VoIP Mailing List Archives

[mwatson at becon.org]

Might want to do a "sip set debug peer <peer id>"

You should then be able to see the sip packet dumps that are going between the phone and *. Might give you some clues.

--
Matt
http://www.mattgwatson.ca
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Henrik Ostergaard Madsen
Sent: Wednesday, May 28, 2008 12:54 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Calling '**1' through Asterisk

I am experiencing problems calling '**1' through astrisk from an VOIP
telephone. Asterisk appearently does not accept the call and never show
any indication of an incoming call (using asterisk -rvvvvvvvvvvvvvvvvvv).
On the phone, I get an immideate 'Call ended'. This occurs on both an
Linksysy SPA941 and an LIP TA100. The same happens with '**'

Using '**11' or '*1' or '*11' instead, the call gets right through..

I can see on tcpdump that the SIP packages does reach the asterisk server
and gets answered.

Does anyone have a clue on what is going on?

Regards

Henrik
-----------------------------------------------------------
Henrik ?stergaard Madsen Phone: +45 44 48 44 92
PhD, M.Sc. Cell: +45 30 94 02 88
Mosegard Park 42 email: Henrik at Ostergaard.net
DK-3500 V?rl?se WWW homepage:
Denmark http://www.Ostergaard.net/Henrik





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[Henrik at ostergaard.net]

I am experiencing problems calling '**1' through astrisk from an VOIP
telephone. Asterisk appearently does not accept the call and never show
any indication of an incoming call (using asterisk -rvvvvvvvvvvvvvvvvvv).
On the phone, I get an immideate 'Call ended'. This occurs on both an
Linksysy SPA941 and an LIP TA100. The same happens with '**'

Using '**11' or '*1' or '*11' instead, the call gets right through..

I can see on tcpdump that the SIP packages does reach the asterisk server
and gets answered.

Does anyone have a clue on what is going on?

Regards

Henrik
-----------------------------------------------------------
Henrik ?stergaard Madsen Phone: +45 44 48 44 92
PhD, M.Sc. Cell: +45 30 94 02 88
Mosegard Park 42 email: Henrik at Ostergaard.net
DK-3500 V?rl?se WWW homepage:
Denmark http://www.Ostergaard.net/Henrik

[brent at texascountryt...]

Check your features.conf file and make sure that combination or a
similar combination (*1 or ** ) for instance isn't defined in there for
some reason.
Matt Watson wrote:

[rizwanhasham at gmail.com]

seems like asterisk could not find the extension number "**1". does your
context "from-internal" have this extensions? if not then create one or
include the context which contains the extension **1 in from-internal and
then try.

On Thu, May 29, 2008 at 1:10 AM, Henrik Ostergaard Madsen <
Henrik at ostergaard.net> wrote:

[Henrik at Ostergaard.net]

No it is unfortunately not that. Calling '**11' in stead DOES come
out with an 'extension not found'. But '**1' does not come with ANY
output except for the sip logging. And I DO have '**1' defined in
the extensions. '**' does not produce any output either, and this is
not defiend in extensions. The sip package dump in the former post
was the entire output from Asterisk..

[stevej456 at gmail.com]

This suggests to me that you might have to make a change to the
phone's internal dialplan string. What is it now in the SPA941 (at
the bottom of the EXT1 tab on the phone's web configuration screen)?

S.

On Wed, May 28, 2008 at 10:53 AM, Henrik Ostergaard Madsen
<Henrik at ostergaard.net> wrote:

[joakimsen at gmail.com]

Asterisk 1.4.20.1
Polycom 501 v 3.0.1.0032

[May 30 00:12:54] NOTICE[26924]: chan_sip.c:14033
handle_request_invite: Call from 'SIP_PEER' to extension '**1'
rejected because extension not found.
[May 30 00:14:25] NOTICE[26924]: chan_sip.c:14033
handle_request_invite: Call from 'SIP_PEER' to extension '**' rejected
because extension not found.

I did not even modify the dial plan on the phone, just press **1 and
dial. Make sure the configuration (dialplan) on the Linksys is
correct, see http://spc.pifiu.com for details.


On Thu, May 29, 2008 at 7:59 AM, Henrik ?stergaard Madsen
<Henrik at ostergaard.net> wrote:

[Henrik at Ostergaard.net]

[blue at cmd.nu]

Hello.

On my Linksys SPA961 I had to change the dialplan string in order for **X to
work.

Greetings,

On Fri, May 30, 2008 at 7:12 AM, Henrik ?stergaard Madsen <
Henrik at ostergaard.net> wrote:

[Henrik at ostergaard.net]

I have now upgraded to the latest Asterisk 1.4.20.1 and add-ons 1.4.6 without any luck - so I
started dismantling my dialplan.

It seems that the incoming catch:
exten => _**[1234567890].,1,NoOp(Incoming call to a quickdial number ${EXTEN})
does NOT work with **1 - in stead of printing some debug, it just closes the channel without
any warning, error or the like.. But it does work with **11!

Changing it to
exten => _**[1234567890],1,NoOp(Incoming call to a quickdial number ${EXTEN})
makes it work (with **1 etc - not **11)

- so I have made it do what I want, even as I still think it is a bug..

/Henrik

[rob at hillis.dyndns.org]

Not a bug at all. "." as a pattern match means "1 or more digits" - so
your pattern match means ** followed by a digit between 0 and 9,
followed by one or more digits. Removing the "." means that you then
match **, followed by a digit between 0 and 9.

If you want to match ** followed by a digit between 1 and 9, followed by
zero or more digits, the use the extension pattern "_**X!". X matches
any digit from 0-9, so your pattern match is unnecessary. ! matches
zero or more digits.

Henrik Ostergaard Madsen wrote:

[Henrik at Ostergaard.net]

Argh - you are perfectly right. I have been too far away from the
dialplan for too long

Thanks for the help!

/Henrik