[Asterisk-Users] Pattern matching in extensions.conf

be combined into one extension. Best regards, Mickey Binder ___ 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

RE: [Asterisk-Users] Pattern matching in extensions.conf

What is 00 and other numbers? Are different destinations prefix ?? Nope, it's just the last 2 digits of some 8 digit numbers that isn't supposed to be reachable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mickey Binder Sent: vendredi 18 mars 2005

[Asterisk-Users] ZAP channel on TE410P doesn't hang up

e know which configs or logs to provide, any help is greatly appreciated. Kind regards, Mickey Binder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update op

[Asterisk-Users] ZAP channel on TE410P doesn't hang up (Plain Text this time)

a workaround. Please let me know which configs or logs to provide, any help is greatly appreciated. Kind regards, Mickey Binder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Differences in CDR files

these differences, could it be a hanging zap channel or something like that? Thank you, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Stuck SIP channels? - SIP show channels

/2 UNKN (d) 36 active SIP channel(s) Is this something that I should worry about? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

RE: [Asterisk-Users] Needed Open Ports

+ports+tcp+udp+as teriskbtnG=S%C3%B8gmeta= Or this search engine: http://asterisk.linkx.net/cgi-bin/asterisk regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

RE: [Asterisk-Users] Use buttons (other than #) after call is bridged?

That should be pretty easy if you already have the interface to the DPIN I/O port. Then it would be just a matter of some system() calls on different extensions which communicate with your DPIN I/O port program. e.x.: exten = 99910,1,system(io_prog 1 0) ;turn port 1 off exten =

[Asterisk-Users] SIP flashhook transfer

pickupgroup=1 restrictcid=yes [43300645] type=friend username=43300645 secret= pickupgroup=1 callgroup=1 dtmfmode=inband host=dynamic defaultip=10.1.1.6 callerid=45 Regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

RE: [Asterisk-Users] Zaptel/PRI problem

the compiler tells my I need a newer zaptel. I therefore downgraded my Asterisk to same date, but are there any other options? Regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface

But nevertheless my mobile is still showing the number I'm dialing from. Our provider is song networks which is a Danish Telco provider. If anymore debug info is needed let me know Hello again Just wanted to say that on another location with exact same setup but another telco provider,

Re: [Asterisk-Users] Hide outgoing CallerId on Zap interface

is a Danish Telco provider. If anymore debug info is needed let me know Regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface

via my PABX. -Original Message- From: Mickey Binder [mailto:[EMAIL PROTECTED] Sent: 13. februar 2004 12:14 To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some

[Asterisk-Users] Hide outgoing CallerId on Zap interface

number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

[Asterisk-Users] Hide outgoing CallerID

SIP device no. 2 then I want to show my number I know it is possible to just show my main number instead of changing my ANI but I want to completely hide the number, is that possible? Mvh Mickey Binder Comflex A/S Roskildevej 342D 2630 Tåstrup Tlf.: 43 99 71 02 Direkte: 43 30 06 34

[Asterisk-Users] Flash hook - SIP device

to tell him in order to get him to fix this. What is happening when you flash hook, I mean how does Asterisk see and handle this? What should the SIP device send to Asterisk so it works properly? regards Mickey Binder [EMAIL PROTECTED])fjåŠËbú?jË^®+$ºÇ«

RE: [Asterisk-Users] Distortion of voice after cvs upgrade

case its only the MOH thats distorted, it sounds like there is a "autumn storm" in the background. The wierd part is thatthe noisestarts after appr. 5-10 seconds,until then the music is clear. My connection to the outside is an E1 on a TE410P. regards Mickey Binder

[Asterisk-Users] Distinguish between voice and data call

voice or data. Is it somehow possible to end/ignore this call already before it is ringing? regards Mickey Binder [EMAIL PROTECTED])fjåŠËbú?jË^®+$ºÇ«

RE: [Asterisk-Users] Gastman crashes on Win32

If you mean how to get the CVS version you just have to do a checkout from digium. export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login password: anoncvs cvs co gastman regards Mickey Binder -Original Message- From: rnc Info Lists [mailto:[EMAIL PROTECTED] Sent: 23

[Asterisk-Users] Different MGCP issues

not sending the correct DTMF, (I live in Denmark where we use DTMF CLIP. Are there anything that I need to setup in mgcp.conf in order to enable incoming callerid? regards Mickey Binder [EMAIL PROTECTED])fjåŠËbú?jË^®+$ºÇ«

[Asterisk-Users] Call pickup - Change shortcode

Hello Is it possible, (without hacking the source), to change the code for call pickup because my SIP gateways uses * key as End-Of-Dial. If I have to hack the source can somebody tell me where to look? Mvh Mickey Binder Comflex A/S Tlf.: 43997102 Ë^®+$RÇ«²f¢–)à–+-Ë^®+$RÇ«²X¬¶Çb‚+¦r‰¡¶ÚþX

RE: [Asterisk-Users] Outgoing CallerID

Change pridialplan to unknown in zapata.conf Martin Called Number (len=11) [ Ext: 1 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2840' ] Cant figure out what's wrong? regards Mickey Binder ÿÿÿÀ

[Asterisk-Users] Problems with TE410P and E1 line -- Unable to open D-channel 24 (No such device or address)

CVS versions of asterisk. regards Mickey Binder ,µêâ²E,z»j)bž b²Ð,µêâ²E,z»%ŠËlv(ºg(šm§ÿåŠËlv(ºg(›ùšŠYšŸùb²Ø§~Ú²×«ŠÉ.±êì

RE: [Asterisk-Users] Problems with TE410P and E1 line -- Unable to open D-channel 24 (No such device or address)

You have the card jumpered as a T1 card, not an E1 card. Look in the middle of the card for the jumpers. -- Alastair Maw Ahhh...sht. Completely forgot about those jumpers. DOH! Thank you for the reminder :O) kind regards Mickey Binder ^+$Rf)+-^+$RXb+rXb+r+-w-z

RE: [Asterisk-Users] Outgoing CallerID

(4330) exten = _XX.,2,Dial,Zap/g1/BYEXTENSION The main isdn number is 4399 and is the only number I can get displayed when calling. regards Mickey Binder ,µêâ²E,z»j)bž b²Ð,µêâ²E,z»%ŠËlv(ºg(šm§ÿåŠËlv(ºg(›ùšŠYšŸùb²Ø§~Ú²×«ŠÉ.±êì

RE: [Asterisk-Users] Outgoing CallerID

(PRI) exten = _XX.,1,SetCallerId(4330) exten = _XX.,2,Dial,Zap/g1/BYEXTENSION The main isdn number is 4399 and is the only number I can get displayed when calling. regards Mickey Binder I've done some more testing and by debugging the pri span I think I've found where it tries

[Asterisk-Users] DTMF CLIP

Hi all Just curious to hear if anything has happenend in the DTMF CLIP matters: http://bugs.digium.com/bug_view_page.php?bug_id=009 I would be very happy to see it implemented regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL

RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer

, and I don't have other clients to test with. -Josh Ok I get same results when using Answer, so I'll just stick with that thx Mickey - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:13 AM

[Asterisk-Users] SIP - DTMF Payload type

on the net) and verified that voice responds now worked, but I don't know if this is the correct type? Still I can't use * or # regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP - DTMF Payload type

Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload type was set to 96 (default) on my Wellgate, Asterisk responded with NOTICE[606227]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 96 received Just wanted to

[Asterisk-Users] I don't think I understand Call pickup

have experimented with those parameters in sip.conf but are not aware of exactly where to use them. Can those be put under the [general] section or should they go under each user definition? regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL

RE: [Asterisk-Users] I don't think I understand Call pickup

and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up the call that comes to channel 1. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1

RE: [Asterisk-Users] I don't think I understand Call pickup

in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able to do a pickup from #2 as well? And how would my configuration look, do I have to specify anything

RE: [Asterisk-Users] I don't think I understand Call pickup

: [Asterisk-Users] I don't think I understand Call pickup You have to do it reverse way ... pickupgroup = 1 for sip phone (since you're picking it up on this one) and callgroup = 1 for zap channels. Martin On Thu, 4 Sep 2003, Mickey Binder wrote: Just have that zap channel in the pickupgroup

RE: [Asterisk-Users] I don't think I understand Call pickup

think I understand Call pickup Just have that zap channel in the pickupgroup = callgroup of the sip phones Martin On Thu, 4 Sep 2003, Mickey Binder wrote: What if I have two sip phones and a call arrives for #1 from my zap interface, should I be able

[Asterisk-Users] MusicOnHold and MP3Player not triggering answer

, but it would be nice to find the source of the problem. regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Change include contexts runtime

That sounds like a brilliant idea, I will try it right away! -Original Message- From: Tilghman Lesher [mailto:[EMAIL PROTECTED] Sent: 2. september 2003 05:05 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Change include contexts runtime On Monday 01 September 2003 03:51, Mickey

RE: [Asterisk-Users] Change include contexts runtime

Mickey Binder wrote: That sounds like a brilliant idea, I will try it right away! Did it work out all right? /t It looks like it. With DBput and DBget im able to change the variable values and then branch to different contexts with GotoIf. Now I just need to implement the right logic

RE: [Asterisk-Users] Change include contexts runtime

-Original Message- From: Tomas Prybil [mailto:[EMAIL PROTECTED] Sent: 2. september 2003 10:50 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Change include contexts runtime Mickey Binder wrote: It looks like it. With DBput and DBget im able to change the variable values

RE: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything

00:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything Hi Mickey, On Sat, Jul 05, 2003 at 18:23:50 +0200, Mickey Binder wrote: Hello there Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After

[Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything

get the undefined symbol error again. -- Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything

Ok, thx -Original Message- From: Peter Zeltins [mailto:[EMAIL PROTECTED] Sent: 5. juli 2003 19:11 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything Yesterday I updated my pwlib, openh323 and Asterisk from

[Asterisk-Users] Asteriks, GnuGk and outgoing calls

directly to Asterisk i can easily place outgoing calls, so the setup for outbound calls works I think. -- Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users