Hello,
I’m new to asterisk and I’m trying to what should be a very simple task - make a call from my sip phone to an outside party using Asterisk as the gateway.
Asterisk version : 11.3.0
Running on: Ubuntu 12.10 X86_64
Service subscribed to: Voicepulse
My SIP phone seems to register fine with the asterisk server (no errors and says “reachable”).
I downloaded the quickstart examples from voicepulse and thought I made the correct changes to them - obviously, I didn’t.
When I try to make a call out from my sipphone, I get the following error on the CLI:
[Apr 7 14:20:09] WARNING[5271][C-00000009] chan_sip.c: Purely numeric hostname (), and not a peer--rejecting!
[Apr 7 14:20:09] WARNING[5271][C-00000009] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 7 14:20:09] WARNING[5271][C-00000009] pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (GotoIf(1?14045511212|GatewayB))
Here’s my sip.conf: (user/password edited)
[code] ;sip.confSample.txt
; Sample /etc/asterisk/sip.conf
;
;
; *** Trixbox and FreePBX users DO NOT USE THIS FILE
; *** See your account center for guides specific to your PBX
;
;
; Updated Feb 24, 2011 for general enhancements
; Updated Jun 13, 2008 to fix inbound legacy calls
; Updated Jul 9, 2008 to migrate to sjc/jfk
; Updated Aug 8, 2008 to be more SIP RFC compliant
;
; Copyright (c) 2003-2011 VoicePulse Inc.
; VoicePulse is a registered trademark of VoicePulse Inc.
;
; =========================================================
; QUICKSTART WITH VOICEPULSE CONNECT FOR ASTERISK
;
; * Login to your VoicePulse SIP PBX account:
; https://www.voicepulse.com/secure/login.aspx
;
; * Go to the Setup tab to see your device login and
; device password. Your device login and passwords are
; NOT THE SAME AS YOUR WEBSITE LOGIN AND PASSWORD.
;
; * Do a text search & replace in this file:
; - Replace MY_DEVICE_LOGIN with your device login
; - Replace MY_DEVICE_PASSWORD with your device password
;
; =========================================================
; ---------------------------------------------------------
; GENERAL SETTINGS
;
; ---------------------------------------------------------
; .........................................................
; REGISTER WITH VOICEPULSE
;
; You should register to the datacenter closest to you.
; There are currently two datacenters you may use:
; * New York, NY (JFK)
; * San Jose, CA (SJC)
;
; The entire "register =>" line below should be on one line
; (with no carriage returns in the middle):
; .........................................................
[general]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=yes
canreinvite=no
; Register to JFK by default
;
; West coast users: replace "jfk" with "sjc" in the following two lines,
; and again in the peer entries below.
register => user:password@jfk-primary.voicepulse.com
register => user:password@jfk-backup.voicepulse.com
; ---------------------------------------------------------
; SIP PEERS
;
; These are the primary and backup peers for calls to/from:
; JFK (New York, NY) or SJC (San Jose, CA)
; ---------------------------------------------------------
[voicepulse-primary]
type=peer
context=voicepulse-in ; <-- the context in extensions.conf
; that you want these calls to go to
host=jfk-primary.voicepulse.com ; <-- west coast users should
; use sjc-primary instead
username=user
secret=password
qualify=yes
allow=all
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes
; ---------------------------------------------------------
; REGISTERED USER -- FOR TESTING SIP ENDPOINTS
;
; This is a test user. You can use Counterpath's X-Lite
; Phone to test your Asterisk configuration. Configure the
; following settings:
;
; Enabled: Yes
; Display Name: sipuser
; User Name: sipuser
; Authorization User: sipuser
; Password: sippassword
; SIP Proxy: <your Asterisk server IP address>
;
; You can get X-Lite at:
; http://www.counterpath.com/x-lite.html
; ---------------------------------------------------------
[niall]
type=friend
host=dynamic
secret=1234 ; Change this to something secure!
context=outgoing
canreinvite=no
allow=all
deny=0.0.0.0/0.0.0.0 ; These entries ensure this extension can
permit=10.0.0.0/255.0.0.0 ; only be used from your internal network.
permit=172.16.0.0/255.240.0.0 ; If your extensions are on a remote network,
permit=192.168.0.100/255.255.0.0 ; update these accordingly.[/code]
Here’s my extensions.conf:
[code] ; Sample /etc/asterisk/extensions.conf
;
;
; *** Trixbox and FreePBX users DO NOT USE THIS FILE
; *** See your account center for guides specific to your PBX
;
;
; Updated Feb 24, 2011 for general enhancements
; Updated Jun 24, 2008 to fix bug with int’l calls
; Updated Jul 9, 2008 to support jfk/sjc
; Updated Aug 8, 2008 to reflect changes in sip.conf
;
; Copyright © 2003-2011 VoicePulse Inc.
; VoicePulse is a registered trademark of VoicePulse Inc.
;
; =========================================================
; QUICKSTART WITH VOICEPULSE CONNECT FOR ASTERISK
;
; * Test your incoming and outgoing calls using the test
; programs mentioned in sip.conf
;
; * Incoming calls should ring the extension set up in
; sip.conf
;
; * Test outgoing calls by dialing all 11 digits, e.g.
; 17323395100
;
; * After testing, modify the OUTGOING CONTEXT and
; INCOMING CONTEXT per your requirements.
;
; =========================================================
; ---------------------------------------------------------
; GENERAL SETTINGS
;
; ---------------------------------------------------------
[general]
static=yes
writeprotect=no
; ---------------------------------------------------------
; OUTGOING CONTEXT
;
; [outgoing] is the context referred to by the test user
; [sipuser] in sip.conf.
;
; This is where your custom outgoing call processing should
; go.
;
; Outgoing calls should be dialed in proper e164 format:
; +<countrycode><number>
;
; For example:
; +17323395100 or
; +441234567890
; ---------------------------------------------------------
[globals]
[outgoing]
; .........................................................
; NANPA calls (US, Canada, Bermuda, parts of the Caribbean)
; .........................................................
; Set your CallerID number
; Setting both the name and number is required due to
; non-standard SIP behavior by previous versions of Asterisk.
;
; The CallerID number must be 10 digits.
exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=0000000000)
exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=0000000000)
; Print the SIP Call-ID to the CLI for troubleshooting.
; You can send this Call-ID to customer service when asked
; for a call example that may have experienced an issue.
exten => _1NXXNXXXXXX,n,NoOp(SIPCALLID: ${SIPCALLID})
; Send your call to VoicePulse using SIP
exten => _1NXXNXXXXXX,n,Dial(SIP/niall+${EXTEN}@${jfk-primary.voicepulse.com})
exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|GatewayB)
exten => _1NXXNXXXXXX,n(GatewayB),Dial(SIP/niall+${EXTEN}@${jfk-primary.voicepulse.com})
; .........................................................
; International calls
; .........................................................
; Set your CallerID number
; The CallerID number must be 10 digits.
exten => _011XXXX.,1,Set(CALLERID(num)=0000000000)
; Print the SIP Call-ID to the CLI for troubleshooting.
; You can send this Call-ID to customer service when asked
; for a call example that may have experienced an issue.
exten => _011XXXX.,n,NoOp(SIPCALLID: ${SIPCALLID})
; Send your call to VoicePulse using SIP
;
; Note the "${EXTEN:3}" in the Dial statement, which removes
; the "011" from the dialed number before sending to our gateway.
exten => _011XXXX.,n,Dial(SIP/+${EXTEN:3}@${VOICEPULSE_GATEWAY_OUT_A})
exten => _011XXXX.,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|GatewayB)
exten => _011XXXX.,n(GatewayB),Dial(SIP/+${EXTEN:3}@${VOICEPULSE_GATEWAY_OUT_B})
; ---------------------------------------------------------
; INCOMING CONTEXT
;
; [voicepulse-in] is the context referred to by the
; [voicepulse-primary] peer in sip.conf and the
; [voicepulse-backup] peer in sip.conf
;
; This is where your custom incoming call processing should
; go.
;
; IMPORTANT: Incoming calls from VoicePulse will have an
; extension of country code + number, so calls from US
; numbers will be 1 + area code + number (11 digits).
; ---------------------------------------------------------
[voicepulse-in]
; .........................................................
; This section catches calls coming from VoicePulse.
;
; The extension used below will catch your incoming calls
; regardless of what phone numbers are on your VoicePulse
; Connect for Asterisk account:
;
; exten => _XX.
;
; If you would like to route your calls based on different
; incoming phone numbers (YOUR numbers, not the caller's
; number), use:
;
; exten => _17325550000,1,Dial(SIP/sipuser)
; exten => _17325550001,1,Dial(SIP/john)
; ...
; ...
; ...
;
; For sample purposes, this section will ring your test
; SIP phone. For this to
; work:
;
; - You must have a test phone set up using the test user
; found at the end of sip.conf
; - You must order a phone number on your account
; - You must dial that phone number from a different phone
;
; .........................................................
exten => _XX.,1,NoOp(Call received from VoicePulse)
exten => _XX.,n,Dial(SIP/niall)
; ---------------------------------------------------------
; BANNED CONTEXT
;
; This context is used by unauthorized incoming or
; outgoing calls
; ---------------------------------------------------------
[banned]
exten => i,1,Hangup
exten => t,1,Hangup
[/code]
I know I’m missing something simple, but clearly not simple enough for me to grasp at the moment.
Thanks is advance for any assistance you can provide.
Sincerely.
Gerald