Hi.
I'm not being able to make my Voicetronix Openswitch 12 work with
Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is
ringing, the Asterisk says that it is ringing, but the phone I'm ringing is
not ringing. I've seen in the mail list that other people have had the
problem that chan_vpb.c is making a call before hearing the dialtone. The
suggestioin was to put a comma or more before the number and this would make
a pause before actually dialing the number. This seemed to be a probable
cause of my problems, so I've defined in extesnsions.conf:
[globals]
OUTDIAL=vpb/1-9/,,3487446196
[default]
exten => _55.,1,Dial(${OUTDIAL},30,r)
but this doesn't work, does someone have suggestions?
Tim
_________________________________________________________________
Hitta r?tt p? n?tet med MSN S?k http://search.msn.se/
Check the extensive thread regarding this EXACT ISSUE in the mailing list archives. On Thu, 2004-03-18 at 04:36, tim mickelson wrote:> Hi. > > I'm not being able to make my Voicetronix Openswitch 12 work with > Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is > ringing, the Asterisk says that it is ringing, but the phone I'm ringing is > not ringing. I've seen in the mail list that other people have had the > problem that chan_vpb.c is making a call before hearing the dialtone. The > suggestioin was to put a comma or more before the number and this would make > a pause before actually dialing the number. This seemed to be a probable > cause of my problems, so I've defined in extesnsions.conf: > > [globals] > OUTDIAL=vpb/1-9/,,3487446196 > [default] > exten => _55.,1,Dial(${OUTDIAL},30,r) > > but this doesn't work, does someone have suggestions? > > Tim > > _________________________________________________________________ > Hitta rätt på nätet med MSN Sök http://search.msn.se/ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting
It is from this extensive thread that I fond that I should put a comma in the dial string, that didn't help, now what should I do? This thread regarding this issue does not help me. tim> Check the extensive thread regarding this EXACT ISSUE in the mailing > list archives. > > On Thu, 2004-03-18 at 04:36, tim mickelson wrote: > > Hi. > > > > I'm not being able to make my Voicetronix Openswitch 12 work with > > Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is > > ringing, the Asterisk says that it is ringing, but the phone I'm ringing is > > not ringing. I've seen in the mail list that other people have had the > > problem that chan_vpb.c is making a call before hearing the dialtone. The > > suggestioin was to put a comma or more before the number and this would make > > a pause before actually dialing the number. This seemed to be a probable > > cause of my problems, so I've defined in extesnsions.conf: > > > > [globals] > > OUTDIAL=vpb/1-9/,,3487446196 > > [default] > > exten => _55.,1,Dial(${OUTDIAL},30,r) > > > > but this doesn't work, does someone have suggestions? > > > > Tim > > > > _________________________________________________________________ > > Hitta r??tt p?? n??tet med MSN S??k http://search.msn.se/ > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > For Asterisk PBX related documentation go to > http://www.digium.com/index.php?menu=documentation and look at the > "Unofficial Links" section also see > http://www.voip-info.org/wiki-Asterisk also see my site at > http://www.fnords.org/~eric/asterisk/ > > BTEL Consulting > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Email.it, the professional e-mail, gratis per te: http://www.email.it/f Sponsor: Il notebook che hai sempre desiderato lo trovi su Ebest Clicca qui: http://adv.email.it/cgi-bin/foclick.cgi?mid=551&d=18-3