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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Henrik Ostergaard Madsen
2008-May-28 16:53 UTC
[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
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:> 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 > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >