[Asterisk-Users] problems compiling zaptel on FC5
Hi, I just updated to FC5, and used zaptel-1.2.1 as it was with my last version, it show this error: CC [M] /usr/local/src/zaptel/wcusb.o /usr/local/src/zaptel/wcusb.c:1452: error: unknown field owner specified in initializer /usr/local/src/zaptel/wcusb.c:1452: warning: initialization from incompatible pointer type make[2]: *** [/usr/local/src/zaptel/wcusb.o] Error 1 make[1]: *** [_module_/usr/local/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.15-1.2054_FC5-i686' make: *** [linux26] Error 2 Same thing happend while compiling zaptel-1.2.4 now, i just edited the Makefile that comes in zaptel directory to disable any usb, as i am not going to use any usb device in my asterisk, and it compiles and work ok. Regards, -=Raul=- -- Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.2.6/288 - Release Date: 3/22/2006 ___ --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] pulsedial on fxo signalling
Hi, On 1.0.9, an old handset with pulse disk was working and now with 1.2.2 is not working. I used pulsedial=yes on the specific channel, but it seems to be only for fxs signalling. How do i enable it on v1.2.2? Regards -=Raul=- ___ --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] FC4 + ztdummy + timming + trunking
Hi, Using a FC4 system without a digium card to centralize a few other systems with digium cards makes sound to break when zaptel module is loaded. Even if ztdummy is present. If i unload zaptel module, it works fine, but when i try to do a call with trunking, asterisk does not even answer. On those FC4 that has digium cards, everything works fine (trunking, timing, everything). So it makes me think that is something about RTC or timer handled by zaptel module. On a 2.4.x kernel, it works fine when i load the usb module as said in many support pages. But kernels 2.6.x should handle timing without the USB module i guess. any hint or idea on how to fix this problem? Regards -=Raul=- ___ --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] fc4 + iax + trunking
Hi, I m using asterisk on a system without a digium hardware, and when i try to use trunk=yes for my iax2 links, i just get this message while debugging: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 6ms SCall: 16384 DCall: 16384 [192.168.1.1:4569] AUTHMETHODS : 2 CHALLENGE : 753310823 USERNAME: 350001 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 16384 DCall: 16384 [192.168.1.1:4569] if i remove trunk=yes, it works, but i would like to run multiple channels at the same time. zaptel was compiled normal (make clean ; make linux26 ; make install) and ztdummy was loaded without problems (modprobe ztdummy) which also loads zaptel driver. Does anyone has this problem too and a hint to fix it? Regards, -=Raul=- ___ --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] trunk timing on 2.6.x
>Unless something about your setup is causing data corruption of some >kind, then I don't think the O/S or kernel has anything to do with this. > >It appears that one side is using CVS-HEAD, and sending trunktimestamps, >and the other side is using stable, which doesn't understand them. So, >you need to turn off trunktimestamps on the CVS-HEAD box. > >-SteveK Actually, both are downloaded from CVS using "cvs -r -v1-0 ..." but they show different versions 2.6.x: Connected to Asterisk CVS-v1-0-06/02/05-01:38:53 2.4.x: Connected to Asterisk CVS-HEAD-06/02/05-00:37:18 i just copied the source tree from the 2.4.x and compiled in the 2.6.x, and now it is working. Thanks for the hing SteveK regards, -=Raul=- ___ 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] trunk timing on 2.6.x
Two asterisk, one in a 2.4.x and another one in a 2.6.x are connection ok using IAX, before i upgraded one of them (the one with 2.6.x) Trunk was working ok. Since i upgraded the one that its now 2.6.x, i m getting this message: chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) ..and no sound comming from the 2.4.x while trunk=yes. It is a FC3, and zaptel was compiled sing "make linux26" and udev configured as mentioned in README.udev from zaptel directory. This fedora is working fine with a few sip hardphones and a TDM22B, which should handle the timing instead of using ztdummy, as i m doing it the other asterisk with 2.4.x. Haven't tried meetme yet, but at least with trunk=yes it is giving me that message constantly while both asterisk are bridged using IAX. Any suggestion? ___ 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] RSA question
Hi, Acording with http://www.voip-info.org/wiki-Asterisk+iax+rsa+auth, both sides should have "auth=rsa" in their respective section at iax.conf. But i've found that if server includes this option, the client keeps saying "No way to send secret to peer". My links with FWD and Iaxtel are working fine, this is a test i m doing with 2 asterisks i configured. Also, if i use "register => user:[name-of-public-key]:[EMAIL PROTECTED]" or "register => user:[EMAIL PROTECTED]", then i get a message from client saying "Asked to authenticate to xx.xx.xx.xx with an RSA key, but they don't allow RSA authentication", but link works anyway and calls both sides can be placed. In my case, i use both sides with "type=friend", so i can do incomming and outgoing calls, and when i do "iax2 show peers" at the client side, i get: Name/UsernameHost Mask Port Status myserver/userxx.xx.xx.xx (S) 255.255.255.255 4569 OK (1 ms) which means that is a (S)ecured password transaction. In both sides, "iax2 show users" shows "004" in the "Authen" column. Once the client authenticate with the server, the server does not need to re challenge the secence of user/secret with the client, because it already has the addres where to point the outgoing calls. My question is Is it still a secured password transaction even the server has no "auth=rsa"? For the server, even if no "inkeys" and "outkey" defined, everything works fine. .key and .pub files are created with astgenkey, and client only has the .pub file and it is specified in the "inkeys" at the clients context for the link with the server at iax.conf. A last question would be Is there anything else missing at the url which explains how to set up a RSA authentication? Cant make it work in that way. Regards... -=Raul=- ___ 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] unable to use addpac-ap200 (sip | h323)
Hi, I am testing an AP200 from addpac i m trying to make it register with Asterisk. It manages 3 protocols (sip, h323 and mgcp). If i use sip, i keep getting this messages: May 2 00:15:21 NOTICE[30545]: chan_sip.c:7711 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.202' May 2 00:15:21 NOTICE[30545]: chan_sip.c:7711 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.202' the config in the sip.conf is this: [108] type=friend host=dynamic username=addpac1 secret=test reinvite=no qualify=1000 canreinvite=no dtmfmode=rfc2833 context=toll-access callerid="addpac" <108> allow=all disallow=g723.1 [83] type=friend host=dynamic username=addpac1 secret=test reinvite=no qualify=1000 canreinvite=no dtmfmode=rfc2833 context=toll-access callerid="addpac" <108> allow=all disallow=g723.1 The addpac config for the sip is this: ! VoIP configuration. ! ! Voice service voip configuration. ! voice service voip busyout monitor gatekeeper busyout monitor voip-interface ! ! ! Voice port configuration. voice-port 0/0 caller-id enable caller-id type bellcore ! voice-port 0/1 caller-id enable caller-id type bellcore ! ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 83 port 0/0 ! dial-peer voice 1 pots destination-pattern 108 port 0/1 ! ! ! Voip peer configuration. ! dial-peer voice 1000 voip destination-pattern T session target sip-server session protocol sip dtmf-relay rtp-2833 ! ! ! ! Gateway configuration. ! ! gateway ! ! SIP UA configuration ! sip-ua sip-username 108 sip-password test sip-server 192.168.1.197 register e164 Thanks to vpp in [EMAIL PROTECTED], that said to commend the bind line in sip.conf, the extension 108 could register, but not the 83. And finnally i could make ring the extension, but after a minute or so, some kind of timeout happends and then i start getting the same "failed" lines again and i cant connect again. -- Using h323, from asterisk/channels/h323 of the cvs downloaded tree (i also downloaded the right versions of pwlib and openh323), it compiles with no problem (takes too much time to compile openh323, as usual). Using the default h323.conf.sample that comes in that directory, i found something strange. Again, i couldnt register configuring my ap200 with the h323 protocol. And doing a netstat i got this: [EMAIL PROTECTED] asterisk]# netstat -lpn | grep asterisk tcp0 0 0.0.0.0:50380.0.0.0:* LISTEN 31178/asterisk tcp0 0 0.0.0.0:17200.0.0.0:* LISTEN 31178/asterisk udp0 0 0.0.0.0:50600.0.0.0:* 31178/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 31178/asterisk udp0 0 192.168.1.197:2427 0.0.0.0:* 31178/asterisk There are missing the other default udp ports for h323 (1721, 1719, 1718). I tryed using netmeeting as well, but no susses. A tcpdump shows this: 192.168.1.200.1660 > 192.168.1.197.h323gatestat: udp 73 192.168.1.197 > 192.168.1.200: icmp: 192.168.1.197 udp port h323gatestat unreachable [tos 0xc0] 192.168.1.200.1660 > 192.168.1.197.h323gatestat: udp 73 192.168.1.197 > 192.168.1.200: icmp: 192.168.1.197 udp port h323gatestat unreachable [tos 0xc0] 192.168.1.197.ntp > 192.168.1.200.ntp: udp 48 (DF) [tos 0x10] 192.168.1.200.ntp > 192.168.1.197.ntp: udp 48 192.168.1.200.1660 > 192.168.1.197.h323gatestat: udp 73 192.168.1.197 > 192.168.1.200: icmp: 192.168.1.197 udp port h323gatestat unreachable [tos 0xc0] The addpac is not the problem in this case, as it can register in a gatekeepr as netmeeting does too. --- I am about to use MGCP, but need more documentation about it, i've never used it before, so i dont have a clue about it, but i'll try it too. Any hint would be apreciated Regards, -=Raul=- ___ 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] variable limit time on Dial
Hi, Is there a way to set a variable time on the execution of the Dial command in an AGI script? The L option uses fixed times for warnings. regards, -=Raul=- ___ 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] fax autoanswer
Hi Adam, Thanks for anwer. Actually, i got an answer. Lemme show the full process... ;this is where my zap/3 context points to [incoming] exten => s,1,Answer exten => s,2,wait(1) exten => s,3,Goto(bienvenida,s,1) exten => t,1,Hangup include => extensions [bienvenida] include => daytime|12:00-19:00 include => morningtime|00:00-11:59 include => nighttime|19:01-23:59 include => menu ;all other contexts ends comming back to [opciones] ;so there is an "Answer" before comming to [menu] or [opciones] [menu] exten => s,1,wait(1) exten => s,2,Background(new/estas) exten => s,3,Background(new/opciones) exten => 1,1,Goto(explicacion,s,1) exten => 2,1,Goto(extensions,101,1) exten => 3,1,Goto(acuari,s,1) exten => 4,1,goto(wizardteam,s,1) exten => 5,1,Goto(fechayhora,s,1) exten => 0,1,rxfax(/tmp/sample.tif) exten => fax,3,rxfax(/tmp/sample.tif) [opciones] exten => s,1,wait(1) exten => s,2,Background(new/opciones) exten => 1,1,Goto(explicacion,s,1) exten => 2,1,Goto(extensions,101,1) exten => 3,1,Goto(acuari,s,1) exten => 4,1,goto(wizardteam,s,1) exten => 5,1,Goto(fechayhora,s,1) exten => fax,1,rxfax(/tmp/sample.tif) > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Adam > Goryachev > Sent: Monday, September 20, 2004 8:01 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] fax autoanswer > > > On Tue, 2004-09-21 at 11:52, Raul Elizondo (wizardteam) wrote: > > Hi, > > > > I am a little bit confused about how extensions.conf recognice > a fax tone. > > For testing i got this in my home pbx: > > > > [opciones] > > exten => s,1,wait(1) > > exten => s,2,Background(new/opciones) > > exten => 1,1,Goto(explicacion,s,1) > > exten => 2,1,Goto(extensions,101,1) > > exten => 3,1,Goto(acuari,s,1) > > exten => 4,1,goto(wizardteam,s,1) > > exten => 5,1,Goto(fechayhora,s,1) > > exten => fax,1,rxfax(/tmp/sample.tif) > > > > but if someone calls from a fax, then the * should inmediatly > recognize it > > and send it to the rzfax app, right? > > > > Any hint? > > Answer the call first. "show application answer" at the asterisk CLI > > Regards, > Adam > > > ___ > 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 ___ 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] fax autoanswer
Hi, I am a little bit confused about how extensions.conf recognice a fax tone. For testing i got this in my home pbx: [opciones] exten => s,1,wait(1) exten => s,2,Background(new/opciones) exten => 1,1,Goto(explicacion,s,1) exten => 2,1,Goto(extensions,101,1) exten => 3,1,Goto(acuari,s,1) exten => 4,1,goto(wizardteam,s,1) exten => 5,1,Goto(fechayhora,s,1) exten => fax,1,rxfax(/tmp/sample.tif) but if someone calls from a fax, then the * should inmediatly recognize it and send it to the rzfax app, right? Any hint? regards, - Raul Elizondo FWD# 486533 ___ 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
[Asterisk-Users] fax autoanswer
-o-o-o-o-o-o- Raul Elizondo http://www.acuari.com FWD# 486533 ___ 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] IAX to IAX connect question
and the answer for my own question is:
using "notransfer=yes" on the iax.conf contexts
-
Raul Elizondo
FWD: 486533
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Raul
> Elizondo (wizardteam)
> Sent: Thursday, September 16, 2004 5:24 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] IAX to IAX connect question
>
>
> I think i got the solution for what i was planing to set. Here is a
> ontheway sample (not what i got but its about the same)
>
> Office iax.conf
> ---
> register => 123456:[EMAIL PROTECTED]
>
> jitterbuffer=no
> tos=lowdelay
>
> [iaxfwd]
> type=user
> context=fromiaxfwd
> auth=rsa
> inkeys=freeworlddialup
> diallow=all
> allow=ulaw
>
> [myofficename]
> type=peer
> host=dynamic
> auth=rsa
> outkeys=myrsa
> username=myofficename
> context=somecontext
>
> [user01]
> type=friend
> user=user01
> host=dynamic
> secret=somepass01
> username=user01
> context=accesslevel01
>
> [user02]
> type=friend
> user=user02
> host=dynamic
> secret=somepass01
> username=user01
> context=accesslevel01
>
> Office extensions.conf
> --
> [general]
> static=yes
> writeprotect=no
>
> [globals]
> MYUSER01=IAX2/myofficename:[EMAIL PROTECTED]
> MYUSER02=IAX2/myofficename:[EMAIL PROTECTED]
> MYOFFICENAMECID="Some name"
> MYFWDUP=IAX2/123456:[EMAIL PROTECTED]
>
> [extensions]
> ; set of extensions
> ; for testing like echotest and others
> ; or whatever else needed
>
> [fromiaxfwd]
> exten => 123456,1,Answer
> exten => 123456,2,Dial(${MYUSER01}&${MYUSER02},60,r)
> exten => 123456,3,Hangup
>
> [toiaxfwd]
> exten => _8.,1,SetCallerId,${MYOFFICENAMECID}
> exten => _8.,2,Dial(${MYFWDUP}/${EXTEN:1},60,r)
> exten => _8.,3,Congestion
>
> [accesslevel01]
> include => extensions
> ignorepat => 8
> include => toiaxfwd
>
> User01 iax.conf
> ---
> register => user01:[EMAIL PROTECTED]
>
> [myofficename]
> type=user
> context=fromoffice
> auth=rsa
> inkeys=myrsa
>
> User01 extensions.conf
> --
> [globals]
> MYOFFICE=IAX2/user01:[EMAIL PROTECTED]
> FWDCIDNAME="My name01"
>
> [extensions]
> ; my local extensions
>
> [fromoffice]
> exten => s,1,goto(extensions,101,1) ; where the zap/1 is located
>
> [toiaxfwd]
> exten => _8.,1,SetCallerId,${FWDCIDNAME}
> exten => _8.,1,Dial(${MYOFFICE}/${EXTEN},60,r)
> exten => _8.,2,Congestion
>
> [localaccess]
> ; set of local pstn access
>
> [dialaccess]
> ; where zap/* or local sip phones should point
> include => extensions
> ignorepat => 8
> include => toiaxfwd
> ignorepat => 9
> include => localaccess
>
> User02 iax.conf
> ---
> register => user02:[EMAIL PROTECTED]
>
> [myofficename]
> type=user
> context=fromoffice
> auth=rsa
> inkeys=myrsa
>
> User02 extensions.conf
> --
> [globals]
> MYOFFICE=IAX2/user02:[EMAIL PROTECTED]
> FWDCIDNAME="My name02"
>
> [extensions]
> ; my local extensions
>
> [fromoffice]
> exten => s,1,goto(extensions,201,1) ; where the zap/1 or sip is located
>
> [toiaxfwd]
> exten => _8.,1,SetCallerId,${FWDCIDNAME}
> exten => _8.,1,Dial(${MYOFFICE}/${EXTEN},60,r)
> exten => _8.,2,Congestion
>
> [localaccess]
> ; set of local pstn access
>
> [dialaccess]
> ; where zap/* or local sip phones should point
> include => extensions
> ignorepat => 8
> include => toiaxfwd
> ignorepat => 9
> include => localaccess
>
>
> So, in this way, i can keep adding users in the office using only one
> context for each user with its own user/pass for validation.
>
> Now, here it comes another thing. When i call from user01 (or
> home) to FWD,
> as soon as it answer it hangsup. There was just a couple times i could do
> the FWD echotest or the 411, but not anymore but incoming calls
> from FWD and
> from office works fine. Does anyone see something wrong?
>
> Regards,
>
>
> Raul Elizondo
> FWD# 486533
>
> ___
> 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
___
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RE: [Asterisk-Users] IAX to IAX connect question
I think i got the solution for what i was planing to set. Here is a
ontheway sample (not what i got but its about the same)
Office iax.conf
---
register => 123456:[EMAIL PROTECTED]
jitterbuffer=no
tos=lowdelay
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
diallow=all
allow=ulaw
[myofficename]
type=peer
host=dynamic
auth=rsa
outkeys=myrsa
username=myofficename
context=somecontext
[user01]
type=friend
user=user01
host=dynamic
secret=somepass01
username=user01
context=accesslevel01
[user02]
type=friend
user=user02
host=dynamic
secret=somepass01
username=user01
context=accesslevel01
Office extensions.conf
--
[general]
static=yes
writeprotect=no
[globals]
MYUSER01=IAX2/myofficename:[EMAIL PROTECTED]
MYUSER02=IAX2/myofficename:[EMAIL PROTECTED]
MYOFFICENAMECID="Some name"
MYFWDUP=IAX2/123456:[EMAIL PROTECTED]
[extensions]
; set of extensions
; for testing like echotest and others
; or whatever else needed
[fromiaxfwd]
exten => 123456,1,Answer
exten => 123456,2,Dial(${MYUSER01}&${MYUSER02},60,r)
exten => 123456,3,Hangup
[toiaxfwd]
exten => _8.,1,SetCallerId,${MYOFFICENAMECID}
exten => _8.,2,Dial(${MYFWDUP}/${EXTEN:1},60,r)
exten => _8.,3,Congestion
[accesslevel01]
include => extensions
ignorepat => 8
include => toiaxfwd
User01 iax.conf
---
register => user01:[EMAIL PROTECTED]
[myofficename]
type=user
context=fromoffice
auth=rsa
inkeys=myrsa
User01 extensions.conf
--
[globals]
MYOFFICE=IAX2/user01:[EMAIL PROTECTED]
FWDCIDNAME="My name01"
[extensions]
; my local extensions
[fromoffice]
exten => s,1,goto(extensions,101,1) ; where the zap/1 is located
[toiaxfwd]
exten => _8.,1,SetCallerId,${FWDCIDNAME}
exten => _8.,1,Dial(${MYOFFICE}/${EXTEN},60,r)
exten => _8.,2,Congestion
[localaccess]
; set of local pstn access
[dialaccess]
; where zap/* or local sip phones should point
include => extensions
ignorepat => 8
include => toiaxfwd
ignorepat => 9
include => localaccess
User02 iax.conf
---
register => user02:[EMAIL PROTECTED]
[myofficename]
type=user
context=fromoffice
auth=rsa
inkeys=myrsa
User02 extensions.conf
--
[globals]
MYOFFICE=IAX2/user02:[EMAIL PROTECTED]
FWDCIDNAME="My name02"
[extensions]
; my local extensions
[fromoffice]
exten => s,1,goto(extensions,201,1) ; where the zap/1 or sip is located
[toiaxfwd]
exten => _8.,1,SetCallerId,${FWDCIDNAME}
exten => _8.,1,Dial(${MYOFFICE}/${EXTEN},60,r)
exten => _8.,2,Congestion
[localaccess]
; set of local pstn access
[dialaccess]
; where zap/* or local sip phones should point
include => extensions
ignorepat => 8
include => toiaxfwd
ignorepat => 9
include => localaccess
So, in this way, i can keep adding users in the office using only one
context for each user with its own user/pass for validation.
Now, here it comes another thing. When i call from user01 (or home) to FWD,
as soon as it answer it hangsup. There was just a couple times i could do
the FWD echotest or the 411, but not anymore but incoming calls from FWD and
from office works fine. Does anyone see something wrong?
Regards,
Raul Elizondo
FWD# 486533
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RE: [Asterisk-Users] IAX to IAX connect question
Hi Benjamin,
Thanks for answering, now i got some other questions.
Acording to http://www.voip-info.org/wiki-Asterisk+iax+rsa+auth, and my
understanding of peer and user, [FWD-service] in the sample you provide me
should be type=peer as it will be the master/server, and [FWD-gw] should be
type=user as it is a client for FWD service.
Now, lets think that office is FWD, and home is a common client of FWD, so
the question is simple. If i set:
Home:iax.conf
register => user:[EMAIL PROTECTED]
[office]
type=user
host=dynoffice.tld
context=fromoffice
auth=rsa
inkeys=myastkey
Home:extensions.conf
[extensions]
exten => 101,1,Dial(Zap/1,20)
exten => 101,2,Voicemail(u${EXTEN})
exten => 101,3,Hangup
exten => 101,102,Voicemail(b${EXTEN}}
exten => 101,103,Hangup
[fromoffice]
exten => s,1,goto(extensions,101,1)
The question is: What would i need to set in office in order to receive
calls from FWD or another service?
-=Raul=-
> Office:/etc/asterisk/iax.conf ...
>
> [FWD-service]
> type=user ; we are letting a remote user use this server to call FWD
> username=rfwduser ; their username with us here
> host=dynamic ; their host may not have a fixed ip address
> context=fwd-service
>
> [FWD-gw] ; outbound connections to FWD from here
> type=peer
> username=12345
> host=iax2.fwdnet.net
>
>
> Home:/etc/asterisk/iax.conf ...
>
> [FWD-gw]
> type=peer ; we are using the remote office server as a gateway to call
out
> username=rfwduser ; our username with the remote office server
> host=ip-or-dns ; the ip address or dns name of the remote office server
>
> Office:/etc/asterisk/extensions.conf ...
>
> [globals]
> FWDUSERID=12345
> FWDUSERNAME=Fred Flintstone Inc
> FWDGW=IAX2/[EMAIL PROTECTED] ; this is FWD's IAX server
>
> [fwd-service]
> ; we provide this context for remote users calling FWD through us
> exten => _X.,1,SetCallerID(${FWDUSERID})
> exten => _X.,2,SetCIDName(${FWDUSERNAME})
> exten => _X.,3,Dial(${FWDGW}/${EXTEN},60,r)
> exten => _X.,4,Hangup
>
> Home:/etc/asterisk/extensions.conf ...
>
> [globals]
> FWDGW=IAX2/[EMAIL PROTECTED] ; this is our office server acting as a gateway
>
> [fwd-service]
> ; we use the remote server at the office to call FWD
> exten => _X.,1,Dial($FWDGW}/${EXTEN},60,r)
> exten => _X.,2,Hangup
> ;
> ; Don't forget to include this context for anybody who is supposed to use
it
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[Asterisk-Users] IAX to IAX connect question
Hi,
I got my * working fine with FWD at office with 2 extensions, i receive
calls and i can make calls thru FWD. I got also my * at home, and i
connected it using auth=rsa. From my home, i can make calls using my office
iax, but if i try to redirect incomming calls from FWD to my * at home, it
rejects the call. I created the pub/key pairs for rsa and its working ok
and i just pasted the sections/contexts involved.
Must of the samples around the internet works with double config in iax.conf
for each server, a peer and a user, i've done that in a VPN and that works
fine. But... imagine setting a new *, i would need to modify the first 2
besides that adding them in the new one. What about a 4th or a 5th? Every
new * in the pbx means to modify all other *. So what i thougth was to set
a main one with some of slaves or users, in this way, i will only need to
add new "slaves" in the master, and this is what i tried.
Office iax.conf
===
register => 99:[EMAIL PROTECTED]
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
[rsaauth]
type=peer
host=dynamic
auth=rsa
outkeys=myast
username=rsaauth
context=extensions
[user01]
type=peer
user=user01
host=dynamic
secret=somepass
username=user01
context=localuse
outkeys=myast
inkeys=myast
Office extensions.conf
==
[officetime]
exten => _99,1,Answer
exten => _99,2,wait(1)
exten => _99,3,Dial(Zap/1,20,tr)
exten => _99,4,Voicemail,u101
exten => _99,102,Voicemail,b101
exten => _99,105,Hangup
[noofficetime]
exten => _99,1,Answer
exten => _99,2,wait(1)
exten => _99,3,Dial(IAX2/user01/${EXTEN})
exten => _99,4,Voicemail,u101
exten => _99,102,Voicemail,b101
exten => _99,105,Hangup
[fromiaxfwd]
include => officetime|09:00-17:30
include => noofficetime|17:30-23:59
include => noofficetime|00:00-08:59
[toiaxfwd]
exten => _8.,1,SetCallerId,"MyName"
exten => _8.,2,Dial(IAX2/99:[EMAIL PROTECTED]/${EXTEN:1},60,r)
exten => _8.,3,Congestion
[localuse]
include => extensions
include => toiaxfwd
Home iax.conf
=
register => user01:[EMAIL PROTECTED]
[rsaauth]
type=user
context=fromoffice
auth=rsa
inkeys=myast
allow=gsm
Home extensions.conf
[fromoffice]
exten => s,1,Dial(Zap/1,20)
exten => s,2,Voicemail(u101)
exten => s,3,Hangup
exten => s,102,Voicemail(b101)
exten => s,103,Hangup
[tooffice]
exten => _7.,1,SetCallerId,"MyName"
exten => _7.,2,Dial(IAX2/user01:[EMAIL PROTECTED]/${EXTEN},60,r)
exten => _7.,3,Congestion
Everything from my home to the office works fine, even the FWD calls. RSA
auth is working without problem. The problem comes when i try out of office
hours. My home * just refuses the calls. I've tried with switch instead of
exten, but even in the samples it is kinda confusing.
Any hint or help with this? (i believe it would be only one line that is not
right "Dial(IAX2/user01/${EXTEN})")
Regards,
-=Raul=-
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[Asterisk-Users] sip does not bind all addreses
Hi, My linuxbox has 2 eth's, one with pppoe for dsl, and also i got an ip_gre tunnel. At the time i run asterisk, even i got bindaddr=0.0.0.0, it does not show any port open for sip (5060), if i change 0.0.0.0 for any ip, next time i reload, it opens the specific ip, changing back to 0.0.0.0 and reloading, it keeps the same ip open. The point is that i cant open all ips i need at the same time. Why? Regards, -=Raul=- ___ 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
[Asterisk-Users] sip does not bind all addreses
Hi, My linuxbox has 2 eth's, one with pppoe for dsl, and also i got an ip_gre tunnel. At the time i run asterisk, even i got bindaddr=0.0.0.0, it does not show any port open for sip (5060), if i change 0.0.0.0 for any ip, next time i reload, it opens the specific ip, changing back to 0.0.0.0 and reloading, it keeps the same ip open. The point is that i cant open all ips i need at the same time. Why? Regards, -=Raul=- ___ 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
[Asterisk-Users] R2MFC cards
Hi guys, I've been searching for a R2MFC E1 card that works with asterisk. As far as i could find, i got the Dialogic DTI/301SC card. Is there a way to make it work with asterisk or anyone else can recomend me another brand that actually works with asterisk? Regards -=Raul=- ___ 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 dtmf problems (with x-lite) (solved)
Hi Steven, >IT is interesting you are even that far along with your AGI application >when you haven't even figured out your mail client. Do you mean my email client? or my voicemail? Voicemail is working fine on digim extensions, i even changed the language to spanish (btw, there are lot of mistakes in the manner to say numbers). The problem comes when i try to use an agi with a x-lite client (softphone), it works fine with dtmf over voicemail and over menu selection, but it does not recognice dtmf for agi's. >What was the connection to the message about voice recognition that you >replied to? I am not using voice recognition, i m trying to use dtmf recognition with agi's, but somehow, asterisk does not listen the x-lite only with agi's. The rest of the digium fxs extensions works fine with my agi. >If you wish to be lazy, go ALL OUT, and learn that you can start a new >message with a new thread by clicking on the address of the mailing list >in the headers section. It will bring up a new message with no content >in it. It is a lot easier than deleting all the text in the old message. huh? >BTW, did you ANSWER() the call before going to AGI? Actually no, and that was my problem. When i begun to work with agi in tcl, i was figuring out how it works from a perl script and i got some problems, then, i used Answer() first times, i thought it was the error, when i debuged my script, it worked fine with out the Answer() for digium extensions. Now is working after i added Answer() on the extensions.conf for that. Thanks for the hint! ;) -=Raul=- ___ 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
[Asterisk-Users] AGI dtmf problems (with x-lite)
Hi guys, May be this is a subject already disscused some where before, but i cant find a solution. By using x-lite, i can dial in menus, or even in voicemail during the process of any of both. But when i run my own AGI (using tcl), it does not detect DTMF when "GET DATA" function. I got no more than a month that i downloaded asterisk via cvs, is this problem already fixed in the mean time? or is there something else i have to set in my AGI? Regards, -=Raul=- ___ 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
[Asterisk-Users] iax to iax link
Hi guys, I got 2 different linux boxex each one with a TDM22B (2 fxs and 2fxo). Bot linuxes are connected to the same dsl company with a dynamic ip and both are doing an ip_gre tunnel VPN. I could work with zapta.conf zapata.conf and extensions.conf locally on each linux. Each linux is connected to an analog panasonic PBX with 2 lines and 2 extensions and the thing is to call linuxB panasonic extensions from the linuxA lines and viceversa. I am new on asterisk, but have experience with linux and openh323gk, and all i need is a sample to start understanding that about iax and sip connections. Both linux will have a permanent thru the vpn, so that wont be a problem neither. Any hint/help? -=Raul=- ___ 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] phonejack and linejack in the same system
This is as far as i could go... I readed about the archives for "ztdummy", and made the fix and recompiled after using ./configure --disable-isa-pnp, i edited the phone.conf and extensions.conf, but i still get this messages: [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 03:52:44 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 03:52:44 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 03:52:44 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 03:52:44 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! [EMAIL PROTECTED] asterisk]# ls /dev/phone0 /dev/phone0 I am going to compile openh323 just to make sure both quicknet cards work in this box, but it was working in another box with redhat9. -Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Raul Elizondo (wizardteam) Sent: Sunday, May 02, 2004 3:28 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] phonejack and linejack in the same system i actually found a couple things interesting in the archives after i wrote my first email, and i deleted the soundcard, but i still got problems with the linejack, i m testing all possible options. -=Raul=- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Sunday, May 02, 2004 1:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] phonejack and linejack in the same system I'm a newbie too -- search the archives for "ztdummy". -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raul Elizondo (wizardteam) Sent: Sunday, May 02, 2004 2:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] phonejack and linejack in the same system Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this error: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console Channel Driver) May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 01:16:42 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 01:16:43 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! this does not happend and asterisk runs ok and gives a CLI> prompt if i comment back these lines. Any hint? Regards, -=Raul=- ___ 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 ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/
RE: [Asterisk-Users] phonejack and linejack in the same system
i actually found a couple things interesting in the archives after i wrote my first email, and i deleted the soundcard, but i still got problems with the linejack, i m testing all possible options. -=Raul=- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Sunday, May 02, 2004 1:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] phonejack and linejack in the same system I'm a newbie too -- search the archives for "ztdummy". -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raul Elizondo (wizardteam) Sent: Sunday, May 02, 2004 2:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] phonejack and linejack in the same system Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this error: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console Channel Driver) May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 01:16:42 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 01:16:43 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! this does not happend and asterisk runs ok and gives a CLI> prompt if i comment back these lines. Any hint? Regards, -=Raul=- ___ 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 ___ 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 ___ 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
[Asterisk-Users] phonejack and linejack in the same system
Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this error: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) May 2 01:16:42 WARNING[16384]: chan_iax2.c:6789 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console Channel Driver) May 2 01:16:42 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp: No such device == No sound card detected -- console channel will be unavailable == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found May 2 01:16:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open '/dev/phone0' May 2 01:16:42 ERROR[16384]: chan_phone.c:1141 load_module: Unable to register channel '/dev/phone0' == Unregistered channel type 'Phone' May 2 01:16:42 WARNING[16384]: loader.c:326 ast_load_resource: chan_phone.so: load_module failed, returning -1 == Unregistered channel type 'Phone' May 2 01:16:43 WARNING[16384]: loader.c:421 load_modules: Loading module chan_phone.so failed! this does not happend and asterisk runs ok and gives a CLI> prompt if i comment back these lines. Any hint? Regards, -=Raul=- ___ 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
