VoIP Mailing List Archives

[Brig.McCoy at tkaccess...]

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:6000 at 10.10.xxx.xxx>;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

--

Please direct emails to ITHelpDesk at tkaccess.com
<blocked::mailto:ITHelpDesk at tkaccess.com> or call 816-767-5549. This
will help with issues getting full exposure to the dept and allow for
the quickest response.



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: Brig.McCoy at tkaccess.com <mailto:Brig.McCoy at tkaccess.com>

Internet: www.tkaccess.com <http://www.tkaccess.com/> www.thelev.com
<http://www.thelev.com>



"Committed to Improving the Quality of Life. ThyssenKrupp Access, the
world's most trusted name in
accessibility and home elevator solutions"


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[jjones at danrj.com]

What does your digitmap on your phone look like? This is what
controls sending the call to * when it recognizes a complete dial
pattern. The phone does not send digit by digit. If it is waiting for
you to press send, then it does not recognize your pattern.
On Mar 26, 2008, at 8:18 AM, Brig C. McCoy wrote: