Vic Cross wrote:
On Sat, 7 Feb 2004, John Fraizer wrote:
snip all the trace data
Here are the configs:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 66.35.64.38 ; Address to bind to
context = default ;
u probably should upgrade to 0.7.2 but as far as the caller id that
would be from your sip.conf being set improperly add to your sip.conf
callerid=Caller Name # for each sip entry and that should clear it
up.
On Sat, 2004-02-07 at 00:23, John Fraizer wrote:
I'm running Asterisk 0.5.0 and using
Hi William,
Thanks for the reply. I don't understand why that information should have
to be set in sip.conf. It is already known. If I set the following on an
extension:
exten = 100,1,SetCallerID(${CALLERIDNUM})
I can see that the information is correct:
-- Executing
OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf.
The behavior is the same.
Caller-ID is sent as Name of Calling Party number of CALLED party
instead of Name of Calling Party number of CALLING party like it should be.
If you look at the sip debug of a call between
quote who=John Fraizer
OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf.
The behavior is the same.
Caller-ID is sent as Name of Calling Party number of CALLED party
instead of Name of Calling Party number of CALLING party like it should
be.
You are not setting the
quote who=Robert Hajime Lanning
quote who=John Fraizer
OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf.
The behavior is the same.
Caller-ID is sent as Name of Calling Party number of CALLED party
instead of Name of Calling Party number of CALLING party like it
Robert Hajime Lanning wrote:
quote who=John Fraizer
OK. I upgraded to 0.7.2 but and also set a callerid= entry in sip.conf.
The behavior is the same.
Caller-ID is sent as Name of Calling Party number of CALLED party
instead of Name of Calling Party number of CALLING party like it should
be.
quote who=John Fraizer
Um, yes I am setting the caller ID right. Asterisk isn't sending the invite
message properly.
[100]
callerid= test name 1234
type=friend
username=100
secret=secret
host=dynamic
fromuser=100
mailbox=100
context=allaccess
canreinvite=yes
dtmfmode=rfc2833
I just set up an X-Lite client on extension and it shows the same
behavior.
Further information:
When I call the X-Lite extension from my Cisco 7960 (extension 100),
I get the following in the recent calls list:
Name: Test
SIPURL: [EMAIL PROTECTED]
ProxyID: ENTERZONE
The sip.conf
quote who=John Fraizer
If your asterisk server does not do this, please do me the favor of setting
up two test extensions for me so I can try to figure out what is wrong
here. You can lock me in a context where I can only call from one test
extension to another. I just need to be able to
Robert Hajime Lanning wrote:
That is real interesting. It seems to work just fine for me. Though, I am
running straigh out of CVS, but older than 0.7.2 release. My SIP phones
(Grandstream) see CallerID just fine and my co-worker's SIP phones (Cisco 7960)
work also.
Can you send your
Would like to see a SIP debug
* The invite from the caller phone to Asterisk
* The invite from Asterisk to the called phone
As well as the configs (extensions.conf and sip.conf)
Can't reproduce in my servers.
/O
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Olle E. Johansson wrote:
Would like to see a SIP debug
* The invite from the caller phone to Asterisk
* The invite from Asterisk to the called phone
As well as the configs (extensions.conf and sip.conf)
Can't reproduce in my servers.
/O
OK. Here is a call from extension 100 to extension
Robert Hajime Lanning wrote:
I can do this, hold on.
OK. I don't know what the deal is. Works fine on your server. Doesn't on
mine.
That is so strange.
John
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quote who=John Fraizer
OK. I don't know what the deal is. Works fine on your server. Doesn't on
mine.
That is so strange.
my version string is: CVS-01/31/04-04:24:34
Also, I noticed that your sip.conf entries are a bit different than mine.
I am curious if canreinvite=no would change your
On Sat, 7 Feb 2004, John Fraizer wrote:
snip all the trace data
Here are the configs:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 66.35.64.38 ; Address to bind to
context = default ; Default for
I'm running Asterisk 0.5.0 and using Cisco 7960 phones in a sip only
configuration currently. Everything is working except that caller ID is hosed.
Say for example extension 100 calls extension 200. 200 sees 100 as the
name but 200 as the number. IE, it gets its own number as the supposed
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