Jason
2004-Feb-23 08:25 UTC
[Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?
Hello. I've just recently purchased the Asterisk Developers Kit so we can figure out how to get away from our Nortel system and go to IP based phones. I have a RH 9 box loaded with Asterisk (a very recent cvs download). Either way, I can call the asterisk box and get their demo playing fine. I can even call the SIP phone I've hooked up when I call in from my cell phone to the asterisk box, and that works. I cannot call out with my SIP phone though. It'll dial, ring my cell phone twice and then give up and complain that its busy. Even if I try to answer the cell phone during the first ring. Does anyone have a config they could share with me on how to make this setup work? This sounds like it should be fairly trivial, but I've beaten my head against the wall on this for a few days. =) Thanks alot, Jason
Regovich, Timothy
2004-Feb-23 08:45 UTC
[Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?
Jason, Include your sip and extensions files so people can take a look. T -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jason Sent: Monday, February 23, 2004 10:25 AM To: asterisk-users@lists.digium.com Cc: horner@med-web.com Subject: [Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone? Hello. I've just recently purchased the Asterisk Developers Kit so we can figure out how to get away from our Nortel system and go to IP based phones. I have a RH 9 box loaded with Asterisk (a very recent cvs download). Either way, I can call the asterisk box and get their demo playing fine. I can even call the SIP phone I've hooked up when I call in from my cell phone to the asterisk box, and that works. I cannot call out with my SIP phone though. It'll dial, ring my cell phone twice and then give up and complain that its busy. Even if I try to answer the cell phone during the first ring. Does anyone have a config they could share with me on how to make this setup work? This sounds like it should be fairly trivial, but I've beaten my head against the wall on this for a few days. =) Thanks alot, Jason _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------------------------------------------------------ Notice: This e-mail message, together with any attachments, contains information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. ------------------------------------------------------------------------------
Regovich, Timothy
2004-Feb-23 11:41 UTC
[Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?
Try moving your sip phone into its own context, instead of default (I use
"sip") and create a [sip] section in your extensions.conf
Add a sepcific extension to test your outgoing, like :
exten => _5XXXX,1,Dial,Zap/1/800551212
T
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jason
Sent: Monday, February 23, 2004 1:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] An example config for using a Wildcard X100P
and a SIP phone?
Timothy,
I have minimally modified the demo files that came with Asterisk, so
what is posted below is most of the comments and the demo section
removed from the config files.
Thanks!
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[sipphone]
type=friend
username=sipphone
fromuser=Sipster ; Specify user to put in "from"
instead
of callerid
secret=password
host=dynamic
defaultip=192.168.1.201
amaflags=default ; Choices are default, omit, billing,
documentation
accountcode=Sipster ; Users may be associated with an
accountcode tp ease billing
mailbox=431
--------------------------
extensions.conf
--------------------------
[general]
static=yes
writeprotect=no
[globals]
;CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/1 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider
[iaxtel700]
exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
;include => default
;include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2}) ; Ring the interface, 20
seconds maximum
exten => s,2,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten => s,3,Goto(default,s,1) ; If they press #,
return to start
exten => s,102,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/ busy announce
exten => s,103,Goto(default,s,1) ; If they press #,
return to start
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => local
exten => 431,1,Dial,SIP/sipphone
Regovich, Timothy wrote:
>Jason,
>
>Include your sip and extensions files so people can take a look.
>
>T
>
>-----Original Message-----
>From: asterisk-users-admin@lists.digium.com
>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jason
>Sent: Monday, February 23, 2004 10:25 AM
>To: asterisk-users@lists.digium.com
>Cc: horner@med-web.com
>Subject: [Asterisk-Users] An example config for using a Wildcard X100P and
a>SIP phone?
>
>
>Hello.
>
>I've just recently purchased the Asterisk Developers Kit so we can
>figure out how to get away from our Nortel system and go to IP based
>phones. I have a RH 9 box loaded with Asterisk (a very recent cvs
download).>
>Either way, I can call the asterisk box and get their demo playing fine.
>I can even call the SIP phone I've hooked up when I call in from my cell
>phone to the asterisk box, and that works.
>
>I cannot call out with my SIP phone though. It'll dial, ring my cell
>phone twice and then give up and complain that its busy. Even if I try
>to answer the cell phone during the first ring.
>
>Does anyone have a config they could share with me on how to make this
>setup work? This sounds like it should be fairly trivial, but I've
>beaten my head against the wall on this for a few days. =)
>
>Thanks alot,
>Jason
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>---------------------------------------------------------------------------
--->Notice: This e-mail message, together with any attachments, contains
information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New
Jersey, USA 08889), and/or its affiliates (which may be known outside the
United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan, as
Banyu) that may be confidential, proprietary copyrighted and/or legally
privileged. It is intended solely for the use of the individual or entity
named on this message. If you are not the intended recipient, and have
received this message in error, please notify us immediately by reply e-mail
and then delete it from your system.>---------------------------------------------------------------------------
--->_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------------------------------------------------------
Notice: This e-mail message, together with any attachments, contains
information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New
Jersey, USA 08889), and/or its affiliates (which may be known outside the United
States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan, as Banyu)
that may be confidential, proprietary copyrighted and/or legally privileged. It
is intended solely for the use of the individual or entity named on this
message. If you are not the intended recipient, and have received this message
in error, please notify us immediately by reply e-mail and then delete it from
your system.
------------------------------------------------------------------------------
dkwok
2004-Feb-24 00:03 UTC
[Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?
|I cannot call out with my SIP phone though. It'll dial, ring my cell |phone twice and then give up and complain that its busy. Even if I try |to answer the cell phone during the first ring. | |Does anyone have a config they could share with me on how to make this |setup work? This sounds like it should be fairly trivial, but I've |beaten my head against the wall on this for a few days. =) | |Thanks alot, |Jason Again most possibily it is codec issue, what sip phone you use and show us your sip.conf. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040223/830f2285/smime.bin