Sip.conf : ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding "nat=1" to each peer definition to ; solve translation problems.
[general] bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying "The number you have dialed is not in service. Please check the ; number and try again." context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown tos=0x68 ; #, in this configuration file, is NOT A COMMENT. This is exactly ; how it should be. #include sip_nat.conf #include sip_registrations_custom.conf #include sip_registrations.conf #include sip_custom.conf #include sip_additional.conf I am calling other external phones, I think they PSTN destinations. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax: 972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson Sent: Thursday, January 24, 2008 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help: dtmf mode Please post your sip.conf entry for your phone and also describe your calling path. Are you having a problem with internal calls (e.g.: to voicemailmain) on the same switch, or are you referring to calls to PSTN destinations via pots/pri/sip/? Also, which versions of Asterisk, Zaptel, linux, etc. are you using? S. On Jan 24, 2008 12:43 PM, Jarga Jallow <[EMAIL PROTECTED]> wrote: > > > > > Hi, > > I am having trouble making a selection when I call a number and need to make > a selection to go to an extension with my polycom phones 301. Anybody have > an idea how to fix this problem? > > Thanks in advance. > > > > > Jarga Jallow > > Technical Support Engineer > > 2985 S. Hwy. 360 > > Grand Praire, Texas 75052 > > Direct: 972-206-1212 ext# 29 > > Mobile: 214-669-9046 > > Fax: 972-999-4113 > > Toll Free: 1-877-801-5511 ext 34 > > Toll Free: 1-877-926-2288 > > > > www.2mcctv.com > > > _______________________________________________ > -- 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 _______________________________________________ -- 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
