD activated. But we disable VAD in all our equipment. Someone have ani clue about it? Regards. jsalas ___ --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] appradius
Hello all. Has anybody works with appradius? where can I find documentation? Regards, Jsalas ___ --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] E1 and hardware Test.
Hello I Have a machine (P3) acting like a E1 - SIP gateway (with a digium TE110P) On this asterisk we are running an AGI doing radius acounting (it works very well!) But now we need to make effort test of the hardware we use. How we can simulate many concurrent calls? Has anybody has some clue. Regards. jsalas. ___ --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] Realtime sip register=
yes, I tested too and it's works. The Problem is when we want to add (or delete) registers without reload the asterisk. We are using it like a border server wich is registered like many users in a SER machine and the real endpoints are registered in the asterisk. Regards. Jsalas -Mensaje original- De: Luca Lafranchi Lists [mailto:[EMAIL PROTECTED] Enviado el: Wednesday, October 26, 2005 5:51 AM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Realtime sip register= Hi, I saw (and it's works), that you can mix the realtime and static mode. In extconfig.conf file configure to use sip.conf in realtime ... ;realtime sipusers = mysql,pbx,PBX_sip_buddies sippeers = mysql,pbx,PBX_sip_buddies ... Don't delete the sip.conf file! In the sip.conf file define only the [general] section with your register variables and use sip_buddies table to define your SIP client. When you start Asterisk server, the sip.conf file will read the [general section (with your register variables) and after the sip_buddies table. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: martedì, 25. ottobre 2005 16:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime sip register= Juan Salas wrote: Hello! As I know, the register is a variable of [general] section in sip.conf. You can't use it in database, ie you can't add new registers without reload the asterisk.. You can have a static config in a database, but you will still have to reload. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ NOD32 1.1264 (20051024) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Realtime sip register=
Hello! As I know, the "register" is a variable of [general] section in sip.conf. You can't use it in database, ie you can't add new registers without reload the asterisk.. I am right? Regards. Jsalas. -Mensaje original-De: tijmen van den brink [mailto:[EMAIL PROTECTED]Enviado el: Tuesday, October 25, 2005 9:26 AMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] Realtime sip register=You could check these links. I'm trying to do the sip peer registration like this but I get some error about username / auth name mismatch.I think I do something wrong in the MySQL table.I hope it works for you and if it works I would like to hear it from you.Good luckhttp://www.voip-info.org/tiki-index.php?page=Asterisk+RealTimehttp://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip On 10/25/05, Fahd [EMAIL PROTECTED] wrote: i want to put sip peer registrationcommand register = in my database . anybody have any idea about it how to do thisfahd___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tijmen van den BrinkWilhelminaweg 463441 XC WoerdenTel: 0642233831MSN: [EMAIL PROTECTED]Skype: [EMAIL PROTECTED]SIP:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Passwords and RealTime
Hello. One question... When we use voicemail with flat file configuration (voicemail.conf) the vaicemail user can change his password by voicemailmain (voice menu) this change the value in voicemail.conf. When we use Realtime the password is stored in the database. What the voicemailmain (voice menu) application do? change the database value? As I see it doesn't work. Regard. Jsalas. 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] IODBC instead of UNIXODBC
Hello. It's possible to use IODBC instead UNIXODBC with realtime? As I see, the res Makefile load a odbcinst.h file, but in IODBC there's not this file. I change the res Makefile (iodbcinst.h instead odbcinst.h) and the make create the res_odbc.so. But when asterisk boot it don't start showing: [res_odbc.so]Oct 4 10:24:43 WARNING[9748]: loader.c:314 __load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory Oct 4 10:24:43 WARNING[9748]: loader.c:543 load_modules: Loading module res_odbc.so failed! There is something else I have do? Thanks. JS 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] codec g723 on Via C3
I have a VIA Samuel 2, I use the Intel primitives (g729) setting the Makefile to a 586 processor. Maybe you can test with this. Regards. Jsalas. -Mensaje original-De: Giordano Grandis [mailto:[EMAIL PROTECTED]Enviado el: Monday, October 03, 2005 7:06 AMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: [Asterisk-Users] codec g723 on Via C3 Hi, just a question: anyone has never installed g729 codec on VIA motherboard with C3 processor ? I'm having problem with IPP libraries, and Intel said that it works only on Inter processor. Any suggestion? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk Realtime.. : Unixodbc drivers
Hi! About realtime... Anybody knows a unixodbc driver for oracle (free or comertial)? I am working with a trial easysoft odbc-driver, but the commertial license is very expensive... Another question. res_odbc.so works with IODBC ? Regards. Jsalas ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Help on RealTime Extensions on Oracle DB
Hello I have the same problems, sip.conf and voicemail.conf works fine but I have problems in extensions.conf. When I use postgres extensions works fine! I have also this warning in cli: Sep 23 15:08:59 WARNING[29303]: res_config_odbc.c:92 realtime_odbc: SQL Prepare failed![SELECT * FROM sip_conf WHERE name = ?] Someone has any idea? Jsalas -Mensaje original- De: Chris Deserva [mailto:[EMAIL PROTECTED] Enviado el: Friday, September 16, 2005 1:27 AM Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Help on RealTime Extensions on Oracle DB Does someone here configured RealTime Extensions using ODBC connecting to Oracle DB? Im having a problem in dialplan patterns, it doesnt work. Pls. help! -Chris __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Help on RealTime Extensions on Oracle DB
hello
I'm working with realtime and oracle.
I'm using two tables in oracle (sip_conf and voicemail_conf)
My extensions.conf looks like this:
[datab]
exten = _3XXX,1,Dial(SIP/${EXTEN})
exten = _3XXX,2,Voicemail(u${EXTEN})
exten = _3XXX,3,Hangup
exten = _3XXX,104,Voicemail(b${EXTEN})
exten = _3XXX,105,Hangup
switch = Realtime
It's work very well!
Regards.
JS.
-Mensaje original-
De: Chris Deserva [mailto:[EMAIL PROTECTED]
Enviado el: Friday, September 16, 2005 1:27 AM
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] Help on RealTime Extensions on Oracle DB
Does someone here configured RealTime Extensions using
ODBC connecting to Oracle DB? Im having a problem in
dialplan patterns, it doesnt work. Pls. help!
-Chris
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[Asterisk-Users] MTA V102
Hello! Anybody has tested the MTA-V102 with asterisk? Thanks. JS. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] IVR Documentation and Samples.
Hi We are using an IVR system based in AGI. The AGI makes querys to our radiator server using radius libraries. Itis based on ASTCC and Net-Radius modules. Look this links: http://www.voip-info.org/tiki-index.php?page=ASTCC http://search.cpan.org/~luismunoz/Net-Radius-1.44/ Regards JS. -Mensaje original-De: PJ Santos [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, September 07, 2005 6:53 PMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] IVR Documentation and Samples. Juan Salas, Are you use IVR System with database? Case yes, please send me one sample. Regards. PJSantos.Juan Salas [EMAIL PROTECTED] escreveu: Hi Evrybody. I have tested Asterisk Real Time (asterisk version 1.2 beta) with postgresql (and unixODBC) and it works verywell! Now I will try with Oracle, Have somebody work with it? Regards Jsalas. -Mensaje original-De: PJ Santos [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, September 07, 2005 6:31 PMPara: Asterisk-Users@lists.digium.comAsunto: [Asterisk-Users] IVR Documentation an Sample. Hi everybody. I need documentation and sample, about IVR. Sample about database access with IVR+Asterisk, if its possible. Thanks. Paulo Santos Brasil/RJ. __Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/ ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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]
Hi Evrybody. I have tested Asterisk Real Time (asterisk version 1.2 beta) with postgresql (and unixODBC) and it works verywell! Now I will try with Oracle, Have somebody work with it? Regards Jsalas. -Mensaje original-De: PJ Santos [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, September 07, 2005 6:31 PMPara: Asterisk-Users@lists.digium.comAsunto: [Asterisk-Users] IVR Documentation an Sample. Hi everybody. I need documentation and sample, about IVR. Sample about database access with IVR+Asterisk, if its possible. Thanks. Paulo Santos Brasil/RJ. __Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 and .NET
look at: http://ipswitchboard.thorben.dk regards. jsalas -Mensaje original- De: Alvin Tan [mailto:[EMAIL PROTECTED] Enviado el: Monday, August 08, 2005 8:37 PM Para: Asterisk-Users@lists.digium.com Asunto: [Asterisk-Users] Asterisk and .NET Hi, Are there any Asterisk interfaces with .NET? Thanks, Alvin ___ 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] asterisk registered in ser proxy
You don't see SIP messages arrives on the SIP proxy? Or you don't see messages leaving your asterisk? What show the sip show registry command? Post the sip debug peer 10.0.0.115 logs to see what it is doing. We have a asterisk registered in SER and it works fine! Saludos. jsalas -Mensaje original- De: Jenna Cole [mailto:[EMAIL PROTECTED] Enviado el: Monday, August 08, 2005 9:51 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] asterisk registered in ser proxy I tried that, but asterisk is still NOT sending a REGISTER sip message to the sip proxy. actually, it is not sending any sip message to the sip proxy. Jenna ;) --- Paul Belanger [EMAIL PROTECTED] escribió: In you sip.conf what if you change: register = 7771::[EMAIL PROTECTED]/7771 to register = 7771:[EMAIL PROTECTED]/7771 PB Jenna Cole wrote: im using iptel.org SER proxy. the proxy is working without authentication. the problem is that the Asterisk is not sending a REGISTER sip message. --- Juan Salas [EMAIL PROTECTED] escribió: Which SIP proxy are you using? Check the authentication parameters (user-id, auth-id, password)? Post the sip debug peer 10.0.0.115 logs. Saludos. jsalas -Mensaje original- De: Jenna Cole [mailto:[EMAIL PROTECTED] Enviado el: Friday, August 05, 2005 12:58 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] asterisk registered in ser proxy if i remove that line, asterisk stop sendind the OPTIONS message to the SIP PROXY, but it's still NOT sending the REGISTER message. i would alse need to register more than one number --- Eric Wieling aka ManxPower [EMAIL PROTECTED] escribió: Jenna Cole wrote: thanx for the reply. i tried it, and now asterisk is doing something. but the problem is that instead of sendind a REGISTER message to the SIP PROXY, it is sendind an OPTIONS message, and the PROXY responds with 404 NOT FOUND ihave in my sip.conf file: register = 7771::[EMAIL PROTECTED]/7771 [10.0.0.115] type=peer context=default secret= username=7771 fromdomain=10.0.0.115 canreinvite=yes dtmfmode=RFC2833 qualify=yes host=10.0.0.115 insecure=very fromuser=7771 Remove the qualify=yes and Asterisk will stop sending the options packets. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Only terrorists use the r option to Dial. ___ 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 __ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar ___ 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 ___ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar ___ 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 __ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar ___ 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] asterisk registered in ser proxy
Yes you can.
In sip.conf you must edit:
register = user in SIP proxy:password in SIP proxy:AUTH-ID in SIP
proxy@IP of SIP proxy/local peer in asterisk where you answer the call
and you must define a peer for the SIP proxy:
[SIP-proxy]
type=peer
context=where you have the peer for answer
secret=password in SIP proxy
username=AUTH-ID in SIP proxy
fromdomain=IP of SIP proxy
canreinvite=yes
dtmfmode=RFC2833
canreinvite=yes
qualify=yes
host=IP of SIP proxy
insecure=very
fromuser=user in SIP proxy
disallow=all
allow=g729
Finally, to make a call from asterisk yo need in the extension.conf
something like this:
exten = _X.,1,Dial(SIP/SIP-proxy/${EXTEN})
This should work!
Regards.
Jsalas
-Mensaje original-
De: Jenna Cole [mailto:[EMAIL PROTECTED]
Enviado el: Friday, August 05, 2005 8:25 AM
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] asterisk registered in ser proxy
is it possible to register asterisk in a sip proxy as
if it were a terminal (like a cisco ATA)? how?
Thanx
Jenna ;)
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RE: [Asterisk-Users] asterisk registered in ser proxy
Which SIP proxy are you using? Check the authentication parameters (user-id, auth-id, password)? Post the sip debug peer 10.0.0.115 logs. Saludos. jsalas -Mensaje original- De: Jenna Cole [mailto:[EMAIL PROTECTED] Enviado el: Friday, August 05, 2005 12:58 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] asterisk registered in ser proxy if i remove that line, asterisk stop sendind the OPTIONS message to the SIP PROXY, but it's still NOT sending the REGISTER message. i would alse need to register more than one number --- Eric Wieling aka ManxPower [EMAIL PROTECTED] escribió: Jenna Cole wrote: thanx for the reply. i tried it, and now asterisk is doing something. but the problem is that instead of sendind a REGISTER message to the SIP PROXY, it is sendind an OPTIONS message, and the PROXY responds with 404 NOT FOUND ihave in my sip.conf file: register = 7771::[EMAIL PROTECTED]/7771 [10.0.0.115] type=peer context=default secret= username=7771 fromdomain=10.0.0.115 canreinvite=yes dtmfmode=RFC2833 qualify=yes host=10.0.0.115 insecure=very fromuser=7771 Remove the qualify=yes and Asterisk will stop sending the options packets. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Only terrorists use the r option to Dial. ___ 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 __ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar ___ 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] h323
Yes you can. Try with oh323 module: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -Mensaje original- De: altus [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register = -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] h323
From wiki... (http://www.voip-info.org/tiki-index.php?page=Asterisk+oh323+channels) The second option is valid only in the case where a gatekeeper is used. OH323 supports only one gatekeeper (or none, but not multiple gatekeepers). OH323 itself only acts as H.323 Gateway. As I look, asterisk didn't act like gatekeeper. JS. Yes, it worked here. part of oh323.conf example: . . . ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=asterisk alias=99001701 alias=99001702 . . . This defines h.323 id and the aliases for each channel. So, now I would like to know if asterisk can support h.323 gateway registration, like SIP. Can a h.323 gateway register on asterisk ? Thanks -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 10:54, Juan Salas wrote: Yes you can. Try with oh323 module: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -Mensaje original- De: altus [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register = -Mensaje original- De: Daniel Varella de Oliveira [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 10:42 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h323 ___ 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] how to compile asterisk-oh323
Look this link: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html Regards. Jsalas -Mensaje original- De: wassim darwish [mailto:[EMAIL PROTECTED] Enviado el: Tuesday, July 26, 2005 7:30 AM Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] how to compile asterisk-oh323 if any one can tell how to compile asterisk-oh323 and what it is dependencies. Regards; wassim Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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] Hung Sip Channels
Does anyone know how to get rid of these hung channels? I am getting this when I do a: show sip channels 209.82.xxx.xxx 0071495217 2591218534@ 00103/1 unknow(d) 209.82.xxx.xxx 0041590104 0690231739@ 00103/1 unknow(d) 209.82.xxx.xxx 0070259259 3265102826@ 00103/1 unknow(d) 209.82.xxx.xxx 0071948143 1927207026@ 00103/1 unknow(d) 209.82.xxx.xxx 0022576786 1752809624@ 00103/1 unknow(d) 209.82.xxx.xxx 0070153955 0085223171@ 00103/1 unknow(d) I have about 60 of them and growing. I have submitted a ticket with my provider to let them know of this problem but I would like to clear them out w/o restarting the asterisk binary. I have the same problem. For me it looks like this some not completed transactions. This hanged transcations don't affect anything, but I worry, how much their number will increase, when I will have bigger load on asterisk box (Currently near 20 concurent calls.) I think dumping of all signalling and analyzing it can help. I have the same problem too. The number of this hanged transactions increase quickly. It didn't cleat with a asterisk reload (but a linux reboot works). Anyone have some news about this? JS. ___ 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] HOWTO capture digits
If you use perl you need to install the perl library for asterisk, look for it in: http://asterisk.gnuinter.net/ A good example to start (it show the use of GET_DATA function) is: http://www.dynx.net/ASTERISK/AGI/ccard/agi-ccard.agi More examples in perl and other languajes: http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI Regards. jsalas. -Mensaje original- De: J.Raborg [mailto:[EMAIL PROTECTED] Enviado el: Wednesday, July 20, 2005 3:39 PM Para: Juan Salas Asunto: RE: [Asterisk-Users] HOWTO capture digits Thanks Juan for your promptly reply, by any chance do you have any examples. What do I try to accomplish is capture the digits into a file and the post it to an URL site. and also use this as if you know your party extension please dial now not sure how to do it. Regards, Thanks in advance. J.Raborg You can use an AGI to get the DTMF digits (for example in perl with the function GET_DATA()) then you can easy save this data in a file. Jsalas -Mensaje original- De: J.Raborg [mailto:[EMAIL PROTECTED] Enviado el: Wednesday, July 20, 2005 2:14 PM Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] HOWTO capture digits Folks: does anybody have an idea? how to capture the DTMF digits to a file, after an extn asnwer? then POST it to a url? Regards, JR ___ 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
