RE: [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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Sent: Monday, February 23, 2004 10:25 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 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:[EMAIL PROTECTED] [iaxtel700] exten => _91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) [trunkint] ; ; International long distance through trunk ; exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9011.,2,Congestion [trunkld] ; ; Long distance context accessed through trunk ; exten => _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91NXXNXX,2,Congestion [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9NXX,2,Congestion [trunktollfree] ; ; Long distance context accessed through trunk interface ; exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91800NXX,2,Congestion exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91888NXX,2,Congestion exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91877NXX,2,Congestion exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91866NXX,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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Sent: Monday, February 23, 2004 10:25 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 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 => _5,1,Dial,Zap/1/800551212 T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Sent: Monday, February 23, 2004 1:02 PM To: [EMAIL PROTECTED] 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:[EMAIL PROTECTED] [iaxtel700] exten => _91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) [trunkint] ; ; International long distance through trunk ; exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9011.,2,Congestion [trunkld] ; ; Long distance context accessed through trunk ; exten => _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91NXXNXX,2,Congestion [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9NXX,2,Congestion [trunktollfree] ; ; Long distance context accessed through trunk interface ; exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91800NXX,2,Congestion exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91888NXX,2,Congestion exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91877NXX,2,Congestion exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _91866NXX,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: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of Jason >Sent: Monday, February 23, 2004 10:25 AM >To: [EMAIL PROTECTED] >Cc: [EMAIL PROTECTED] >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,