Talkin2ya

This page gives a quick howto for setting up an outbound connection using Asterisk with Talkin2ya; it works in at least two setups.

First, add an entry for budgetphone in sip.conf (replace all occurences of 31707110311 by your assigned number)
register => 31707110311:XXX@budgetphone.nl/31707110311

[31707110311]
type=friend
context=from-budgetphone
host=budgetphone.nl
fromuser=31707110311
fromdomain=budgetphone.nl
username=31707110311
insecure=very
nat=yes
secret=XXX
qualify=no
port=5060

Reload your SIP config using the command 'sip reload' and asterisk should register itself with the remote server. You can double check this by dialling your number using a landline; asterisk should now receive a call.

To use it, you need to add some extensions; right now, I have the following in my extensions.conf for outbound calls (it probably is a bit baroque but it works for me)
[to-budgetphone]
exten => _9.,1,Set(CALLERID(num)=31707110311)
exten => _9.,2,Set(CALLERID(name)=Jasper Spaans)
exten => _9.,3,Dial(SIP/0${EXTEN:1}@31707110311)
exten => _9.,4,Hangup()

Setting up an incoming context is interesting: budgetphone / talkin2ya uses (at the moment) a cluster of 3 servers. An incoming call might not come from the sip server you just registered with. The solution is to recognize calls for your budgetphone number in the context for sip guest users.

Given the following in sip.conf general section:
[general]
context=sipincoming ; Default context for incoming calls

You will something like the following in extensions.conf:

; sip guest?
[sipincoming]
exten => 31707110311,1,Goto(incomingcall,s,1)

[from-budgetphone]
exten => 31707110311,1,Goto(incomingcall,s,1)


Incoming calls with witheld callerid will show as CALLERID(num)="anonymous"

If the registration doesn't work, it might be that your Asterisk server is not using SIP srvlookups. This is enabled by default, so double check your sip.conf for lines containing
srvlookup=no
.
If these are present, change them to
srvlookup=yes
in the relevant places!

If it still doesn't work, there might be a mismatch between the (RFC compliant) SRV records Talkin2ya uses and what Asterisk expects. A workaround for this is to add the following line to /etc/hosts on your asterisk-server:
81.23.228.150 budgetphone.nl


Now, all should work well.
This page gives a quick howto for setting up an outbound connection using Asterisk with Talkin2ya; it works in at least two setups.

First, add an entry for budgetphone in sip.conf (replace all occurences of 31707110311 by your assigned number)
register => 31707110311:XXX@budgetphone.nl/31707110311

[31707110311]
type=friend
context=from-budgetphone
host=budgetphone.nl
fromuser=31707110311
fromdomain=budgetphone.nl
username=31707110311
insecure=very
nat=yes
secret=XXX
qualify=no
port=5060

Reload your SIP config using the command 'sip reload' and asterisk should register itself with the remote server. You can double check this by dialling your number using a landline; asterisk should now receive a call.

To use it, you need to add some extensions; right now, I have the following in my extensions.conf for outbound calls (it probably is a bit baroque but it works for me)
[to-budgetphone]
exten => _9.,1,Set(CALLERID(num)=31707110311)
exten => _9.,2,Set(CALLERID(name)=Jasper Spaans)
exten => _9.,3,Dial(SIP/0${EXTEN:1}@31707110311)
exten => _9.,4,Hangup()

Setting up an incoming context is interesting: budgetphone / talkin2ya uses (at the moment) a cluster of 3 servers. An incoming call might not come from the sip server you just registered with. The solution is to recognize calls for your budgetphone number in the context for sip guest users.

Given the following in sip.conf general section:
[general]
context=sipincoming ; Default context for incoming calls

You will something like the following in extensions.conf:

; sip guest?
[sipincoming]
exten => 31707110311,1,Goto(incomingcall,s,1)

[from-budgetphone]
exten => 31707110311,1,Goto(incomingcall,s,1)


Incoming calls with witheld callerid will show as CALLERID(num)="anonymous"

If the registration doesn't work, it might be that your Asterisk server is not using SIP srvlookups. This is enabled by default, so double check your sip.conf for lines containing
srvlookup=no
.
If these are present, change them to
srvlookup=yes
in the relevant places!

If it still doesn't work, there might be a mismatch between the (RFC compliant) SRV records Talkin2ya uses and what Asterisk expects. A workaround for this is to add the following line to /etc/hosts on your asterisk-server:
81.23.228.150 budgetphone.nl


Now, all should work well.
Created by: jsp, Last modification: Wed 18 of Feb, 2009 (21:13 UTC) by koosvdhout
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