search for: phone1,20

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another 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....

Hi all I have registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones. i configured the extensions.conf file in both the server. the extensions.conf file on server 192.168.0.9 is exten=>301,1,Dial(SIP/301@192.168.0.6,20,tr) exten=>401,1,Dial(SIP/phone1,20,tr)...

...mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all 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...

...When I sip debug, I can see that I 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,WaitMusic...

...th 238 on 8/24/06 ;exten => s,n,Dial(Sip/201&Sip/209&Sip/211|20) exten => s,n,Dial(Sip/238&Sip/209&Sip/211|20) exten => s,n,Dial(Sip/227&Sip/225&Sip/213|20) exten => s,n,Goto(ivr,s,1) exten => _2XX,1,Macro(extensions,${EXTEN}) exten => 1234,1,Dial(Sip/phone1,20) ;Aastra 480iCT [closed] exten => s,1,Goto(ivr,s,1) [ivr] exten => s,1,Answer exten => s,n,Set(LOOPCOUNT=0) exten => s,n(begin),Set(TIMEOUT(digit)=3) exten => s,n,Set(TIMEOUT(response)=10) exten => s,n,Background(aa_1) exten => s,n,WaitExten(10) exten =&gt...

...56789,123456780 device => /dev/ttyI0 device => /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&...

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 c...

...o my iptel account??? I have try to this configuration, but it doesn't work: In sip.conf: register => my_account_name:xxxx@iptel.org [iptel.org] type=friend host=iptel.org fromuser=my_account_name secret=xxxx nat=yes in extensions.conf: [fromiptel] exten => my_iptel_number,1,Dial(SIP/phone1,20,r) [toiptel] exten => _002.,1,SetCallerId,my_iptel_number exten => _002.,2,Dial(SIP/my_iptel_number@iptel.org/${EXTEN:3},60,r) exten => _002.,3,Congestion What is it wrong???? Note: I'm sorry for my english.

I have an OpenBSD 3.7 gateway. This gateway run Asterisk. I have two windows box which use X-Lite softphone, 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 f...

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 conte...

Dear All, I installed an Asterisk on a linux PC, and X-Lite on two Windows PCs, all in a LAN. But, when I make phone call from one X-Lite to another, I always get Call Failed: 404 not found. 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...

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

...nt. I think I am having a codec problem here. What am I doing wrong. 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]...

...nt. I think I am having a codec problem here. What am I doing wrong. 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]...

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 regist...