search for: phone2,20

...er VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid="Me" <2124> [phone2] type=friend ;secret=blah host=dynamic defaultip=192.168.1.107 dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid="Mini Me" <2123> And in extensions.conf at the very end: [sip] exten => 1,1,Dial(SIP/...

...am registered with FWD, and when I call the number from the BT100 I can see all the incoming information but still nothing on my X-Lite. My extensions.conf: [general] static=yes writeprotect=no [globals] [sip] exten => 1,1,Dial(SIP/phone1,20,tr) exten => 2,1,Dial(SIP/phone2,20,tr) exten => 2,2,VoiceMail,u1234 exten => 2,102,VoiceMail,b1234 ;exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr) exten => 1001,1,Ringing exten => 1001,2,Wait(2) exten => 1001,3,VoicemailMain,s1234 exten => 6601,1,WaitMusicOnHold(60) exten => 232999,1,Dial(SI...

...ice => /dev/ttyI1 (as found on another post to the list) In extensions.conf I have: [globals] TRUNK=Modem/ttyI0 [trunk] xten => _9XXXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}||Ttm) exten => _9XXXXXXXXXX,2,Congestion [sip] exten => 7201,1,Dial(SIP/phone1,20,Ttr) exten => 7205,1,Dial(SIP/phone2,20,Ttr) exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr) [s0bus] exten => s,1,Wait,1; exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => s,1,Dial(SIP/phone1&SIP/phone2,20,tr) Any advice would be...

What is the proper method to install/configure an X100P FXO card? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031009/ce8a7803/attachment.htm

...codecs allow=all ; Allow codecs in order of preference ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ..... [Phone1] type=friend host=dynamic defaultip=192.168.3.103 dtmfmode=rfc2833 context=from-sip callerid=" Win box " <1> [Phone2] type=friend host=dynamic defaultip=192.168.3.119 dtmfmode=rfc2833 context=from-sip callerid=" Deepak" <2> [Phone3] type=friend host=dynamic defaultip=192.168.3.106 dtmfmode=rfc2833 context=from-sip callerid=" Ravi " <3> [extensions.conf] [from-sip] exten=>1,1,Di...

Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very serious problem of modutils and iptable. Can anybody help me out. Thanx and Regards sanchal singh

Dear, When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14) hardware config: server - phone1 - phone2 - laptop configuratio...

Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. -Brian On Feb 11, 8:07am, "Juki" wrote: } Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins } Hi all, } } I have Asterisk running on FreeBSD 4.x and I have made configurations to }

...e, and each box connect to Asterisk using this softphone (X-Lite). Asterisk use the following configuration : /etc/asterisk/sip.conf ; Phone #1 [Phone1] type=friend host=dynamic nat=yes defaultip = 192.168.10.12 # windows box IP context = sip callerid="Phone1" <1> ; Phone #2 [Phone2] type=friend host=dynamic nat=yes defaultip = 192.168.10.5 # second windows box IP context = sip callerid="Phone" <2> i have the following extension : /etc/asterisk/extensions.conf [sip] exten => 1,1,Dial(SIP/Phone1,20,tr) exten => 2,1,Dial(SIP/Phone2,20,tr) One windows b...

I have the following asterisk configuration (sip.conf) : [General] externip=82.79.81.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone3" <3> disallow=all allow=gsm [Phone4] type=friend host=dynamic nat=yes qualify=yes conte...

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions, 1000 and 1001 after a long time with forbidden messages on phones. My questions are, 1. Do these phones need to register with the server 2. Where does the auth...

...Here is my sip.conf: [Phone1] type=friend host=dynamic ;defaultip=192.168.1.103 dtmfmode=rfc2833 context=SIP callerid = "Me" <2124> [Phone2] type=friend host=dynamic ;defaultip=192.168.1.101 dtmfmode=rfc2833 context=SIP callerid = "Mini Me" <2123> Following is my extensions.conf: ex...

I have an Asterisk 0.4.0 install working with two grandstream budgetone 100 phones, gnophone, and kphone. This is a private network segment (172.17.x.x), with the PBX configured on my outbound firewall which has a public address (66.x.x.x). - I can make calls between phones - all extensions are working. - I can make IAX calls to IAXTEL. No problems (apparently gsm only) - I can call SIP phone

...rong. We would sincerely appreciate any help/pointers. Thank you all Edward Banfa ******EXTENSION.CONF******* [general] static=yes [from-sip] exten => 100,1,Dial(SIP/edward,20) exten => 100,2,Hangup exten => 101,1,Dial(SIP/phone1,20) exten => 101,2,Hangup exten => 102,1,Dial(SIP/phone2,20) exten => 102,2,Hangup exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxx@NuFone/${EXTEN} exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN} *****IAX.CONF***** [general] port=5036 bind=0.0.0.0 bandwidth=low disallow=lpc10 [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx disallow=all a...

...rong. We would sincerely appreciate any help/pointers. Thank you all Edward Banfa ******EXTENSION.CONF******* [general] static=yes [from-sip] exten => 100,1,Dial(SIP/edward,20) exten => 100,2,Hangup exten => 101,1,Dial(SIP/phone1,20) exten => 101,2,Hangup exten => 102,1,Dial(SIP/phone2,20) exten => 102,2,Hangup exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxx@NuFone/${EXTEN} exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN} *****IAX.CONF***** [general] port=5036 bind=0.0.0.0 bandwidth=low disallow=lpc10 [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx disallow=all a...

Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to