Re: [Asterisk-Users] Voicemail and musiconhold sound stopped working
On 7/26/05, Arnd Vehling [EMAIL PROTECTED] wrote: Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. [snip] I am using an installation with several different sip-fones, zaptel+zaprtc as well as fcpci+capi on a teles isdn card. Any ideas where to look for? This could be way off but make sure that when you do a make install for zaptel that you get no errors. This happenned to me when I was still installing wcusb.o when I had disabled usb in the kernel. Cheers, RAB ___ 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
Re: [Asterisk-Users] Asterisk not startin anymore.
On Sun, 28 Nov 2004 14:10:29 -0300, Andres Junge [EMAIL PROTECTED] wrote: Hello. I have this problem. In my asterisk box, I was running debian woody with asterisk package from backports.org. Last friday I upgraded from debian to sarge and change from kernel 2.4.18-1-686 to kernel 2.6.8-1-686, rebuild zaptel kernel module and also upgrade to asterisk 1.0.2. But now asterisk won't start. Here is more info [snip] This has happenned to me now too - so I doubt that your hardware is faulty... Regards, RAB ___ 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
Re: [Asterisk-Users] Asterisk not startin anymore.
On Wed, 1 Dec 2004 20:13:16 +1000, Robert Barnes [EMAIL PROTECTED] wrote: This has happenned to me now too - so I doubt that your hardware is faulty... Oops - wcfxs was renamed to wctdm some time ago... Working again now. RAB ___ 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
Re: [Asterisk-Users] Inbound IAX2 calls has no music on hold
On Tue, 17 Aug 2004 10:19:17 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hmm, My music on hold has always worked fine. But I discovered that under incoming IAX2 calls they don't get any MOH! All I could find was a comment saying let me know if you find a solution... Nor does the debugger does say: Started music on hold So it's not starting the MOH, why? I do have it configured and it does play under other types of calls. - -- Steve Steve and All, I am sure that my voicemail was once working too. However now, depending on the incoming codec (I am talking IAX2) the message recording gets cut off after a few seconds. The codecs that don't seem to work are iLBC and G729. Sorry to chime in with a variation on the subject, but maybe the cause is the same. Regards, Rob ___ 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
Re: [Asterisk-Users] Asterisk Recommended Distro
On Wed, 11 Aug 2004 16:32:03 -0700, Bryan Vyhmeister [EMAIL PROTECTED] wrote: I am trying to run asterisk on Trustix 2.1 w/updates. Has anyone had any troubles related to zaptel hardware like sporadic errors when trying to dial out? Also, what is the consensus for the best Linux distribution to run asterisk on? Trustix seems like a nice choice but I would like to hear the recommendations. Thank you for your help. The best Linux distro is the one you are most comfortable with. Don't forget to check the Wiki: http://www.voip-info.org/wiki-Asterisk+OS+Platforms Regards, Rob Barnes ___ 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
Re: [Asterisk-Users] Firefly and *... Argh!
On Tue, 10 Aug 2004 20:18:29 +1000, David MacKinnon [EMAIL PROTECTED] wrote:
Okay, I've read as much as I can, and I think i've followed
instructions, but I'm still having problems with * and firefly... I can
get outgoing to other freshtel working, but not incoming (I get the not
available voicemail), or outgoing to landline.
I'm using the debian asterisk package (0.9.1-RC1-4)
I think the general consensus is that it is best to track the cvs
yourself, that Debian package is probably quite old.
My iax.conf has in general (under my FWD register, which works...)
register = (MYNUMBER):(MYPASSWORD)@cts-au.freshtel.net
Try this:
register = (MYNUMBER):(MYPASSWORD)@gateway.freshtel.net
[firefly]
type=friend
host=cts-au.freshtel.net
context=from-firefly
I have this:
;
; Firefly (Freshtel)
[my_number] ; Firefly
context=firefly-number
qualify=no
username=my_number
secret=my_secret
auth=md5
type=friend
host=gateway.freshtel.net
I never see a registered message when starting up * (I do for FWD)
When sniffing the interface I can see the following IAX2 data on startup.
REGREQ (From me - freshtel)
REGAUTH (Freshtel - Me)
REGREQ (Me - Freshtel)
REGACK (Freshtel - Me)
ACK (Me - Freshtel)
My extensions.conf looks like
exten = _394.,1,SetCallderId(MYNUMBER)
exten =
_394.,2,Dial(IAX2/MYNUMBER:[EMAIL PROTECTED]/${EXTEN:3},60,r)
Again, similar setup is working with FWD.
My extensions.conf looks like:
; RAB - Firefly (Freshtel)
[outgoing-firefly-peers]
exten = _62,1,Macro(outgoingfirefly,${EXTEN:2},70) ; Firefly
; RAB
[macro-outgoingfirefly]
exten = s,1,SetCallerID(my_number my_number)
exten = s,2,Dial(IAX2/my_number:[EMAIL PROTECTED]/${ARG1},${ARG2},r)
exten = s,3,Congestion
; RAB
[macro-outgoingfreshtel]
exten = s,1,SetCallerID(my_number my_number)
exten = s,2,Dial(IAX2/my_number:[EMAIL PROTECTED]/${ARG1},${ARG2},r)
exten = s,3,Congestion
[snip]
So I assume my auth is being refused. What have I missed here? Does
anyone currently have firefly working like this with * and would care to
share their setups? :) I've just started playing with this stuff recently.
I am not sure what you have missed, but I do have Freshtel working to
my satisfaction. I hope my snippets help..
Thanks,
-David
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Re: [Asterisk-Users] Firefly and *... Argh!
On Tue, 10 Aug 2004 21:07:53 +1000, David MacKinnon [EMAIL PROTECTED] wrote:
gateway.freshtel.net gives Number is disconnected messages for all the
PSTN numbers I try, cts-au.freshtel.net gives me the no such
context/extension errors. Are you using firefly for PSTN calls?
Yes, I am using Freshtel for local, std, mobile and international calls.
Are you putting the 61 in front of the outgoing number? Freshtel
always needs the full number including the country code. Can make PSTN
calls using Firefly OK?
Here is a snippet from my extensions.conf where I take an STD number
and send it out using Freshtel:
; RAB
[outgoing-std]
exten = _0[238],1,Macro(outgoingfreshtel,61${EXTEN:1},70) ; Freshtel
Thanks for the help :)
No problem,
RAB
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[Asterisk-Users] Called ID in Australia
Hello All, Can any Australians who have any info or current patches relating to Caller ID in Australia please drop me a line? There is little or no info on the Wiki regarding this topic, although I am aware of a related patch mentioned in the bug tracker. Regards, Rob Barnes ___ 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
