use the 1.6 branch and patch ??
I'll have to try this myself at some point.
Thanks,
Adrian
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrea
Cristofanini
Sent: 05 May 2010 14:22
To: asterisk
Hi list,
Anyone have successfully compiled amr codec for asterisk 1.6.1.X ?
I still have no problem compiling and playing with it on Asterisk 1.4.X.
I have used the following patch :
https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/
Hare is what i get while loading
tested those versions on the
same system? Can you reproduce those results?
What exact versions? On what system?
--
---
Andrea Cristofanini Chief Technical Officer
ZeroZero39 srlTel: +39 02 61294759
--
---
Andrea Cristofanini Chief Technical Officer
ZeroZero39 srlTel: +39 02 61294759
Viale Brianza, 20 Fax: +39 02 87365813
20092 Cinisello Balsamo (MI) Mob:+39 329 1871756
web
Hi There
http://www.2n.cz/company/2n_history.html
offer this kind of products.
they works vey well with asterisk.
Ciao Andrea
Michael Graves ha scritto:
On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote:
On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote:
The quad-band
You have to run DeadAGI, in h .
Regards
Andrea Cristofanini
voip crazy ha scritto:
Which is the way to run an AGI after hangup a call?
The problem I have is when the call dies the AGI dies too
I try the Dial command g option, but it does not work for me
Any clue will be welcomed.
Thanks
http://www.mgamble.ca/oss/iphone_asterisk/
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Hi
I just saw this now !
does the microphone and speaker works ?
Can you use it like softphone for recive calls ?
Regards Andrea
C F ha scritto:
TODAY I have managed to hack the iPhone and install Asterisk on it.
Detailed instructions to follow.
Cosini SS7 work for 99% of all cases.
Femi ha scritto:
Hi,
Has anyone on this list used the Cosini iAN7s SS7 gateway with Asterisk?
Please let me know how it performed and what issues you faced
Thanks
Femi
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yes i see
you have to enable in ADVANCED SETTING
Challenge Response on Phone: = OFF
Regards
/a
Matt Riddell ha scritto:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Has anyone ever seen an Snom320 lose settings?
It's been working fine for months and then I got a call this morning
Try zoiper
bilal ghayyad ha scritto:
Hi All;
I tried Firefly softphone with IAX and it gave very
poor quality.
Any one advise a good strong softphone that can work
with IAX fine?
Regards
Bilal
Hello
I'm running the same with no-problem.
CDR(userfield)=INCOMING
We use also a quite nice patch from Matt Riddell
http://bugs.digium.com/view.php?id=9424
that allow to have extra userfield.
CDR(userfield2)=${CODEC-IN}
CDR(userfield3)=${CODEC-OUT}
and so on...
This is quite good for custom
Andrea Cristofanini
CTO - VoIP
Gedam Europe S.r.l. - (Torino,Italy)
Gedam Advanced Communication Ltd - (Dunedin,New Zealand)
Strada da Bertolla all'abbadia di Stura, 151 - 10156 Torino - IT
GSM. +39-329.1871756 -
PSTN. +39-011.19824516 -
FAX. +39-011.8338622 -
http
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Cheers Andrea
Andrea Cristofanini
CTO - VoIP
Gedam Europe S.r.l. - (Torino,Italy)
Gedam Advanced Communication Ltd
-dstchannel)))
continue; /* people don't want to see unanswered single-channel
events */
Regards,
Atis
Andrea Cristofanini -- [GedamEurope] wrote:
Hi there !
So it is possible to have logged a single unanswered channels ?
What sould be stted into the cdr.conf ?
Regards Andrea
when i have CALLS WAITING in the queue, all the agent
are busy, after some times some Agent began available again and the
call is passed to an Agent, in this case callerid(name) began blank, so
look like that i have lost my variable.
Any idea ??
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Cheers Andrea
Andrea
cd /usr/local/bin
/bin/touch /usr/local/bin/$1.call
/bin/touch -r /usr/local/bin/$1.call -m -F 150 /usr/local/bin/$1.call
cp --preserve=timestamps /usr/local/bin/$1.call
/var/spool/asterisk/outgoing/
Doug
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Cheers Andrea
Andrea Cristofanini
Gedam Europe Srl
Gedam Advanced
Andrea
Andrea Cristofanini
Gedam Europe Srl
Gedam Advanced Communication Ltd
Torino, Italy
C.so Re Umberto 21
Mobile : + 39 329 1871756
PSTN: + 39 011 19824516
FreeVoip: 6838601
http://www.gedameurope.com
http://freevoip.gedameurope.com
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Hi list
i use SVN branch , i have real time working good with IAX2
The problem i have is for sip_buddies , any SIP acount register does
not store ip addres inside the table.
This only for SIP iax2 works great.
i also have in sip.conf
rtupdate=yes
any ideas ?
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Cheers Andrea
Andrea
://lists.digium.com/mailman/listinfo/asterisk-users
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Cheers Andrea
Andrea Cristofanini
Gedam Europe S.r.l.
Gedam Advanced Communication LTD
mobile : +39 3291871756
office : +39 011 5694900
freevoip : 6838602
MSN : [EMAIL PROTECTED]
http://www.gedameurope.com
http://www.asterisknews.it
http
Hi list
i have a Asterisk box that use 10 phone with sccp, and some iax2
Every 8 10 hours , my linux machine change ip address and route, and the
cisco and iax phone cannot see the server ...
What can do that?
there are no other linux box , no any pc that provide DHCP
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Cheers Andrea
Andrea
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Cheers Andrea
Andrea Cristofanini
Gedam Europe S.r.l.
Gedam Advanced Communication LTD
mobile : +39 3291871756
office : +39 011 5694900
MSN : [EMAIL
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Cheers Andrea
Andrea Cristofanini
CEO Gedam Europe S.r.l.
CEO Gedam Advanced Communication LTD
mobile : +39 3291871756
office : +39 011 5694900
www.gedameurope.com
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Andrea Cristofanini
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CEO Gedam Advanced
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