Differences between version 5 and revision by previous author of AsteriskItalkSetup.
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| Newer page: | version 5 | Last edited on Tuesday, June 2, 2009 7:17:41 pm | by DavidIvory | Revert |
| Older page: | version 4 | Last edited on Tuesday, July 3, 2007 3:38:06 pm | by GerwinVanDeSteeg | Revert |
@@ -1,5 +1,5 @@
-[iTalk|http://www.italk.co.nz/] is a service run by [Auckland] [ISP] [Slingshot|http://www.slingshot.co.nz/]. It provides a SIP gateway to the NZ PSTN for $10/month. "local" calls and calls to other iTalk users are free, with reasonably cheap rates after that. Currently only 09 (
Auckland)
PSTN numbers are
available, so local is only Auckland, but it is expected that this will expand to cover other parts of the country
.
+[iTalk|http://www.italk.co.nz/] is a service run by [Auckland] [ISP] [Slingshot|http://www.slingshot.co.nz/]. It provides a SIP gateway to the NZ PSTN for $10/month. "local" calls and calls to other iTalk users are free, with reasonably cheap rates after that. Check the iTalk website for current availability of non -
Auckland PSTN numbers. Number portability is also
available.
The best thing about iTalk is that it is built on open protocols such as [SIP], making it easy for you to integrate it with [Asterisk].
In order to make your [Asterisk] pbx "talk" to iTalk, you need to define an instance for it in your sip.conf so you can reference it to dial outside numbers eg.
@@ -11,16 +11,22 @@
username=649974xxxx
fromuser=649974xxxx
host=akl.italk.co.nz
dtmfmode=rfc2833
- insecure=very
;important
+ ;
insecure=very ; deleted
+ insecure=port,invite
; added
nat=yes
canreinvite=no
disallow=all
allow=ulaw
- allow=gsm
+ allow=alaw ; see note further down
+ ;
allow=gsm ; deleted
</verbatim>
+
+When upgrading to the current version of Asterisk (Asterisk 1.6.0.6 ) I found that calls into iTalk were being rejected and callers were hearing a voice "Currently Unavailable". This even when outbound calls were fine. Looking at the logs I understood that the insecure=very line was being flagged as an error.
+
+I have amended the sip.conf file with the offending lines commented out. Earlier versions of Asterisk may still work with insecure=very and not with the updated version - but you should be using the latest. Also allow=gsm has been deleted.
On 21 March 2007 I found that I needed to add the following to my sip.conf
<verbatim>
allow=alaw
@@ -31,10 +37,9 @@
<verbatim>
register => $username:$password@akl.italk.co.nz/$extension_to_forward_incoming_calls_to
</verbatim>
-
-inside
extensions.conf (aside from having a valid extension to forward incoming calls to as specified in the register line above) you need to have rule to push egress calls out the "iTalk" sip define.
+Inside
extensions.conf (aside from having a valid extension to forward incoming calls to as specified in the register line above) you need to have rule to push egress calls out the "iTalk" sip define.
I route all egress calls out via it when people 'dial 1 to get out' so my rule inside extensions.conf looks like
<verbatim>
@@ -46,10 +51,9 @@
<verbatim>
exten => _09X.,1,Dial(SIP/iTalk/${EXTEN:1},30,Tr)
</verbatim>
-
-And when
[Slingshot/iTalk|http://italk.co.nz] allow
registering other area codes you could do your own toll bypass network all for only $10/areacode/month for unlimited calls ;)
+[Slingshot/iTalk|http://italk.co.nz] now allows
registering other area codes so
you could do your own toll bypass network all for only $10/areacode/month for unlimited calls ;)
ie.
have a declaration for each area code in sip.conf, then route via them based on what the user dials eg.
