search for: phones1vm

...receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten => _9.,1,Dial(Zap/1/$EXTEN:1) [flat] include => home include => outgoing zapata.conf contains the following - I have 2 x101p cards installed [channels] language=en group=1 context=from-analog signalling=fxs_ks usecallerid=no echocancel=yes...

...me secret=******** callerid="OFFICE PHONE #2" <7890> mailbox=7890 dtmfmode=rfc2833 nat=0 AND HERE IS MY EXTENSIONS.CONF FILE [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp PHONES1=SIP/7890 ; Phone 1 Def PHONES1VM=7890 ; Phone 1 VM Def FWDUSERID1=691657 MYNAME1=My name MYPHONE1=691657 TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [fwd-forced-fwd1] ; Check to see if the called number starts with a "7" and ; if so, set the call parameters and bounce the call to the ; Free...

...sometimes and other times it just goes straight to a busy signal. Despite the fact that outgoing works some of the time I'm wondering if I have the configuration wrong. Here is the outbound fragment from extensions.conf. [outbound] ; Press * to reach voice mail exten => *,1,VoiceMailMain(${PHONES1VM}) exten => *,2,Hangup ; local calls exten => _NXXXXXX,1,Playback(transfer) ; "Please hold while..." exten => _NXXXXXX,2,Dial(SIP/${EXTEN:0}@phone.gedanken.org:5061,20) exten => _NXXXXXX,3,Congestion ; long distance exten => _1NXXNXXXXXX,1,Playback(transfer) ; "Ple...

...al] static=yes writeprotect=no [globals] TRUNK=Gradwell TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) PHONES1=SIP/2201 [flat] include => home include => outgoing [home] exten => 2201,1,Dial(${PHONES1},20,Ttm) exten => 2201,2,Macro(vmessage,${PHONES1VM}) exten => 2201,3,Hangup [outgoing] ignorepat => 9 ignorepat => 8 exten => _9.,1,Dial(SIP/${EXTEN:1}@Talklite) exten => _8.,1,Dial(SIP/${EXTEN:1}@Gradwell) =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- linux:/etc/asterisk # tethereal -R "sip" Capturing...

...e> - <extension 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] e...

...e> - <extension 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] e...

...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 question is whether or not the same thing will work for...

...NEUSERID}) exten => _3.,2,SetCIDName(${MYNAME}) exten => _3.,3,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com) exten => _3.,4,Playback(invalid) exten => _3.,5,Hangup [from-sipphone] exten => ${SIPPHONEUSERID},1,Dial(${PHONES1},30,Ttm) exten => ${SIPPHONEUSERID},2,Voicemail2(u${PHONES1VM}) exten => ${SIPPHONEUSERID},3,Hangup -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050103/8e48fa14/attachment.htm

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

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

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:

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