[asterisk-users] Calls from PSTN on SPA3102
Hello list, this is my first post on this list.
I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones.
I have configured the SPA PSTN line as trunk to receive and send
calls.
I can call outside from SIP phone throw the PSTN line and all is OK,
the problem is when I receive a call from the PSTN, on the out caller
phone there is a demo playback. I want to redirect the call to a
internal SIP phone.
This is the extensions.conf:
[spa]
include = saliente_pstn
include = entradas_pstn
include = sips
[saliente_pstn]
exten = _9ZXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
exten = _9ZXXX,n,Hangup
[entradas_pstn]
exten = s,1,Dial(SIP/103,20,tm)
exten = s,2,VoiceMail(103)
exten = s,3,Hangup
[sips]
exten = 100,1,Dial(SIP/100,20,Ttm) ; extensión 100
exten = 100,2,Voicemail(100)
exten = 100,3,Hangup
exten = 101,1,Dial(SIP/101,20,Ttm) ; extensión 101
exten = 101,2,Voicemail(101)
exten = 101,3,Hangup
exten = 102,1,Dial(SIP/102,20,Ttm) ; extensión 102
exten = 102,2,Voicemail(102)
exten = 102,3,Hangup
exten = 103,1,Dial(SIP/103,20,Ttm) ; extensión 103
exten = 103,2,Voicemail(103)
exten = 103,3,Hangup
When I receive a call from outside this is the asterisk console log:
== Using SIP RTP CoS mark 5
-- Executing [s@default:1] wait(SIP/pstn-0004, 1)
-- Executing [s@default:1] answer(SIP/pstn-0004, )
-- Digit timeout set to 5.000
-- Response timeout set to 10.000
-- Executing [s@default:1] background(SIP/pstn-0004, demo-congrats)
-- SIP/pstn-0004 Playing 'demo-congrats.slin' (language 'en')
[Oct 31 20:55:55] NOTICE[4001]: channel.c:3066 __ast_read: Dropping
incompatible voice frame on SIP/pstn-0004 of format ulaw since our
native format has changed to 0x8 (alaw)
-- Executing [s@default:1] background(SIP/pstn-0004, demo-instruct)
-- SIP/pstn-0004 Playing 'demo-instruct.slin' (language 'en')
-- Executing [s@default:1] waitexten(SIP/pstn-0004, )
-- Timeout on SIP/pstn-0004, going to 't'
-- Executing [t@default:1] playback(SIP/pstn-0004, demo-thanks)
-- SIP/pstn-0004 Playing 'demo-thanks.slin' (language 'en')
-- Executing [t@default:1] hangup(SIP/pstn-0004, )
== Spawn extension (default, t, 1) exited non-zero on 'SIP/pstn-0004'
== Using SIP RTP CoS mark 5
-- Executing [s@default:1] wait(SIP/pstn-0005, 1)
-- Executing [s@default:1] answer(SIP/pstn-0005, )
-- Digit timeout set to 5.000
-- Response timeout set to 10.000
-- Executing [s@default:1] background(SIP/pstn-0005, demo-congrats)
-- SIP/pstn-0005 Playing 'demo-congrats.slin' (language 'en')
[Oct 31 20:59:33] NOTICE[4015]: channel.c:3066 __ast_read: Dropping
incompatible voice frame on SIP/pstn-0005 of format ulaw since our
native format has changed to 0x8 (alaw)
-- Executing [s@default:1] background(SIP/pstn-0005, demo-instruct)
-- SIP/pstn-0005 Playing 'demo-instruct.slin' (language 'en')
How could I make to redirect the call to the 103 extension?
Thanks for your help, best regards.
--
Josu Lazkano
--
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Re: [asterisk-users] Calls from PSTN on SPA3102
2011/10/31 Jeroen Eeuwes jeroeneeu...@gmail.com: Hi Josu, How could I make to redirect the call to the 103 extension? In the sip.conf you have to put the correct context of your SPA3102-peer/friend. So put a line like context=entradas_pstn there. That should get it out of the default context it is now going to. Your s extension in entradas_pstn is already dialling to SIP/103 so that should be OK. Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks Jeroen!!! It works! == Using SIP RTP CoS mark 5 -- Executing [s@entradas_pstn:1] Dial(SIP/pstn-0002, SIP/103,20,tm) in new stack == Using SIP RTP CoS mark 5 -- Called 103 -- Music class default requested but no musiconhold loaded. -- SIP/103-0003 is ringing == Spawn extension (entradas_pstn, s, 1) exited non-zero on 'SIP/pstn-0002' Now I have a working Asterisk system at home, best regards. -- Josu Lazkano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] international calls with national rates
Hello, I want to know if is posible to call from a mobile to my Asterisk (national call) and then I insert an international number in my mobile and Asterisk call with a Voipbuster account. Is this possible? Thanks a lot. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] international calls with national rates
Thanks Jared, but I don't understand this. This is part of my extensiones.conf: [incoming] exten = 943712666,1,Wait(2) exten = 943712666,2,Answer() exten = 943712666,3,Background(/home/lazkano/bienvenido) exten = 943712666,4,Wait(1) exten = 943712666,5,Background(/home/lazkano/extension) exten = 943712666,6,WaitExten(5) exten = 943712666,7,Dial(SIP/102|7|tm) exten = 943712666,8,Dial(SIP/103|7|tm) exten = 943712666,9,Dial(SIP/104|7|tm) exten = 943712666,10,Dial(SIP/101|7|tm) exten = 943712666,11,Hangup() exten = 102,1,Dial(SIP/102|5|tm) exten = 102,2,Dial(mISDN/1/66502,30,twW) exten = 101,1,Dial(SIP/101|30|tm) exten = 103,1,Dial(SIP/103|30|tm) exten = 104,1,Dial(SIP/104|30|tm) exten = 110,1,... I want to do that when I enter on the 110 extension. Thanks a lot. 2007/7/25, Jared Smith [EMAIL PROTECTED]: On Wed, 2007-07-25 at 16:46 +0200, Josu Lazkano wrote: Hello, I want to know if is posible to call from a mobile to my Asterisk (national call) and then I insert an international number in my mobile and Asterisk call with a Voipbuster account. Sure! You'll want to check out the DISA() dialplan application. Check out more information at http://www.voip-info.org/wiki/view/Asterisk+cmd +DISA. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wait to numbers
Hello everybody. I have a problem with my dialplan. That my extensions.conf: [incoming] exten = 943712666,1,Wait(2) exten = 943712666,2,Answer() exten = 943712666,3,Background(/home/lazkano/welcom) exten = 943712666,4,Wait(1) exten = 943712666,5,Background(/home/lazkano/extension) exten = 943712666,6,Wait(4) exten = 943712666,7,Dial(SIP/104|30|tm) exten = 943712666,8,Hangup() exten = 101,1,Dial(SIP/101|30|tm) exten = 102,1,Dial(SIP/102|30|tm) exten = 103,1,Dial(SIP/103|30|tm) exten = 104,1,Dial(SIP/104|30|tm) When someone call to the office the a recorded voice tell welcom, them an other record says if you know the extension, press it and wait 4 seconds. The problem is that in exten = 943712666,6,Wait(4) it doesn't take any naumber you must enter the extension in the exten = 943712666,5,Background(/home/lazkano/extension). There is an other command to wait 4 seconds and wait for numbers? Thanks for all. Enjoy your day. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wait to numbers
Thankyou Jared, that it! it works! 2007/6/27, Jared Smith [EMAIL PROTECTED]: On 6/27/07, Josu Lazkano [EMAIL PROTECTED] wrote: There is an other command to wait 4 seconds and wait for numbers? Use the WaitExten() application instead of Wait(). -Jared ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outging select
Hello everybody.
I have a analog line in the office and a ISDN (with mISDN) line.
I want to call outside from the analog line, but when this is busy, I want
to call outside the second call from the ISDN line.
That my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,tm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,tm)
exten = 102,2,Hangup
exten = 103,1,Dial(SIP/103,30,tm)
exten = 103,2,Hangup
exten = 104,1,Dial(SIP/104,30,tm)
exten = 104,2,Hangup
include = outgoing_RTB
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/103,30,tm)
exten = fax,1,Dial(IAX2/200)
[incoming]
exten = 943712666,1,Wait(2)
exten = 943712666,2,Answer()
exten = 943712666,3,Background(/home/lazkano/bienvenido)
exten = 943712666,4,Wait(1)
exten = 943712666,5,Background(/home/lazkano/extension)
exten = 943712666,6,Wait(4)
exten = 943712666,7,Dial(SIP/104|30|tm)
exten = 943712666,8,Hangup()
exten = 101,1,Dial(SIP/101|30|tm)
exten = 102,1,Dial(SIP/102|30|tm)
exten = 103,1,Dial(SIP/103|30|tm)
exten = 104,1,Dial(SIP/104|30|tm)
How can I do that?
Thanks for all.
Have a nice day.
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Re: [asterisk-users] outging select
KO, thank you very much.
i will try it.
2007/6/25, Steve Totaro [EMAIL PROTECTED]:
You could combine your two contexts or use goto.
Instead of:
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
Do:
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Goto(outgoing_RDSI,_9,1)
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
Or
[outgoing_trunks]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,3,Hangup()
Thanks,
Steve Totaro
Josu Lazkano wrote:
Hello everybody.
I have a analog line in the office and a ISDN (with mISDN) line.
I want to call outside from the analog line, but when this is busy, I
want to call outside the second call from the ISDN line.
That my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,tm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,tm)
exten = 102,2,Hangup
exten = 103,1,Dial(SIP/103,30,tm)
exten = 103,2,Hangup
exten = 104,1,Dial(SIP/104,30,tm)
exten = 104,2,Hangup
include = outgoing_RTB
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/103,30,tm)
exten = fax,1,Dial(IAX2/200)
[incoming]
exten = 943712666,1,Wait(2)
exten = 943712666,2,Answer()
exten = 943712666,3,Background(/home/lazkano/bienvenido)
exten = 943712666,4,Wait(1)
exten = 943712666,5,Background(/home/lazkano/extension)
exten = 943712666,6,Wait(4)
exten = 943712666,7,Dial(SIP/104|30|tm)
exten = 943712666,8,Hangup()
exten = 101,1,Dial(SIP/101|30|tm)
exten = 102,1,Dial(SIP/102|30|tm)
exten = 103,1,Dial(SIP/103|30|tm)
exten = 104,1,Dial(SIP/104|30|tm)
How can I do that?
Thanks for all.
Have a nice day.
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Re: [asterisk-users] chan problem
Hello, the problem is nearly solved. When I restart the computer, Asterisk load prefectly but the ISDN calls doen't go. I must stop the Asterisk and run /etc/init.d/misdn-init start and then start Asterik. I have a Debian machine, I need to to do something like this: update-rc.d asterisk defaults update-rc.d misdn-init defaults but the problem is that Asterisk run before misdn-init, I and I want to start misd-init first. I dont know how to do. thanks a lot. 2007/6/21, [EMAIL PROTECTED] [EMAIL PROTECTED]: Thx, However it appears to be something else. Still need to find out what it is. Loading during boot does not work. After unloading (rmmod) modules mISDN, zaptel, wctdm etc, then reloading them manually in any particular order it works. On Mon, Jun 18, 2007 at 12:36:10PM -0400, Bob Chiodini wrote: [EMAIL PROTECTED] wrote: I experienced the same problem. The only way I could get both ISDN and analog working was unloading kernel modules for zaptel and mISDN after boot and then load them in the order: zaptel first and then mISDN. Still need to find out how to configure load order in linux. Hans, Have a look at the man page for modprobe.conf, specifically the install directive. There is an example of how to force the order. But it is already heavily abused. You may actually want to load one and not the other, and with that directive you can't . One alternative guess is the need to blacklist a third module. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan problem
update-rc.d defaults 30 10 ¿¿?? update-rc.d asterisk defaults 30 10, isn't 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
Re: [asterisk-users] chan problem
thank you Tzafrir. 2007/6/21, Tzafrir Cohen [EMAIL PROTECTED]: On Thu, Jun 21, 2007 at 10:41:29AM +0200, Josu Lazkano wrote: update-rc.d defaults 30 10 ¿¿?? update-rc.d asterisk defaults 30 10, isn't it? Right. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] chan problem
Hello, I have the same problem as the begining. I reinstall all the system and i have the same error: asterisk:/usr/src/asterisk-1.2.19# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel and wctdm modules are loaded correctly. And the zapata and zaptel files are correctly too. Thanks for all. 2007/6/21, Josu Lazkano [EMAIL PROTECTED]: thank you Tzafrir. 2007/6/21, Tzafrir Cohen [EMAIL PROTECTED]: On Thu, Jun 21, 2007 at 10:41:29AM +0200, Josu Lazkano wrote: update-rc.d defaults 30 10 ¿¿?? update-rc.d asterisk defaults 30 10, isn't it? Right. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto: [EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] chan problem
Hello, I have solved. I must delete the netjetpci module from /etc/modprobe.d/blacklist: blacklist netjetpci thanks for all 2007/6/21, Tzafrir Cohen [EMAIL PROTECTED]: On Thu, Jun 21, 2007 at 12:48:46PM +0200, Josu Lazkano wrote: Hello, I have the same problem as the begining. I reinstall all the system and i have the same error: asterisk:/usr/src/asterisk-1.2.19# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel and wctdm modules are loaded correctly. cat /proc/zaptel/* And the zapata and zaptel files are correctly too. cat /etc/zaptel.conf BTW: what is the output of: ./xpp/utils/genzaptelconf -l -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] mISDN problem
Hello everybody.
I have an other problem with mISDN.
The outgoing calls goes perfect, but the incoming no.
When people call in the CLI puts that:
*CLI Jun 20 12:32:08 WARNING[2315]: chan_misdn.c:4920 chan_misdn_log:
Extension can never match, so disconnecting
this is my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
include = outgoing_RDSI
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
[incoming]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
and my misdn.conf this:
[general]
misdn_init=/etc/misdn- init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh
[default]
context=misdn
language=es
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
dialplan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
presentation=-1
screen=-1
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
[isdn]
ports=1
context=incoming
msns=*
I don't know if the [isdn] is well
someone how has the mISDN?¿
thanks for all
Josu Lazkano
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Re: [asterisk-users] mISDN problem
Thank yo very much, it works! I had a 0 before the number, in misdn.conf -natiolapreffix=0 2007/6/20, Ex Vitorino [EMAIL PROTECTED]: You have only one extension in the [incoming] context and that is 's'. You probably need a different one -- the one the telco sends you... Ideas: 1. Try using a generic wildcard such as '_X.' instead of 's', then check the CLI after incrementing verbosity to at least 3 (BTW: don't forget reloading extensions!) 2. Enable misdn debugging to leve 3 and check its log at /var/log/asterisk/misdn.log. You will have the destination extension as the dad field, IIRC. Good luck -- Ex Vito On 6/20/07, Josu Lazkano [EMAIL PROTECTED] wrote: Hello everybody. I have an other problem with mISDN. The outgoing calls goes perfect, but the incoming no. When people call in the CLI puts that: *CLI Jun 20 12:32:08 WARNING[2315]: chan_misdn.c:4920 chan_misdn_log: Extension can never match, so disconnecting ___ --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] make config
Hello everybody, when I run make config I have this error: install: cannot stat `init.asterisk': No such file or directory make: *** [config] Error 1 I don't understand. For what is make config? to put on /etc/init.d/? Thanks for all ___ --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] problem with mISDN
Hello, I have some problems with mISDN.
I can't send or receive call from the Billion ISDN card
Mi configuration files are thoose:
extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
include = outgoing
[outgoing]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
misdn.conf:
[general]
misdn_init=/etc/misdn-init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh
[default]
context=misdn
language=es
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
dialplan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
presentation=-1
screen=-1
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
[isdn]
ports=1
context=incoming
msns=*
misdn-init.conf:
card=1,hfcpci
te_ptmp=1
poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0
When I make a call the CLI tells that:
*CLI -- Executing Dial(SIP/101-081990e8, mISDN/1/943833473|45|tTwW)
in new stack
Jun 19 12:32:25 WARNING[2153]: channel.c:2618 ast_request: No channel type
registered for 'mISDN'
Jun 19 12:32:25 NOTICE[2153]: app_dial.c:1076 dial_exec_full: Unable to
create channel of type 'mISDN' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/101-081990e8, ) in new stack
== Spawn extension (SOME, 943833473, 102) exited non-zero on
'SIP/101-081990e8'
It looks like the channel isn't registered, but I don't know what to do.
Thank you everybody!!!
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Re: [asterisk-users] problem with mISDN
I can't make a misdn show channels on the CLI.
It looks like the mISND isn`t registered.
thanks for all
2007/6/19, Josu Lazkano [EMAIL PROTECTED]:
Hello, I have some problems with mISDN.
I can't send or receive call from the Billion ISDN card
Mi configuration files are thoose:
extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
include = outgoing
[outgoing]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
misdn.conf:
[general]
misdn_init=/etc/misdn-init.conf
debug=0
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh
[default]
context=misdn
language=es
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
dialplan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
presentation=-1
screen=-1
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
[isdn]
ports=1
context=incoming
msns=*
misdn-init.conf:
card=1,hfcpci
te_ptmp=1
poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0
When I make a call the CLI tells that:
*CLI -- Executing Dial(SIP/101-081990e8,
mISDN/1/943833473|45|tTwW) in new stack
Jun 19 12:32:25 WARNING[2153]: channel.c:2618 ast_request: No channel type
registered for 'mISDN'
Jun 19 12:32:25 NOTICE[2153]: app_dial.c:1076 dial_exec_full: Unable to
create channel of type 'mISDN' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup(SIP/101-081990e8, ) in new stack
== Spawn extension (SOME, 943833473, 102) exited non-zero on
'SIP/101-081990e8'
It looks like the channel isn't registered, but I don't know what to do.
Thank you everybody!!!
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[asterisk-users] chan problem
Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) And when I run asterisk -vvvc this: Jun 18 12:50:15 WARNING[2218]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Jun 18 12:50:15 ERROR[2218]: chan_zap.c:7038 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 18 12:50:15 ERROR[2218]: chan_zap.c:10472 setup_zap: Unable to register channel '1' Jun 18 12:50:15 WARNING[2218]: loader.c:415 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 18 12:50:15 WARNING[2218]: loader.c:555 load_modules: Loading module chan_zap.so failed! My configuration files: zaptel.conf: loadzone=es defaultzone=es fxsks=1 zapata.conf: [channels] signalling=fxs_ks usecallerid=yes callwaiting=no threewaycalling=no transfer=yes cancallforward=yes echocancel=yes echotraining=yes echocancelwhenbridged=no rxgain=0 txgain=0 group=1 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming answeronpolarityswitch=yes hanguponpolarityswitch=yes polarityonanswerdelay=600 progzone=es channel = 1 I don't know what is the problem, if someone knows... Thanks and have a nice day. ___ --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] chan problem
Hello, my OpneVox card is an A400P01. And the output of lspci is: 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host Bridge 00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge 00:0b.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) 00:0c.0 VGA compatible controller: S3 Inc. 86c325 [ViRGE] (rev 06) 00:0e.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) 00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 74) and the cat /proc/zaptel/* is: cat: /proc/zaptel/*: No such file or directory there is no files in /proc/zaptel/ thanks for all 2007/6/18, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Jun 18, 2007 at 12:50:23PM +0200, Josu Lazkano wrote: Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) This means your /etc/zaptel.conf is incorrect. Start by correcting it. Or maybe the driver has not loaded correctly. Which OpenVox card is it? What is the output of: lspci cat /proc/zaptel/* zaptel.conf: loadzone=es defaultzone=es fxsks=1 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] chan problem
The card is a OpenVox 2007/6/18, Josu Lazkano [EMAIL PROTECTED]: Hello, my OpneVox card is an A400P01. And the output of lspci is: 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host Bridge 00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge 00:0b.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) 00:0c.0 VGA compatible controller: S3 Inc. 86c325 [ViRGE] (rev 06) 00:0e.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) 00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 74) and the cat /proc/zaptel/* is: cat: /proc/zaptel/*: No such file or directory there is no files in /proc/zaptel/ thanks for all 2007/6/18, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Jun 18, 2007 at 12:50:23PM +0200, Josu Lazkano wrote: Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) This means your /etc/zaptel.conf is incorrect. Start by correcting it. Or maybe the driver has not loaded correctly. Which OpenVox card is it? What is the output of: lspci cat /proc/zaptel/* zaptel.conf: loadzone=es defaultzone=es fxsks=1 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] chan problem
how can i see that??? thanks 2007/6/18, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Jun 18, 2007 at 02:02:53PM +0200, Josu Lazkano wrote: Hello, my OpneVox card is an A400P01. And the output of lspci is: 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host Bridge 00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge 00:0b.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) 00:0c.0 VGA compatible controller: S3 Inc. 86c325 [ViRGE] (rev 06) 00:0e.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) 00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 74) and the cat /proc/zaptel/* is: cat: /proc/zaptel/*: No such file or directory there is no files in /proc/zaptel/ lsmod | grep zaptel Anything relevant in the kernel logs from the loading of the module wctdm ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] call from ISDN
Hello, I removed the two lines: nationalprefix = 0 internationalprefix = 00 And I run bri debug span 1: *CLI bri debug span 1 Enabled debugging on span 1 1 Timed out looking for release complete 1 Protocol Discriminator: Q.931 (8) len=8 1 Call Ref: len= 1 (reference 2/0x2) (Originator) 1 Message type: RELEASE (77) 1 [1 081 021 811 901 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] 1 Final time-out looking for release complete 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Jun 13 11:23:40 WARNING[2226]: chan_zap.c:8463 pri_fixup_principle: Call specified, but not found? -- Executing Dial(SIP/101-d08c, ZAP/g1/943833473|45|tTwW) in new stack 1 -- Making new call for cr 131 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (8) len=32 1 Call Ref: len= 1 (reference 3/0x3) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 811 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 051 211 811 311 301 311 ] 1 Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number passed network screening (1) '101' ] 1 [1 701 0a1 801 391 341 331 381 331 331 341 371 331 ] 1 Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '943833473' ] 1 [1 a11 ] 1 Sending Complete (len= 1) -- Called g1/943833473 1 No response to SETUP message 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending 1 Protocol Discriminator: Q.931 (8) len=8 1 Call Ref: len= 1 (reference 3/0x3) (Originator) 1 Message type: DISCONNECT (69) 1 [1 081 021 811 921 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (18), class = Normal Event (1) ] -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Request, peerstate Disconnect Indication -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/101-d08c, ) in new stack == Spawn extension (SOME, 943833473, 102) exited non-zero on 'SIP/101-d08c' Is the same, anyones know what is??? Thanks a lot. bye!!! 2007/6/12, Tzafrir Cohen [EMAIL PROTECTED]: On Tue, Jun 12, 2007 at 06:13:19PM +0200, Josu Lazkano wrote: Of course, thanks for respose Tzafrir. Here is my zapata.conf: [trunkgroups] [channels] language=es context=default switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = unknown overlapdial = no usecallerid = yes callerid = asreceived callprogress = no hidecallerid = no nationalprefix = 0 internationalprefix = 00 Try removing the above two lines and a restart. If that doesn't help, please run: bri debug span 1 and provide a trace of an outgoing call. immediate = no faxdetect = incoming echocancel = yes echotraining = yes echocancelwhenbridged = yes context = incoming group = 1 threewaycalling = yes transfer = yes channel = 1-2 I am having a lot of interrupt problem. Thanks for all. Bye bye. 2007/6/12, Tzafrir Cohen [EMAIL PROTECTED]: On Tue, Jun 12, 2007 at 03:54:13PM +0200, Josu Lazkano wrote: Hello everybody, I have installed the Billion ISDN on a Debian machine. I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI -- Executing Dial(SIP/101-f9eb, ZAP/g1/943833473|45|tTwW) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/943833473 -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/101-f9eb, ) in new stack == Spawn extension (SOME, 943833473, 102) exited non-zero on 'SIP/101-f9eb' My extensions.conf is this one: [general] static=yes writeprotect=yes [SOME] exten = 101,1,Dial(SIP/101,30,Ttm) exten = 101,2,Hangup exten = 102,1,Dial(SIP/102,30,Ttm) exten = 102,2,Hangup exten = 103,1,Dial(SIP
[asterisk-users] zaphfc problem
Hello everybody. I have a problem with my Billion ISDN card. When I run Asterisk (asterisk -vvvc) on five minutes (aprox.) it puts in the screen this: zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xff, card = 0). in the framelen it change 3 and 2. Anyone knows something about it? Thanks a lot. bye! ___ --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] re:zaphfc problem (Josu Lazkano)
where can I download that patch thanks for respons 2007/6/13, Mauro Zanin [EMAIL PROTECTED]: Try florz patch, when installing your Bristuff, for me it worked. Ciao Mauro ___ --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] mISDN problem
Hello everybody. I am trying to configure an Asterisk on Debian with the Billion ISDN card. I am using mISDN. But when I call on the CLI apears this: -- Executing Dial(SIP/101-081805b8, mISDN/1/943833473|45|tTwW) in new stack -- Called 1/943833473 P[ 1] empty_chan_in_stack: cannot empty channel 255 P[ 1] -- we have already send Release_complete == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/101-081805b8, ) in new stack == Spawn extension (SOME, 943833473, 102) exited non-zero on 'SIP/101-081805b8' I dont't know what happen. Some can help me??? Thanks to everybody. How can I saw the status of the ISDN??? ___ --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] call from ISDN
Hello everybody, I have installed the Billion ISDN on a Debian machine.
I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So
I connect the Billion ISDN to the ISDN Box and I call from a extension to
outside.
But it doesn't work, that is what I have in the CLI:
*CLI -- Executing Dial(SIP/101-f9eb, ZAP/g1/943833473|45|tTwW) in
new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/943833473
-- Channel 0/1, span 1 got hangup, cause 42
-- Zap/1-1 is circuit-busy
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/101-f9eb, ) in new stack
== Spawn extension (SOME, 943833473, 102) exited non-zero on
'SIP/101-f9eb'
My extensions.conf is this one:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
exten = 103,1,Dial(SIP/103,30,Ttm)
exten = 103,2,Hangup
exten = 104,1,Dial(SIP/104,30,Ttm)
exten = 104,2,Hangup
include = outgoing
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,Dial(SIP/101,30,Ttm)
[outgoing]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
Why is that?
Thanks everybody.
Have a nice day!!!
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[asterisk-users] Warning on CLI
Hello everybody again. I have a warning message in the CLI: *CLI Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle: Call specified, but not found? *CLI Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle: Call specified, but not found I don't know what it means. Can you help with this??? Thankyou very much. Bye bye... ___ --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] call from ISDN
Of course, thanks for respose Tzafrir.
Here is my zapata.conf:
[trunkgroups]
[channels]
language=es
context=default
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = unknown
overlapdial = no
usecallerid = yes
callerid = asreceived
callprogress = no
hidecallerid = no
nationalprefix = 0
internationalprefix = 00
immediate = no
faxdetect = incoming
echocancel = yes
echotraining = yes
echocancelwhenbridged = yes
context = incoming
group = 1
threewaycalling = yes
transfer = yes
channel = 1-2
I am having a lot of interrupt problem.
Thanks for all.
Bye bye.
2007/6/12, Tzafrir Cohen [EMAIL PROTECTED]:
On Tue, Jun 12, 2007 at 03:54:13PM +0200, Josu Lazkano wrote:
Hello everybody, I have installed the Billion ISDN on a Debian machine.
I proved to call with a ISDN telephone conected to ISDN Box and it is
OK. So
I connect the Billion ISDN to the ISDN Box and I call from a extension
to
outside.
But it doesn't work, that is what I have in the CLI:
*CLI -- Executing Dial(SIP/101-f9eb, ZAP/g1/943833473|45|tTwW)
in
new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/943833473
-- Channel 0/1, span 1 got hangup, cause 42
-- Zap/1-1 is circuit-busy
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/101-f9eb, ) in new stack
== Spawn extension (SOME, 943833473, 102) exited non-zero on
'SIP/101-f9eb'
My extensions.conf is this one:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
exten = 103,1,Dial(SIP/103,30,Ttm)
exten = 103,2,Hangup
exten = 104,1,Dial(SIP/104,30,Ttm)
exten = 104,2,Hangup
include = outgoing
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,Dial(SIP/101,30,Ttm)
[outgoing]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
Why is that?
Could you please provide your zapata.conf ?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Billion on Debian Etch
Hello everybody, I am 20 days with the same item and I can't configure it. I want to know if someone has the Billion ISDN card on a Debian Etch, because everybody tells me to do that, then the other one but no one has the same configuration. If some one has the same configuration (Billion + Debian Etc), can you help? What packages install and what steps continue. Thanks to all and have a good day. Josu Lazkano ___ --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] OpenVox A400P01on thin client?
I have this card and no problem. It is very simple to configure. Go on! 2007/5/29, Gilles Ganault [EMAIL PROTECTED]: Hello, I'm thinking of ordering an OpenVox A400P01 (A400P + 1 PORT FXO Bundle) for use in a old IBM 8364 thin client: http://www.openvox.com.cn/products_detail.php?genre_id=9id=28 http://silicon-verl.de/home/flo/software/netstation-8364/ Has someone already used this hardware with Asterisk, especially on a small piece of hardware like this, and could offer some feedback? Thank you. ___ --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] redirect on AT-530 IP Phone
Good morning everybody! I have two AT-530 IP phones, when a call entry from outside (zap channel9 it goes to 101 extension. When I take the call and start to speak with the other person, how can I redirect this call to 102 extension? Thank to all. Bye!!! ___ --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] modprobe
Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all. ___ --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] modprobe
Hello john, thanks for response. I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term apt-get install libncurses5 libncurses5-dev apt-get install bison apt-get install openssl apt-get install libssl0.9.8 apt-get install libssl-dev apt-get install libeditline0 libeditline-dev libedit-dev libedit2 apt-get install gcc apt-get install zlib1g-dev To install Asterisk with Bristuff I do that: in usr/src: wget http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-current.tar.gz http://212.91.251.199/%7Ejunghanns.net/downloads/bristuff-0.3.0-current.tar.gz tar zxvf bristuff-0.3.0-current.tar.gz cd bristuff-0.3.0-PRE-1r ./install.sh That could help? Thanksss 2007/5/24, John covici [EMAIL PROTECTED]: We would need more details to help -- version of asterisk and zaptel and what you did to try to install them -- hardware you have, etc and why you did that modprobe statement. on Thursday 05/24/2007 Josu Lazkano([EMAIL PROTECTED]) wrote Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all. Hello every boy againbrbrI have some problems with modprobe. When I type quot;modprobe zaphfcquot;, this error happens quot;FATAL: Module zaphfc not found.quot;brbrAnd when I tyoe quot;ztcfg -vvquot; this error happens: brbrNotice: Configuration file is /etc/zaptel.confbrline 0: Unable to open master device #39;/dev/zap/ctl#39;brbr1 error(s) detectedbrbrSomeone can help me???brbrThanks to all.br ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] modprobe
Hello Tzafrir, thanks for response. I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term apt-get install libncurses5 libncurses5-dev apt-get install bison apt-get install openssl apt-get install libssl0.9.8 apt-get install libssl-dev apt-get install libeditline0 libeditline-dev libedit-dev libedit2 apt-get install gcc apt-get install zlib1g-dev To install Asterisk with Bristuff I do that: in usr/src: wget http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-current.tar.gz http://212.91.251.199/%7Ejunghanns.net/downloads/bristuff-0.3.0-current.tar.gz tar zxvf bristuff-0.3.0-current.tar.gz cd bristuff-0.3.0-PRE-1r ./install.sh That could help? Thanksss 2007/5/24, Tzafrir Cohen [EMAIL PROTECTED]: On Thu, May 24, 2007 at 11:17:57AM +0200, Josu Lazkano wrote: Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. zaphfc is part of bristuff. have you installed brisuff (or any other bristuffed zaptel package, such as the one from Debian)? And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' This is normally an indication that the module zaptel is not loaded. Which makes sense, as the driver you modprobed for did not exist nd hence could not pull zaptel with it. What zaptel hardware do you have? How have you installed Zaptel? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] modprobe
Thanks Giorgio!!! I made modprobe zaptel and then ztcfg -vv anI have this: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) I think is better but not enough, thanks for that. Anyone uses the Billion ISDN PCI? Thanks every body!!! 2007/5/24, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Josu, I had the same problem with wctdm.I just loaded zaptel before wctdm and it was all ok. Hope it can help you. :) Giorgio Incantalupo Josu Lazkano wrote: Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module zaphfc not found. And when I tyoe ztcfg -vv this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all. ___ --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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ --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] Bristuff with Billion ISDN
Hello, I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term apt-get install libncurses5 libncurses5-dev apt-get install bison apt-get install openssl apt-get install libssl0.9.8 apt-get install libssl-dev apt-get install libeditline0 libeditline-dev libedit-dev libedit2 apt-get install gcc apt-get install zlib1g-dev To install Asterisk with Bristuff I do that: in usr/src: wget http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-current.tar.gzhttp://212.91.251.199/%7Ejunghanns.net/downloads/bristuff-0.3.0-current.tar.gz tar zxvf bristuff-0.3.0-current.tar.gz cd bristuff-0.3.0-PRE-1r ./install.sh when it is compilin I have that error: LIBGSM installed. Press Enter to continue, or CTRL + C to abort. rm -f ztgsm.o *.ko *.mod.c *.mod.o .*o.cmd *~ rm -rf .tmp_versions Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 uno/duo/quad GSM PCI driver installed. Press Enter to continue, or CTRL + C to abort. anyone knows why? I think is somethink with kernel sources, but I don't know where are they. Thanks to all ___ --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] record voice
Hello everybody! I have a problem recording voices for my Asterisk menu. I used the Record(/home/lazkano/bienvenido:gsm) function to record the menu voices, but when I call from outside or from an extension the voice listen so low. is there any software to record my voice properly and convert to gsm format? Someone use an other function for that? Thank a lot to everybody. Enjoy your weekend!!!___ --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] record voice
thankyou very much, i will probe it byee - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 11, 2007 10:35 AM Subject: Re: [asterisk-users] record voice On Fri, 11 May 2007, Josu Lazkano Lete wrote: Hello everybody! I have a problem recording voices for my Asterisk menu. I used the Record(/home/lazkano/bienvenido:gsm) function to record the menu voices, but when I call from outside or from an extension the voice listen so low. is there any software to record my voice properly and convert to gsm format? Someone use an other function for that? Audacity can record sound from a PC's microphone, (or better, the line-in socket if you have a good pre-amp and proper microphone) manipulate it, etc. It's also cross platform (Win/Linux/Mac) http://audacity.sourceforge.net/ You could then store your prompts in all the codec formats you support, then asterisk wouldn't have to do transcoding either. Gordon ___ --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] AT530 Telephone
Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten = 105,1,Answer exten = 105,2,Background(/home/user/suport) exten = 1,1,Dial(SIP/101,30,Ttm) exten = 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option 1 it doesn't redirect to 101 extension. Otherwise with the X-Lite extension I select 1 or 2 options and it works perfectly. Anyone has the same problem? I must push another button to redirect well? Thanks to all. Bye!___ --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] AT530 Telephone
I have DTMF_RELAY which do you recomend? the options are. DTMF_RELAY DTMF_RFC2833 DTMF_SIP_INFO thanks - Original Message - From: Alexandre VERNIOL [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 10, 2007 10:15 AM Subject: Re: [asterisk-users] AT530 Telephone Hi, What sort of DTMF do you use in the AT530 ? It seems that just a problem of DTMF otherwise it don't work with your softphone. Cheers, Josu Lazkano Lete a écrit : Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten = 105,1,Answer exten = 105,2,Background(/home/user/suport) exten = 1,1,Dial(SIP/101,30,Ttm) exten = 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option 1 it doesn't redirect to 101 extension. Otherwise with the X-Lite extension I select 1 or 2 options and it works perfectly. Anyone has the same problem? I must push another button to redirect well? Thanks to all. Bye! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AT530 Telephone
Perfect!!! It works!!! on sip.conf where I must put dtmfmode=rfc2833? on the extensions (101, 102, ...) or in the sip accounts (voipbuster)? thank you very much have a nice day!!! - Original Message - From: Alexandre VERNIOL [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 10, 2007 10:33 AM Subject: Re: [asterisk-users] AT530 Telephone Use this one DTMF_RFC2833 Be sure to have in your peers definition this line (sip.conf): [peer] dtmfmode=rfc2833 Cheers, Josu Lazkano Lete a écrit : I have DTMF_RELAY which do you recomend? the options are. DTMF_RELAY DTMF_RFC2833 DTMF_SIP_INFO thanks - Original Message - From: Alexandre VERNIOL [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 10, 2007 10:15 AM Subject: Re: [asterisk-users] AT530 Telephone Hi, What sort of DTMF do you use in the AT530 ? It seems that just a problem of DTMF otherwise it don't work with your softphone. Cheers, Josu Lazkano Lete a écrit : Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten = 105,1,Answer exten = 105,2,Background(/home/user/suport) exten = 1,1,Dial(SIP/101,30,Ttm) exten = 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option 1 it doesn't redirect to 101 extension. Otherwise with the X-Lite extension I select 1 or 2 options and it works perfectly. Anyone has the same problem? I must push another button to redirect well? Thanks to all. Bye! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] fax receiving
Hello everybody, I am receiving faxes and I don`t know how to receive, is there any posibility to receive it on amail account?¿ in the console the message is this: May 9 15:47:44 NOTICE[2618]: chan_zap.c:3703 zt_handle_dtmfup: Fax detected, but no fax extension -- SIP/101-0819b4f8 answered Zap/1-1 thanks to all___ --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] select menu
Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten = s,1,Answer() exten = s,2,Wait(1) exten = s,3,Dial(SIP/101,30,Ttm) sorry about my english, thanks to all be ___ --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] outgoing calls
hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME (domain 101) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.0.0.9:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport From: asterisk sip:[EMAIL PROTECTED];tag=as0cc11f28 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED]
[asterisk-users] outgoing calls
hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME (domain 101) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.0.0.9:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport From: asterisk sip:[EMAIL PROTECTED];tag=as0cc11f28 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED]
[asterisk-users] outgoing calls
hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME (domain 101) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.0.0.9:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport From: asterisk sip:[EMAIL PROTECTED];tag=as0cc11f28 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED]
Re: [asterisk-users] outgoing calls
thank you very much! it works - Original Message - From: Dijkstra, Roelof To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, May 08, 2007 1:53 PM Subject: RE: [asterisk-users] outgoing calls Hello Josu, In you're sip.conf you have the 2 phones configured that they are in the SOME context. Looking at the SOME contect in extensions.conf you only have the 2 phones defined. If you want to call ouside from the SOME context as well, you need to include the outgoing context there as well. Regards, Roelof Dijkstra Network Engineer EMEA Compuware Europe BV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Josu Lazkano Lete Sent: Tuesday, May 08, 2007 1:36 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] outgoing calls hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME
[asterisk-users] load modules
Hello again, I have a little problem, every time I switch on the Asterisk server I must load two modules: modprobe zaptel and modprobe wctdm Is there any way to load there automatically when the server start? I have a Debian Etch. One more cuestion, it's posible to start Asterisk (asterisk -vvvc)as well? What metod do you prefer? asterisk or asterisk -vvvc? Thanks very much to all of you. Bye.___ --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] ISDN with Billion
Hello again. I can't configure the Billion PCI in my ISDN. I want to know if AsteriskNow and the TrixBox LiveCDs configure it automatically. Thanks to all___ --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] can´t anserd the call
hello, I have instaled a analog line, and when I call on the console apears
that:
I want to redirect the call to 101 extension.
*CLI -- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to context
'default'
Apr 27 08:15:53 WARNING[3494]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent
into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Apr 27 08:15:58 NOTICE[3497]: chan_zap.c:6223 ss_thread: Got event 18 (Ring
Begin)...
Apr 27 08:16:00 NOTICE[3497]: chan_zap.c:6223 ss_thread: Got event 2
(Ring/Answered)...
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to context
'default'
Apr 27 08:16:00 WARNING[3497]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent
into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'Zap/1-1'
mi configuration files are this:
extensions.conf:
[general]
static=yes
writeprotect=yes
;autofallthrough=yes
;clearglobalvars=no
;priorityjumping=no
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,Dial(SIP/101,30,Ttm)
[outgoing]
exten =_94XXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =_94XXX,2,Hangup()
exten =_94XXX,102,Hangup()
zapata.conf:
[channels]
signalling=fxs_ks
usecallerid=yes
callwaiting=no
threewaycalling=no
transfer=yes
cancallforward=yes
; valores validos 256(32ms),512(64ms),1024(128ms)
echocancel=yes
echotraining=yes
echocancelwhenbridged=no
rxgain=0
txgain=0
group=1
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
;busydetect=yes
;busycount=10
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
polarityonanswerdelay=600
;callprogress=no
progzone=es
channel = 1
zaptel.conf:
loadzone=es
defaultzone=es
fxsks=1
sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[101]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME
[102]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME
thanks for all!!!___
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[asterisk-users] problem with A400P01 OpenVox
Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line
connected.
I am new in Linux and Asterisk, my steps are theese:
1. Install CentOS 4.4 (basic instalation).
2. Command line:
yum -y update
yum install gcc kernel-devel bison openssl-devel
yum install openssl-devel
3. Download the source:
wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz
4. Uncompress:
tar xvfz asterisk-1.2.17.tar.gz
tar xvfz zaptel-1.2.16.tar.gz
5. Compile:
cd zaptel-1.2.16
make clean
make
make install
cd ..
cd asterisk-1.2.17
make clean
make
make install
make samples
make config
Mi configuration files:
zaptel.com
loadzone=es
defaultzone=es
fxsks=1
zapata.conf
[channels]
signalling=fxs_ks
usecallerid=yes
callwaiting=no
threewaycalling=no
transfer=yes
cancallforward=yes
; valores validos 256(32ms),512(64ms),1024(128ms)
echocancel=yes
echotraining=yes
echocancelwhenbridged=no
rxgain=0
txgain=0
group=1
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
;busydetect=yes
;busycount=10
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
polarityonanswerdelay=600
;callprogress=no
progzone=es
channel = 1
sip.conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[101]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME
[102]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME
extensions.conf
[general]
static=yes
writeprotect=yes
;autofallthrough=yes
;clearglobalvars=no
;priorityjumping=no
[SOME]
exten = 101,1,Dial(SIP/101,30,Ttm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,Ttm)
exten = 102,2,Hangup
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,Dial(SIP/101,30,Ttm)
[outgoing]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
Command line:
modprobe zaptel
modprobe wcfxo
modprobe wctdm
Then I start Asterisk (asterisk -vvvc), and when I call to the analog line
number, the console shows that:
*CLI -- Starting simple switch on 'Zap/1-1'
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to context
'default'
Apr 26 19:34:33 WARNING[3818]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1'
sent into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Apr 26 19:34:38 NOTICE[3821]: chan_zap.c:6223 ss_thread: Got event 18 (Ring
Begin)...
Apr 26 19:34:40 NOTICE[3821]: chan_zap.c:6223 ss_thread: Got event 2
(Ring/Answered)...
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to context
'default'
Apr 26 19:34:40 WARNING[3821]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1'
sent into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Apr 26 19:34:47 NOTICE[3824]: chan_zap.c:6223 ss_thread: Got event 18 (Ring
Begin)...
Apr 26 19:34:49 NOTICE[3824]: chan_zap.c:6223 ss_thread: Got event 2
(Ring/Answered)...
== Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at default,s,1 still failed so falling back to context
'default'
Apr 26 19:34:49 WARNING[3824]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1'
sent into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'Zap/1-1'
The call doesn't ring, I want to redirect to extension 101.
Thank you very much for your time.
See you,
Josu Lazkano
___
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[asterisk-users] help please
hello, I have a A400P01 PCI from OpenVox. I have installed some extension and a VoipBuste account to callo out of my LAN. How can I receive and send calls from a nd to outside by my analog line??? I want to receive dthe calls from 20100 extension. Here you have my config files, thanks for all. zaptel.conf Description: Binary data extensions.conf Description: Binary data sip.conf Description: Binary data zapata.conf Description: Binary data ___ --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] help please
hello, I have a A400P01 PCI from OpenVox. I have installed some extension and a VoipBuste account to callo out of my LAN. How can I receive and send calls from a nd to outside by my analog line??? I want to receive dthe calls from 20100 extension. Here you have my config files, thanks for all. asterisk.rar Description: Binary data ___ --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] help please
hello, I have a A400P01 PCI from OpenVox. I have installed some extension and a VoipBuste account to callo out of my LAN. How can I receive and send calls from a nd to outside by my analog line??? I want to receive dthe calls from 20100 extension. Here you have my config files, thanks for all.fxsks=1 loadzone=es defaultzone=es[general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [miprimerejemplo] exten = 2,1,Dial(SIP/2,30,Ttm) exten = 2,2,Hangup exten = 2,102,Voicemail(2) exten = 2,103,Hangup exten = 20100,1,Dial(SIP/20100,30,Ttm) exten = 20100,2,Hangup exten = 20100,102,Voicemail(20100) exten = 20100,103,Hangup exten = 20200,1,Dial(SIP/20200,30,Ttm) exten = 20200,2,Hangup exten = 202000,102,Voicemail(20200) exten = 20200,103,Hangup exten = 20300,1,Dial(SIP/20300,30,Ttm) exten = 20300,2,Hangup exten = 203000,102,Voicemail(20300) exten = 20300,103,Hangup exten = 20400,1,Dial(SIP/20400,30,Ttm) exten = 20400,2,Hangup exten = 204000,102,Voicemail(20400) exten = 20400,103,Hangup exten = 3,1,VoicemailMain exten = _9,1,Dial(SIP/[EMAIL PROTECTED]) exten = _9,2,Hangup[general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [2] type=friend secret=some qualify=yes nat=no host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [20100] type=friend secret=some qualify=yes nat=yes host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [20200] type=friend secret=some qualify=yes nat=no host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [20300] type=friend secret=some qualify=yes nat=no host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [20400] type=friend secret=some qualify=yes nat=no host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [VoipBuster] type=peer host=sip.voipbuster.com username=somesi3 fromuser=somesi3 secret=some[channels] language=es context=incoming switchtype=euroisdn usercallid=yes hidecallerid=no musiconhold=default callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes inmediate=no canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbriged=yes rxgain=0.0 txgain=0.0 group=1 signalling=fxs_ks context=incoming channel=4___ --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 on Debian Etch
hello, I have two new cards, one is A400P01 from OpenVox and the other is a BILLION ISDN. I have Debian Etch installed. I want install this packages: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz I need some other packages??? I need other libraries befero install thoose packages??? thanks a lot___ --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] Billion ISDN problem
hello friends, I am configurin my Billion ISDN and when I start asterisk (asterisk -vvvc) I have this error message: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 29 of zapata.conf Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 30 of zapata.conf Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 31 of zapata.conf -- Registered channel 1, PRI Signalling signalling Apr 23 15:27:23 WARNING[2205]: chan_zap.c:1099 zt_open: Unable to specify channel 2: No such device Apr 23 15:27:23 ERROR[2205]: chan_zap.c:7241 mkintf: Unable to open channel 2: No such device here = 0, tmp-channel = 2, channel = 2 Apr 23 15:27:23 ERROR[2205]: chan_zap.c:12011 setup_zap: Unable to register channel '1-2' Apr 23 15:27:23 WARNING[2205]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 -- Unregistered channel 1 Apr 23 15:27:23 WARNING[2205]: loader.c:554 load_modules: Loading module chan_zap.so failed! can you help me please??? thanks a lot___ --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] A400P01 from OpenVox
hello, I have the A400P01 from OpenVox. Is necesary to install all this packages? http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz or just with asterisk and zaptel is enough. thanks a lot___ --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] simplify
hello friends, is there any way to simplify that extensions.conf file? [miprimerejemplo] exten = 2,1,Dial(SIP/2,30,Ttm) exten = 2,2,Hangup exten = 2,102,Voicemail(2) exten = 2,103,Hangup exten = 20100,1,Dial(SIP/20100,30,Ttm) exten = 20100,2,Hangup exten = 20100,102,Voicemail(20100) exten = 20100,103,Hangup exten = 20200,1,Dial(SIP/20200,30,Ttm) exten = 20200,2,Hangup exten = 202000,102,Voicemail(20200) exten = 20200,103,Hangup exten = 20300,1,Dial(SIP/20300,30,Ttm) exten = 20300,2,Hangup exten = 203000,102,Voicemail(20300) exten = 20300,103,Hangup exten = 20400,1,Dial(SIP/20400,30,Ttm) exten = 20400,2,Hangup exten = 204000,102,Voicemail(20400) exten = 20400,103,Hangup thanks to all___ --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] just on my LAN
hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 thanks ___ --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] just call to user
hello i have installed Asterisk on a Debian machine by apt-get install asterisk I only want to call a user inside the LAN, what files I have to edit??? sip.conf??? thanks for all___ --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 on debian
hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks___ --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
