[Asterisk-Users] Zyxel 2000W (WI-FI) Problems
Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fineif Icall the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" 205 host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729 dtmfmode=rfc2833 Sip debug: headers, 0 lines Retransmitting #4 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=C8355813679C716AFCA To: sip:[EMAIL PROTECTED];tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 to 60.64.250.254:5060 Retransmitting #5 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=C8355813679C716AFCA To: sip:[EMAIL PROTECTED];tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 ___ 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] ztdummy and Debian
Hi ! What is the easiest esyest way for implementation of ztdummy on a Debian (testing) system? Thore ___ 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] ztdummy and Debian
Hi ! I this working with kernel 2.4? Thore - Original Message - From: Samuel T. Cossette [EMAIL PROTECTED] To: Thore [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 24, 2005 1:45 PM Subject: Re: [Asterisk-Users] ztdummy and Debian Hi, This is how I got ztdummy on debian sarge: $ apt-get install kernel-headers-2.6.8-2-386 dpatch kernel-package zaptel zaptel-source $ cd /usr/src $ ln -s kernel-headers-2.6.8-2-386/ linux $ cd linux $ make-kpkg modules_image $ dpkg -i ../zaptel-modules-2.6.8-2-386_1.0.7-3+10.00.Custom_i386.deb Selecting previously deselected package zaptel-modules-2.6.8-2-386. (Reading database ... 51551 files and directories currently installed.) Unpacking zaptel-modules-2.6.8-2-386 (from .../zaptel-modules-2.6.8-2-386_1.0.7-3+10.00.Custom_i386.deb) ... Setting up zaptel-modules-2.6.8-2-386 (1.0.7-3+10.00.Custom) ... $ depmod -a $ modprobe ztdummy $ dmesg look at this (Les plus / Kit Zaptel) http://terminaux.levinux.org/wakka.php?wiki=LaTelephonie bye, samuel Hi ! What is the easiest esyest way for implementation of ztdummy on a Debian (testing) system? Thore ___ 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 Samuel T. Cossette 1.418.8o2.784o ___ 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] call forwarding and parking
Hi !
What is wrong with my dial plan?
I can't get my call forwarding and parking to work.
Do I need to edit more config files?
Thore
extensions.conf :
[general]
static=yes
writeprotect=no
[macro-dialout]
; ${ARG1} CIDNAME
; ${ARG2} Device
; ${ARG3} Num
; ${ARG4} SIP EXT
exten = s,1,SetCIDName(${ARG1})
exten = s,2,Dial(${ARG2}${ARG3}${ARG4},,t)
exten = s,3,Playback(invalid)
exten = s,4,Hangup
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto
102
exten=s,2,Dial(Local/[EMAIL PROTECTED]/n) ; Unconditional forward
exten=s,3,Dial(${ARG2},20) ; 20sec timeout
exten=s,4,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto
105
exten=s,5,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on busy or unavailable
[globals]
[apps]
; Unconditional Call Forward
exten = _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten = _*21*X.,2,Hangup
exten = #21#,1,DBdel(CFIM/${CALLERIDNUM})
exten = #21#,2,Hangup
; Call Forward on Busy or Unavailable
exten = _*61*X.,1,DBput(CFBS/${CALLERIDNUM}=${EXTEN:4})
exten = _*61*X.,2,Hangup
exten = #61#,1,DBdel(CFBS/${CALLERIDNUM})
exten = #61#,2,Hangup
[iconnect]
exten = _47XXX.,1,Dial(Sip/iconnect/${EXTEN},120,t)
exten = _1XXX.,1,Dial(Sip/iconnect/${EXTEN},120,t)
[outgoing-40]
include = apps
include = parkedcalls
exten = _820X,1,Hangup
exten = _,1,Dial(Sip/33297540/${EXTEN},120,t)
exten = _,2,Congestion
exten = _820X,2,Congestion
[outgoing-45]
include = apps
include = parkedcalls
exten = _820X,1,Hangup
exten = _,1,Dial(Sip/voip/${EXTEN},120,t)
exten = _,2,Congestion
exten = _820X,2,Congestion
[local]
include = apps
include = parkedcalls
exten = 101,1,Dial(Sip/101,120)
exten = 102,1,Dial(Sip/102,120)
exten = 201,1,Dial(Sip/201,120)
[dialout-40]
include = outgoing-40
include = local
include = apps
include = parkedcalls
include = iconnect
[dialout-45]
include = outgoing-45
include = local
include = apps
include = parkedcalls
features.conf:
[general]
parkext = 700
parkpos = 701-720
context = parkedcalls
parkingtime = 60
;transferdigittimeout = 3
;courtesytone = beep
adsipark = yes
pickupexten = *8
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[Asterisk-Users] call forwardin and parking
Hi !Vat is wrong with my dial plan?I cant
get my call forwarding and parking to work LDo I need to edit moor config
files?
Thore
extensions.conf :
[general]static=yeswriteprotect=no
[macro-dialout]; ${ARG1} CIDNAME; ${ARG2} Device; ${ARG3}
Num; ${ARG4} SIP EXTexten = s,1,SetCIDName(${ARG1})exten =
s,2,Dial(${ARG2}${ARG3}${ARG4},,t)exten = s,3,Playback(invalid)exten
= s,4,Hangup
[macro-stdexten];; Standard extension macro (with
call forwarding):; ${ARG1} - Extension(we could have used
${MACRO_EXTEN} here as well; ${ARG2} - Device(s) to
ring;exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if
not existing, goto 102exten=s,2,Dial(Local/[EMAIL PROTECTED]/n) ;
Unconditional forwardexten=s,3,Dial(${ARG2},20) ; 20sec
timeoutexten=s,4,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not
existing, goto 105exten=s,5,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on busy or
unavailable
[globals]
[apps]; Unconditional Call Forwardexten =
_*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})exten =
_*21*X.,2,Hangupexten =
#21#,1,DBdel(CFIM/${CALLERIDNUM})exten = #21#,2,Hangup
; Call Forward on Busy or Unavailableexten =
_*61*X.,1,DBput(CFBS/${CALLERIDNUM}=${EXTEN:4})exten =
_*61*X.,2,Hangupexten =
#61#,1,DBdel(CFBS/${CALLERIDNUM})exten = #61#,2,Hangup
[iconnect]exten =
_47XXX.,1,Dial(Sip/iconnect/${EXTEN},120,t)exten =
_1XXX.,1,Dial(Sip/iconnect/${EXTEN},120,t)
[outgoing-40]include = appsinclude = parkedcalls
exten = _820X,1,Hangupexten =
_,1,Dial(Sip/33297540/${EXTEN},120,t)exten =
_,2,Congestionexten = _820X,2,Congestion
[outgoing-45]include = appsinclude = parkedcalls
exten = _820X,1,Hangupexten =
_,1,Dial(Sip/voip/${EXTEN},120,t)exten =
_,2,Congestionexten = _820X,2,Congestion
[local]include = appsinclude = parkedcallsexten =
101,1,Dial(Sip/101,120)exten = 102,1,Dial(Sip/102,120)exten =
201,1,Dial(Sip/201,120)
[dialout-40]include = outgoing-40include = localinclude
= appsinclude = parkedcallsinclude = iconnect
[dialout-45]include = outgoing-45include = localinclude
= appsinclude = parkedcalls
features.conf:
[general]parkext =
700
parkpos =
701-720
context =
parkedcalls
parkingtime =
60
;transferdigittimeout = 3
;courtesytone =
beep
adsipark =
yes
pickupexten = *8
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Re: [Asterisk-Users] Sip registration Problems With Zyxel P2000W
Hi I have a Zyxel P2002 (ATA) with this config. Registration works but i cant call inn. Outgoing works fine. Any clue? Thore - Original Message - From: Paul Dracevich [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, April 03, 2005 6:51 AM Subject: RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi ya I have also three of these phone, here is my entry in my sip.conf [4701721] type=friend username=4701721 secret=password721 host=dynamic canreinvite=no context=internal disallow=all allow=g729 dtmfmode=rfc2833 qualify=4 permit=0.0.0.0/0.0.0.0 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ugur GUNCER Sent: Sunday, 3 April 2005 4:37 p.m. To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi all, I bougth zyxel wifi phone but i cant register when i want to register phone to asterisk i recieve These errors I spend 6 hours to fix regist problem but i cant find the solution [9875] type=friend username=9875 secret=5789 host=dynamic context=default callerid=Ugur Guncer 9875 canreinvite=no dtmfmode=rfc2833 nat=no Sip read: REGISTER sip:213.139.225.82:5060 SIP/2.0 Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Contact: sip:[EMAIL PROTECTED]:43956;transport=udp Expires: 300 Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 85.99.110.143 : 43956 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 85.99.110.143:43956 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 From: sip:[EMAIL PROTECTED];user=phone;tag=5175B05114E474A31693 To: sip:[EMAIL PROTECTED];user=phone;tag=as369f8960 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0f3403ce Content-Length: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call forwarding
Hi! I need a sample dilplan with call forwarding This did not help me to get it work: http://www.voip-info.org/wiki-Asterisk+call+forwarding Thore ___ 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] Zyxel Prestige 2002 (ATA)
Hi ! I cant get my Zyxel Prestige 2002 (ATA) to answer the phone. Outgoing calls i working perfect, but i get no incoming calls. Everything sems normal on Asterix This is my setup for P2002 (sip.conf): [203] type=friend username=203 secret=302 callerid=Office 203 203 host=dynamic context=dialout nat=yes canreinvite=no disallow=all allow=ulaw allow=alaw Thore ___ 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] sip provider
Hi
I have a
voip provider use sip. To telephones with exten. 201 and 202.
My voip
providergive me thisnumbers 33297540 and 33297545.
Is it
possible to get exten 201 to ring out on 33297540 and 202 - 33297545
?
This is my config now :
sip.conf: [general]
context=default realm=olsen.com port=5060 bindaddr=0.0.0.0
srvlookup=yes ; fjern ";" fra følgende hvis Asterisk er bag NAT og har
statisk IP: ;externip=1.2.3.4 ; erstat med din statiske IP adresse
register =33297540:[EMAIL PROTECTED]/33297540
register = 33297545:[EMAIL PROTECTED]/33297545 ; Voip [33297540]
type=friend host=voip.dk dtmfmode=rfc2833 canreinvite=no
username=33297540 secret=nisse context=voip_incoming nat=yes
fromuser=33297540 fromdomain=voip.dk insecure=very
[33297545]
type=friend host=voip.dk dtmfmode=rfc2833 canreinvite=no
username=33297545secret=nisse context=voip_incoming nat=yes
fromuser=33297545 fromdomain=voip.dk insecure=very; SPA-2000
Line 1 [201] type=friend host=dynamic context=dialout
username=201 secret=spapassword callerid="Thore" 201
nat=
[202] type=friend
host=dynamic context=dialout username=202secret=spapassword
callerid="Tom" 202 nat=no ; extensions.conf:
[general] static=yes writeprotect=no [globals]
[voip_outgoing] exten = _X.,1,Dial(Sip/voip/${EXTEN},120)
exten = _X.,2,Congestion [dialout] include =
voip_outgoing [voip_incoming] exten =
33297540,1,Dial(Sip/201,120) exten = 33297540,2,Congestion
exten =
33297545,1,Dial(Sip/202,120) exten = 33297545,2,Congestion
Thore
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Re: [Asterisk-Users] sip provider
This work well for incoming calls, but not for outgoing call. Those i call get the wrong number in the display. Thore - Original Message - From: administrator tootai [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 27, 2005 6:19 PM Subject: Re: [Asterisk-Users] sip provider Thore a écrit : Hi I have a voip provider use sip. To telephones with exten. 201 and 202. My voip provider give me this numbers 33297540 and 33297545. Is it possible to get exten 201 to ring out on 33297540 and 202 - 33297545 ? [...] ; Voip [33297540] type=friend host=voip.dk dtmfmode=rfc2833 canreinvite=no username=33297540 secret=nisse context=voip_incoming context=voip_incoming-phone1 nat=yes fromuser=33297540 fromdomain=voip.dk insecure=very [33297545] type=friend host=voip.dk dtmfmode=rfc2833 canreinvite=no username=33297545 secret=nisse context=voip_incoming context=voip_incoming-phone2 nat=yes fromuser=33297545 fromdomain=voip.dk insecure=very [...] [voip_incoming] exten = 33297540,1,Dial(Sip/201,120) exten = 33297540,2,Congestion exten = 33297545,1,Dial(Sip/202,120) exten = 33297545,2,Congestion [voip_incoming-phone1] exten = 33297540,1,Dial(Sip/201,120) exten = 33297540,2,Congestion [voip_incoming-phone2] exten = 33297545,1,Dial(Sip/202,120) exten = 33297545,2,Congestion -- Daniel ___ 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] Micronet SP5001 ATA
Does anyone here have any experience with the Micronet SP5001 ATA and Asterisk ? Some sip.conf samples will help a lot. Thore ___ 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
