Yeee-haaaa!

doesn't this look pretty?

*******************************************
Asterisk Ready.
*CLI> -- Accepting AUTHENTICATED call from 192.168.0.201, requested format = 4, actual format = 4
   -- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "IAX2/z2|20|tr") in new stack
   -- Called z2
   -- Call accepted by 192.168.0.202 (format ulaw)
   -- Format for call is ulaw
   -- IAX2/z2/2 is ringing
   -- IAX2/z2/2 answered IAX2/[EMAIL PROTECTED]/1
   -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/1 and IAX2/z2/2
   -- Channel 'IAX2/[EMAIL PROTECTED]/1' ready to transfer
   -- Channel 'IAX2/z2/2' ready to transfer
   -- Releasing IAX2/z2/2 and IAX2/[EMAIL PROTECTED]/1
   -- Hungup 'IAX2/z2/2'
 == Spawn extension (geograph, 202, 1) exited non-zero on 'IAX2/[EMAIL 
PROTECTED]/1'
   -- Hungup 'IAX2/[EMAIL PROTECTED]/1'
   -- Registered 'z1' (AUTHENTICATED) at 192.168.0.201:4569
   -- Registered 'z2' (AUTHENTICATED) at 192.168.0.202:4569

*CLI>
*******************************************

Rich! Carlos! - Pizzas on me when you come to Cape Town. (or I get to where you are - where's that?)

What a learning curve - big thanks and let me give you a suggestion: Take leave the week after next - I'm going to be plugging in 2 internal ISDN BRI cards ;-) (next week will be to sort out the choppy sound & to move from my SuSE 9.3 toy box to the production FC3 (dare I try FC4) box - so maybe take next week off as well :-)

(Carlos - I'll respond to your zaprtc query later today)

Cheers & thanks - sincerely hope to be able to return the effort one day.
regards to all,
Zoltan


Rich Adamson wrote:

all0w=ulaw
all0w=alaw
all0w=gsm

Look closely at the above four lines. In the "allow" statement, that
appears to be a zero. Change that to "allow". Also, I don't know which codecs the phone supports, but you might start playing with
disallow=all
allow=ulaw
and go from there.
you're 100% right - I saw the typo when the lines were commented out and the codecs were in the [z1] section. I then changed back in order to shorten the iax.conf file but forgot about the typos. Thanks - it could've taken many more hours for me to notice them again :-)

[z1]
type=friend
user=z1
secret=z1
context=geograph
host=dynamic
dtmfmode=rfx2833

If you look at /usr/src/asterisk/configs/iax.conf.sample, you'll find that dtmfmode=rfx2833 is not a valid iax statement. Plus its
spelled wrong (its rfc2833). Remove it, but add it into your sip.conf
if you're going to play with sip.

jeeze - dislexia rulz (never change a config file when in a hurry to do something else)


*************** asterisks response as I dial ************
Asterisk Ready.
*CLI> iax2 show p
peers         provisioning
*CLI> iax2 show peers
Name/Username    Host                 Mask             Port      Status
z2               192.168.0.202   (D)  255.255.255.255  4569      Unmonitored
z1               192.168.0.201   (D)  255.255.255.255  4569      Unmonitored
*CLI> iax2 show users
Username         Secret                Authen           Def.Context
A/C
z2               z2                    000000000000003  geograph
No
z1               z1                    000000000000003  geograph
No
*CLI>     -- Accepting AUTHENTICATED call from 192.168.0.201, requested
format = 4, actual format = 256

Here is the key:     ^^^^^^^^^^^^
That is telling you it can't find a compatible codec to allow the
call to complete. That's the basis for the comments above about the
allow=ulaw.

    *-- Executing Dial("IAX2/[EMAIL PROTECTED]/3", "IAX/z2|20|tr") in new stack*

Note the above "IAX". I think that should be IAX2, so look in your
extensions.conf for a dial statement that looks like "Dial(IAX/"
and change it to "Dial(IAX2/".

Yep - this too would have taken me a while to notice.


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