>Also, I'm sure you've probably checked on this one, >but are the phones registered with asterisk? >You can make outbound calls on them without them >actually being registered. I'm assuming you can >still get in and see the CLI. What does "sip show peers" >look like? What does "sip show peer xxx" show? >What does the CLI show when you dial a phone?
The Phones are registered with Asterisk. They ring each other, through Asterisk, but DIALSTATUS appears as NOANSWER even when you lift the handset The CLI Log is here when Extension 50 Called 51. VAR: agi_request: init.sh VAR: agi_channel: SIP/50-6c72 VAR: agi_language: en VAR: agi_type: SIP VAR: agi_uniqueid: asterisk-28947-1110484788.110 VAR: agi_callerid: 50 <50> VAR: agi_dnid: 51 VAR: agi_rdnis: unknown VAR: agi_context: default VAR: agi_extension: 51 VAR: agi_priority: 1 VAR: agi_enhanced: 0.0 VAR: agi_accountcode: Detected protocol 'sip' ... 200 result=1 Detected caller '50' ... 200 result=1 Set limit - 24 200 result=1 Limit not exceeded (3 < 24) for localextensions 200 result=1 Set limit - 5 200 result=1 Limit not exceeded (3 < 5) for 50_out 200 result=1 Detecting destination for '51' ... 200 result=1 Found Destination localextensions (range 51 - 51) 200 result=1 Setting destination 'localextensions' ... 200 result=1 This is local extension, skipping Time Based Dialing/miniLCR ... 200 result=1 Set limit - 24 200 result=1 Limit not exceeded (4 < 24) for localextensions 200 result=1 Detecting Vertical Services ... 200 result=1 Set limit - 1 200 result=1 Limit not exceeded (1 < 1) for 51_in 200 result=1 Checking for channel SIP/51 ... 200 result=1 APP: exec ChanIsAvail SIP/51 200 result=0 Dialing '51' ... 200 result=1 APP: exec Dial SIP/51|32|tr 200 result=0 APP: get variable DIALSTATUS 200 result=1 (NOANSWER) Dialing Voicemail 51 ... 200 result=1 APP: exec Voicemail u51 200 result=-1 APP: answer 200 result=0 Playing macro 'vm-goodbye' ... 200 result=1 APP: stream file vm-goodbye 200 result=-1 endpos=6880 APP: hangup -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Thursday, March 10, 2005 4:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Polycom phones do not talk to each other >> We have bought PBXware GUI from Bicom systems and configured >> extensions with Polycom Phones as UAs. >> >> The Polycom Phones can dial out and make calls but I cannot make >> extension to extension calling. >> >> Googling did not help much. >> >> As you may be aware PBXware is a closed source software GUI from >> Bicom Systems for configuring extensions. It is a good tool to >> configure and manage users and phones but it does not allow to do any >> of the customization tasks that are possible by directly editing the >> .conf files, which may be required in for Polycom. >> >> However if this is an issue of configuration on the Phone itself, we >> want to be able to make changes and fix this problem. > Never used pbxware, but the context the sip phones dial out using > specified in sip.conf needs to include the dialplan context of the > phones in extensions.conf. -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. _______________________________________________ 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
