Hi...
I've been fighting this for a while now, trying clean builds of Asterisk
1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today.
No workee. :-(
Here's the results for various calls made off-hook (push the blue
Speakerphone button on the Polycom 430):
988852700 - Phone waits for me to either hit the soft-key "Send" or
"EndCall". If I hit "Send", it dials through with no problem.
98168852700 - Before I get the last "0" pressed, the phone presents me
with a second dial tone and a prompt at the top of the screen, "Enter
more digits". Asterisk console presents
"== Using SIP RTP CoS mark 5"
917852963296 - Before I get the "96" pressed, results as immediately
above.
If I dial these numbers with the phone on-hook, and press "dial" they
work fine.
If I modify my dialplan to remove the dial nine requirement, all three
methods of dialing out, off-hook, work fine...although I do have to
press "Send" when dialing 8852700.
The seemingly relevant portion of the dialplan is as follows:
;********************************************************************
; BEGIN - Outbound Call Handling
;********************************************************************
;
[outbound-local]
exten => _9NXXXXXX,1,Dial(${TRUNK0}/${EXTEN:1})
exten => _9NXXXXXX,n,Congestion()
exten => _9NXXXXXX,n,Hangup()
exten => _9NXXNXXXXXX,1,Dial(${TRUNK0}/${EXTEN:1})
exten => _9NXXNXXXXXX,n,Congestion()
exten => _9NXXNXXXXXX,n,Hangup()
exten => 911,1,Dial(${TRUNK0}/911)
exten => 9911,1,Dial(${TRUNK0}/911)
[outbound-long-distance]
exten => _91NXXNXXXXXX,1,Dial(${TRUNK0}/${EXTEN:1})
exten => _91NXXNXXXXXX,n,Congestion()
exten => _91NXXNXXXXXX,n,Hangup()
[hang-up]
; Hang up
;
exten => s,1,Playback(thank-you-for-calling)
exten => s,n,Playback(goodbye)
exten => s,n,Hangup()
;
;
;********************************************************************
; END - Outbound Call Handling
;*******************************************************************
The only difference between the Asterisk versions is the presence on the
Asterisk console of an error message with Asterisk 1.4.18 and 1.4.19rc3,
which is similar to the one noted on the forums: "NOTICE[6145]:
chan_sip.c:13795 handle_request_invite: Failed to authenticate user
"6000" <sip:[EMAIL PROTECTED]>;tag=whatever it was" I do not see that
error message on the Asterisk console for 1.6 Beta 6.
The forums note which seems in the neighborhood is at
http://forums.digium.com/viewtopic.php?p=63872&sid=aff61bbd5ddeea61bc831
239b220db23
Anyone have any bright ideas on what might be wrong and/or
troubleshooting tips?
...brig
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Brig C. McCoy
IT Help Desk
ThyssenKrupp Access Corporation
4001 East 138th Street
Grandview, MO 64030 USA
Phone: +1 816-767-5577
Fax: +1 816-765-6459
Email: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
Internet: www.tkaccess.com <http://www.tkaccess.com/> www.thelev.com
<http://www.thelev.com>
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world's most trusted name in
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