the developers.
Thanks much.
Darren Ellis
Apr 11 15:45:13 vongate PALSDIAL[22783]: Post-call CCC at 2
Apr 11 15:45:27 vongate PALSDIAL[22785]: Call to
IAX2/[EMAIL PROTECTED]/1aaannn4201 queued successfully.
Apr 11 15:45:27 vongate PALSDIAL[22785]: My channel to
IAX2/[EMAIL PROTECTED]/1aaannn4201 is IAX2
Hi All,
Could someone send me a code frag on how to get a record from the
asterisk database into a PHP variable via the Manager API?
I can issue calls, etc. from Manager. But I'm not comprehending how to
manipulate database variables.
Thanks much.
Darren Ellis
Hi,
I have an asterisk 1.2.6 server working well as a voicemail/VOIP
integration for a Merlin Magix. Special thanks to Dan Polk for his
help. Everything is working as expected except that one person in the
organization has the surname Le The only way to navigate to him is by
typing 53#.
()
As can be seen, I've tried calling voicemailmain with the ${EXTEN:4:3}
digit stripping as part of the command, and also I've tried moving the
digit stripping to a variable.
I'd very much appreciate any help you folks can offer.
Thanks much.
Darren Ellis
.
They would like to have the SPA-941 accept internal calls while DND is
set. If any of you know how to make this happen, I'd very much
appreciate your help.
The paging feature is not what they want, and the SPA-041 ignores the
answer-after=0 SIP header when DND is on anyway.
Thanks much
Darren Ellis
Olle E Johansson wrote:
This morning we discovered a serious bug that stopped all bridged
audio in our Asterisk servers. Mark found the problem and soon fixed it.
If you get this problem today, please update your Asterisk server. A
fix has been commited to the subversion repository for 1.2
Guys,
I'm not familiar enough with mantis to tell what version of asterisk are
affected by this bug?
I have 1.09, 1.10, 1.2.1 and 1.2.2 (as a test) deployed.
Can someone tell me what the real impact is going to be?
Thanks
Darren
___
--Bandwidth
Zoa wrote:
All versions released in the last 2 weeks i think.
take the newest versions from svn or the ftp. (1.2.3 is released).
Thanks Zoa,
/emergency mode off
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
Hello,
I just bought a Snom-320 from ATAComm. I plugged it into my LAN,
registered it with *, etc. All my other SIP gear is Sipura and works
fine, both on the LAN and over the Internet.
The new Snom seems like it can't process the audio from the handset
mic. A steady tone is garbled, even
Hi,
If there's anyone out there who has successfully compiled * 1.0.7 on
10.4.1, could you contact me off-list? I've tried the astmasters
mailing list, but it continually rejects my messages.
Thanks much.
Darren Ellis
[EMAIL PROTECTED
Rich Adamson wrote:
Is there any reason to avoid * on Fedora Core 3 at this time?
Have most/all of the issues been resolved now?
Rich,
Both my Asterisk servers run FC3. The only issue I ran into was the
change in RPMs for the source. FC doesn't distribute the
kernel-source RPM any more.
Hi,
I have two asterisk machines, chomper and otao.
otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no
PSTN connections.
chomper is at my house, connected to cable modem behind NAT, but has a single
X100P PSTN connection.
I would like to establish two way calling between
Hi,
I have two asterisk machines, chomper and otao.
otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no
PSTN connections.
chomper is at my house, behind NAT, but has a single X100P PSTN connection.
I would like to establish two way calling between otao and chomper.
Right
Hi,
I have * working with X-Lite and Sipura adapters, but I have one person
who is linux based, and is trying to use Linphone and Kphone. His end
works, but I get very bad echo on my end. Have any of you folks been
able to get linux based soft phones working well with *?
I'd appreciate links
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