[Asterisk-Users] different pridialplan for different channels in zapata.conf
Hello, I read the previous postings in asterisk-users mailinglists and I didnt found any postings related to my problematic topic. Problem: If I need to set different pridialplan for different channels. For example: group1 has first 15 channels and all calls what are sendt via this group are "pridialplan=national" group2 has next 15 channels and all calls what are sent via this group are "pridialplan=international" Is there any way to present it so, as I described? Thanks for the attention. Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] different pridialplan for different channels in zapata.conf
Hello, I read the previous postings in asterisk-users mailinglists and I didnt found any postings related to my problematic topic. Problem: If I need to set different pridialplan for different channels. For example: group1 has first 15 channels and all calls what are sendt via this group are "pridialplan=national" group2 has next 15 channels and all calls what are sent via this group are "pridialplan=international" Is there any way to present it so, as I described? Thanks for the attention. Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP overlap (early dial) 484 response
Hello, I have one question again. I checked archive and I found that somebody before me asked this question already. But no responses for this posting. http://lists.digium.com/pipermail/asterisk-users/2003-September/020065.html So, is it supported or no? If yes, what I need to configure? Thank you. Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR (does not exist in any format)(No such file or directory)
Hello, I configured one simple IVR ### exten => 15001,1,Goto(ivrmenu,s,1) ;IVR [ivrmenu] exten => s,1,Ringing exten => s,2,DigitTimeout,30 exten => s,3,Background(welcome.wav) exten => 1,1,Dial(SIP/[EMAIL PROTECTED]) exten => 2,1,Dial(SIP/[EMAIL PROTECTED]) ### But "Background" does not work. I got error: *CLI> Feb 19 12:57:12 WARNING[245776]: file.c:446 ast_openstream: File than welcome.wav does not exist in any format Feb 19 12:57:12 WARNING[245776]: file.c:734 ast_streamfile: Unable to open welcome.wav (format ALAW): No such file or directory I tried "Background(/full/path/where/is/welcome.wav)" also... I dont understand where is the problem. Any hints? ________ Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100
Hello,
I have a problem with asterisk and Grandstream BudgeTone-100.
With default configuration everything works (in anonymous mode and fixed
IP), but if Im trying to enable registering, it dos not work.
I used 'sip debug' and verbose level 10, nothing happens if I switch
telephone on (no messages about bad auth etc). As I understood, after
switching phone on at first it will try to register in asterisk if Im
trying to call somewhere.
I searched in list-archive and I didnt found that anybody else has this
kind of problem. I read also:
http://lists.digium.com/pipermail/asterisk-users/2003-June/013288.html
and I did so.
sip.conf
-
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
disallow=all; Disallow all codecs
allow=g729
[cisco]
context=in
type=friend
insecure=yes
host=
dtmfmode=rfc2833
[grandstream1]
type=friend
secret=grandstream1
host=dynamic
context=class1
dtmfmode=rfc2833
[grandstream2]
type=friend
secret=grandstream2
nat=yes
host=dynamic
context=class1
dtmfmode=rfc2833
Asterisk ver: Asterisk CVS-01/22/04-18:13:23
Grandstream ver: Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18
* And as I mentioned before, without registration and with static IP
everything works, it seems, that something is misconfigured in my setup
for authentication or this phone firmware is buggy? (but its latest, I
checked www.grandstream.com)
Best Regards:
Key Aavoja
/* Never argue with an idiot. They drag you down to their level, then beat
you with experience.*/
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