is
connected it reenables GSM on Xlite and uses that instead.
The obvious answer is to reenable gsm in the general section, but this
causes a problem with unknown inbound SIP calls, which I want force to G729.
Is this a bug, by design or is it a problem specific to Xlite?
Any help appreciated.
--
Jamie
before the
signed code.
Nobody I've asked has mentioned any bugs that are not present in the pre
5 code, but then again there don't seem to be any advantages either.
I would be interested to hear if there were some useful new features in 5.x.
--
Jamie Neil | [EMAIL PROTECTED] | 0870 454
:).
Just in case anyone is interested, AVM also say that you can't use the
Fritz card and the B1 card in the same box. However I have found it
seems to work fine provided the B1 CAPI driver is loaded *after* the
Fritz driver.
--
Jamie Neil | [EMAIL PROTECTED] | 0870 454
Versado I.T. Services Ltd
at the same time, which in my
experience produces pretty good results.
I posted the process that I use to the list a few weeks ago:
http://lists.digium.com/pipermail/asterisk-users/2003-July/016399.html
Regards,
--
Jamie Neil [EMAIL PROTECTED]
Versado I.T. Services Ltd.
http://versado.net
Quoting Nathan Littlepage:
Has anyone had any major issues with the Cisco 7940 and or 7960 phones?
Not yet - mind you I only got my 7940 working half an hour ago ;)
I'm running SIP v4.4 and everything seems to work fine: hold, transfer,
message waiting etc. So far I'm very impressed.
Out of
from CVS a couple of times.
Can anyone confirm this?
Jamie Neil
Versado I.T. Services Ltd.
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MOH working
with 1050 before though...
Haven't noticed any new bugs yet, but they have fixed the sound card
selection problem which makes it much more usable.
Jamie
Rgds,
Stuart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jamie Neil
Sent: 18
Quoting [EMAIL PROTECTED]:
While you gugs are on the subject of MOH and SIP, exactly where in my
configs do I turn on MOH for my SIP clients?
Also while we're on the XLite would anyone like to help in getting my
XLite client to work. I worked with several people on the irc
channel the
other
Quoting Oliver Brandt:
Hi,
I'm planning to buy a new ISDN-PBX (I hope this is the right term for an
ISDN phone system). I would also like to connect it to asterisk. As far
as I know there is no ISDN card where I can connect an ISDN-Phone to
directly working together with asterisk (please
Quoting WipeOut:
Hi,
I have just been playing with the latest X-Lite.. It works fine
with Asterisk..
As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only
one that didn't work.. not sure why..
Did you get Speex working? I've tried, but although I can get it to connect,
there is
Quoting Paul Cheng:
Hi,
We have our * box configured to receive inbound SIP calls from FWD and
enter into an autoattendant where someone can enter an extension
directly.
However, the inbound DTMF is not being correctly detected in most
cases. Entering 8050 results in a correct detection,
Quoting Michael Manousos:
Michael Ulitskiy wrote:
Michael,
With all due respect to both of you, it's not related to h.323 driver.
The result is the same whether h.323 channel participates in the
call or it's pure sip-to-sip call.
Did you try it without the ztdummy and zaprtc?
I
' oh323 channel
driver was _not_ affected by this problem (this was before Jeremy McNamara's
h323 channel was usable), maybe because it doesn't use the same * libraries
as other voip channels, however it's been a while since I played with h323
so I don't know if that's still the case.
Jamie Neil
Versado
RFC 2833 it doesn't seem to pick up 0 properly.
For example if I dial the voicemail app and then enter my extension (600) it
says Login incorrect even though I haven't been prompted to enter my
password, and the console shows the dialed number as 6 or 60.
Has anyone else got this to work?
Jamie
extension (600) it
says Login incorrect even though I haven't been prompted to enter my
password, and the console shows the dialed number as 6 or 60.
Has anyone else got this to work?
Jamie Neil
Versado I.T. Services Ltd.
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then the RFC2833 signalling is ignored anyway.
Jamie
Thanks
Lee Goodman
- Original Message -
From: Jamie Neil [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 31, 2003 7:31 AM
Subject: [Asterisk-Users] RFC2833 problems with X-Lite
Hi,
I've managed to get X-Lite (v2 build 1050
Quoting [EMAIL PROTECTED]
What have people found to be the ideal setup for recording asterisk
prompts?
I'm looking for both the ideal application to record them in, the ideal
format, as well as hardware (do I need a fancy studio mic or will a
headset mic work?).
I had pretty good results
this single (easily
repeatable) error.
Jamie Neil
Versado I.T. Services Ltd.
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of getting it working again is to restart *.
When I switch to softdtmf, everything seems to work fine, but I noticed that
even though DTMF signalling works fine on the IVR menu, once the call is
bridged DTMF digits entered on the PSTN phone are not displayed on the
console like before.
Jamie Neil
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