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...a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061027/6dfa6909/attachment.htm

...xtension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel we're enable block dial, so alcatel only send to asterisk when user end dialing all digits. How i permit the first case to work ?? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061027/719b3a08/attachment.htm

...cron some kind of script or application that read asterisk logs and send via e-mail a complete report for pbx activity in specified period ?? I like to see how simultanios calls was made, total time in conversation, averege time of calls, most routes calls, etc.... Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061127/26a717f4/attachment.htm

...ut with our link provider to see if he has some firewall rules that can cause this problem I'm very very confuse becouse the invite message in every time come from my softswitch with ip of my softswitch so, why only invite originate on B side has this problem ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

Hi guys, My asterisk is show me some errors on line registration. This message appear on console: Request to schedule in the past?!?! What it mean ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

...ten=> _XXXXXXXX,2,Hangup exten=> _0XXXXXXXXXX,1,Dial(SIP/${EXTEN}@voipprovider,60,Tt) # Long distance Calls exten=> _0XXXXXXXXXX,2Hangup exten=> _00XXXXXXXXXX,1,Dial(SIP/${EXTEN}@voipprovider,60,Tt) # Internacional Calls exten=> _00XXXXXXXXXX,2Hangup Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061109/b39d4482/attachment-0001.htm

...fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes . . . [provider-25461099] type=friend context=default secret=xxxx username=25461099 host=sip.provider.com fromuser=25461099 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061127/86daf828/attachment.htm

Hi, I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1 and libpri-1.4.0 on a Debian machine with a TDM400P card. Everything goes ok but when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. My

...g. Tha calls stay without sound until the called part answer the call. At this point, conversation follow normaly. 2. When an alcatel extension use asterisk to make a call, after some time, around 2 minutes the calls is hangup. How i can resolve this two problems ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061127/44a8d517/attachment.htm

...d in 5060 port so when someone call to 4040.0001 the call arrive on asterisk but arrive to last number registered 4040.0004becouse it is listening on same port as all others. How i make each number register in one different port, like 5060,5061,5062,5063 and 5064 ?? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061109/8354d0dd/attachment-0001.htm

...com nat=yes insecure=very canreinvite=no qualify=yes If i disable 30 lines and restartr asterisk all lines are register normaly. So, Have any limit in network stack or in asterisk ? Have any tunning that can i make on linux or in asterisk to resolve this question ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

...account: register=>number:pass@sip.provider.com/number [fonar-number] type=friend context=default secret=pass username=number host=sip.provider.com fromuser=number fromdomain=sip.provider.com ;nat=yes ;insecure=very canreinvite=no ;qualify=10000 dtmfmode=rfc2833 Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

...ead send another register with authentication header, send another register message without authentication header. In most part of the time this asterisk work fine, except for this problems that happen 4 or 5 times per day. What could be the cause of this problem ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

...My asterisk was working fine but today my calls won't out of my asterisk box. Restarting asterisk i need to wait around 10 min to can run sip show registry command. If i try to run this command before, i receive a error like: no such command. Why this happen ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

...ted by our asterisk he can't transfer this call until called party answer that call. He can't transfer call when it's only ringing. This is a issue of Asterisk or from Alcatel. This PABX have 2 ISDN links. One with asterisk and other with other carrier. Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

...o me he can't hear ring back tone. My asterisk sent 180 ringing message to him. He told me that in 180 ringing there isn't a media attributes and i need to reconfigure my side to send 180 ringing with media attributes. How can i enable this on asterisk ? thanks. -- Frederico Madeira fmadeira at gmail.com www.madeira.eng.br

...if {${CALLERID(num)}=3000) { exten=> _X.,1,Dial(SIP/${EXTEN}@provider-302333-${CALLERID(num)},60,Tt) exten=> _X.,2,Hangup } else if {${CALLERID(num)}=3001) { exten=> _X.,1,Dial(SIP/${EXTEN}@provider-302222-${CALLERID(num)},60,Tt) exten=> _X.,2,Hangup } Thanks. -- Frederico Madeira fmadeira at gmail.com www.madeira.eng.br

...onse from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br

Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap'

I was getting permission errors (seen in /var/log/messages) in accessing these two directories within my chroot tree. I was pulling out what little hair I have, as the permissions were identical to those on my Centos 5.5 server. So I switched selinux into permissive mode and now I have /var/named/chroot/var/named/data/named.run and my ..../named/slave/ stubs. What is the selinux magic to