search for: phones2

Can you also post information such as: Type of phone (model Number would be idela) How is it conencted, SIP, ZAP, IAX, Channel Bank. Corresponding config files would also help. Help us help you. >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Paul A Brown >>Sent:

...nsion number> ;------------------------------------------------ [2201] type=friend host=192.192.192.220 context=home secret=xxxxxx callerid="Paul" <2201> mailbox=2201 dtmfmode=rfc2833 nat=no EXTENSIONS.CONF writeprotect=no [globals] PHONES1=SIP/2201 PHONES1VM=2201 PHONES2=SIP/2202 PHONES2VM=2202 CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [iaxtel700] exten => _91...

...nsion number> ;------------------------------------------------ [2201] type=friend host=192.192.192.220 context=home secret=xxxxxx callerid="Paul" <2201> mailbox=2201 dtmfmode=rfc2833 nat=no EXTENSIONS.CONF writeprotect=no [globals] PHONES1=SIP/2201 PHONES1VM=2201 PHONES2=SIP/2202 PHONES2VM=2202 CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [iaxtel700] exten => _91...

...ers I know that I can simply have one registration statement along with an inbound context, then in extension.conf map the outbound context. from iax.conf: register => in-xxxxxxxx:xxxxxxxx@gw5.voicepulse.com from extensions.conf [voicepulse-in] exten => 212xxxxxxx,1,Dial(${PHONES1}&${PHONES2},30) exten => 212xxxxxxx,2,Voicemail2(u${PHONES1VM}) exten => 212xxxxxxx,3,Hangup I know this way I only have to register once, but can receive calls on several inbound DID numbers without any problem, provided they are all mapped similar to what I have above within extensions.conf. My qu...

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

...{DB_EXISTS(blacklist/${CALLERID(number)})}?custom-blacklisted,s,1) exten => s,4,Set(DB(CALLTRACE/lastcaller)=${CALLERID(number)}) exten => s,5,AGI(MisterHouse.agi,"CallerID") exten => s,6,Answer exten => s,7,Playtones(ring) exten => s,8,Dial(${PHONES0}&${PHONES1}&${PHONES2}&${PHONES7}&${PHONES11},20,tr) exten => s,9,Goto(s-${DIALSTATUS},1) ; if no fax, branch on dialstatus exten => s-NOANSWER,1,Macro(voicemail,${PHONES0VM}) exten => s-NOANSWER,2,Hangup() exten => s-BUSY,1,Macro(voicemail,${PHONES0VM}) exten => s-BUSY,2,Hangup() exten => _s-....

Hi, I wish to use Asterisk as a SIP server. How do I use Asterisk in a redundent network? So, if one Asterisk server fails, how does failover work? James