Displaying 20 results from an estimated 20 matches for "comella".
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comellas
2005 Aug 27
3
Low handset microphone volume with Sipura SPA-841
...far. I've tried tweaking several configuration options
but nothing I has helped so far.
Has anybody else experienced this problem? There are only two holes for the
microphone in the handset and they are really small. I was thinking that
myabe this is the cause. Any thoughts?
--
Juan Jose Comellas
(juanjo@comellas.com.ar)
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323connections
.....o-------------------------------------------------------o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose
Comellas
Sent: Friday, September 30, 2005 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] No ringback tone generated by Asterisk with
OH323connections
I am using Asterisk (Debian unstable packages) with an OH323 connection
to my
provider. Everything is worki...
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323 connections
...n=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
bandwidthLimit=2000
gatekeeper=DISABLE
gatekeeperTTL=600
userInputMode=RFC2833
The package versions I'm using are:
asterisk 1.0.9.dfsg-5
asterisk-oh323 0.6.6pre3-4
libopenh323-1.15.3c2 1.15.3-4
--
Juan Jose Comellas
(juanjo@comellas.com.ar)
2004 Sep 01
1
Dynamic dialplan
...to active calls after the dialplan configuration is updated?
- Can we do partial updates of the dialplan (e.g. update a specific context
instead of the whole dialplan configuration)?
- Can Asterisk have its dialplan in a database instead of having it always in
memory?
Thanks.
--
Juan Jose Comellas
(juanjo@comellas.com.ar)
2005 Jul 11
1
Zaptel configuration for Argentina
...7)
signalling=fxs_ks
language=en
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
callgroup=1
pickupgroup=1
group=1
useincomingcalleridonzaptransfer=yes
callerid=asreceived
context=pstn-inbound-voice
channel => 2-7
Thanks
--
Juan Jose Comellas
(juanjo@comellas.com.ar)
2005 Jul 27
19
Full T38 sip Faxing now Available
Hello everybody, for all of you that have searched for a real fax
solution, look no further. We now have T38 faxing. Please contact me for
more information.
Thanks
Michael D. Schelin
ShellTel
626-814-2354
2005 Oct 04
3
Asterisk as H323 gateway
Is there anyone who is currently using Asterisk as a production H323
gateway?
And using which combination of asterisk and H323 (chan_h323, chan_oh323?)
The main issue is interoperability with other H323 parties (Cisco AS53xx,
Nextone, etc).
Searching the mailing list it seems that both h323 and oh323 are not so
stable, is it only an impression or using h323 is really not so advisable?
2005 Feb 07
1
Conferencing without Meetme
...apparently
they only allow two channels. Is there any special precaution that I have to
be aware of when doing this? Do I have to masquerade the channels that are
inserted into the conference? The channels will mainly use SIP (maybe IAX2
too occasionally).
Thanks for your help.
--
Juan Jose Comellas
(juanjo@comellas.com.ar)
2005 Jul 02
3
LDAP search application for Asterisk
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2005 Aug 26
1
Is LDAPget module stable enough for enterprise usage?
Hi, all. I am building a SER+asterisk PBX airming at around 10k
persons' usage. For authentication purpose I am in favor of ldap
storage, while I am not sure the current ldap module for
asterisk(0.9.9.2) is stable enough? sorry I do not master the proper
testing mechanisms to find out myself.
Thanks in advance.
2007 Nov 04
4
Can''t run samples - Mac OS 10.3 w/ RubyGems
Hello -
I''m interested in leveraging wxRuby on my PowerPC-based iBook.
I''ve got Mac OS 10.3.9 and I installed wxRuby 1.9.2 using gems.
However, when I try to run samples such as minimal.rb, I get the
following error:
dyld: ruby Undefined symbols:
_DataBrowserChangeAttributes
_PMGetDuplex
_PMSetDuplex
_TXNGetCommandEventSupport
_TXNSetCommandEventSupport
2005 May 16
0
spandsp in 64 bit Linux on AMD64
...ffix or operands invalid for `popf'
/tmp/ccXxGHg6.s:15: Error: suffix or operands invalid for `pushf'
/tmp/ccXxGHg6.s:16: Error: suffix or operands invalid for `pop'
/tmp/ccXxGHg6.s:17: Error: suffix or operands invalid for `popf'
make[2]: *** [testcpuid.lo] Error 1
--
Juan Jose Comellas
(juanjo@comellas.com.ar)
2005 Jun 27
1
Native MoH patch for 1.0.8?
Hi all,
I was reading http://bugs.digium.com/view.php?id=2639 and it seems that
anthm's great native MoH patch only works on HEAD. Does anyone have a
version of the native MoH patch that works on 1.0.8? If so please point
me to its location or email it off-list.
Thanks and regards,
Patrick
2005 Sep 20
3
sipuras 841 bad sound
Hi Guys!
I have a problems with some sipuras 841 and asterisk 1.0.9.
Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with
steve's unicall.
Everything compiled fine and in fact I can make and receive calls but I have
a problem with bad sound when the sipuras call the outside E1's lines. I can
listen to the caller without problems but they heard me with a choppy
2007 Jan 07
5
Some queries on g729 license.
Hi, all
I am a pabx vendor from Singapore. Recently we are going to implement a
failover solution for our customers using heartbeat, the asterisk server
can failover perfectly, however the g729 codec canot work, because it is
binded the mac address, we have bought two set of licenses, can you
provide us some workaround for this scenario?
Regards,
Liangliang
2005 Feb 16
0
broadcast and switch mode problem
...tinc v1.0.3 does have a bug that prevents broadcast
packets to be sent when the switch mode is used. I actually found that while
trying to set an vpn using the Debian unstable packages of tinc (which are
v1.0.3). Are any plans to fix this in an official release soon?
Regards
--
Eduardo D?az Comellas
ediaz@ultreia.es
2005 Sep 16
4
Caller Name: Asterisk reading too fast
I asked my telco to release caller name on the PRI. Earlier they were
releasing only the phone number.
I still did not see the name, but only the number in caller id. Actually
I now see number twice. When I inquired with them this is the response I
got:
"I ran a trace on your TG. I see that your switch is
picking up the call so fast that it is not able to pick
up the name. The
2005 Jun 30
0
Re: Asterisk-Users Digest, Vol 11, Issue 181
...;
>
> Today's Topics:
>
> 1. Asterisk ended with exit status 1 (Federico
> Alves)
> 2. Re: Re: teliax [Was: LiveVoip is Bankrupt]
> (Rich Adamson)
> 3. RE: Polycom & VPN trouble (gw@adcomcorp.com)
> 4. Re: Native MoH patch for 1.0.8? (Juan Jose
> Comellas)
> 5. Re: Polycom & VPN trouble (Tim Pushor)
> 6. Re: Re: teliax [Was: LiveVoip is Bankrupt]
> (r00t)
> 7. Newbie Confusion on Call Forward and
> DBput/DBdel (Jeffrey Starin)
> 8. Eicon equipment, BRI Server or PRI?
> (gw@adcomcorp.com)
> 9. Re: Level 3 S...
2006 Mar 25
2
Asterisk spanDSP / Faxing problem
Hi There.
I have the following setup :
Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24
My problem is as follows :
If I set up a very simple extensions.conf. when I dial from a fax
machine, it seems as if no fax is being recognised.
If I answer the call, I can hear the fax machine beeping.
extensions.conf :
2008 Feb 29
0
Skewed RTP timestamps in SIP calls on Asterisk 1.4.18
Last week I migrated some of our servers to Asterisk 1.4.18 and we started
seeing audio drops of several seconds during SIP calls. After investigating
it we noticed that Asterisk was increasing the RTP timestamps abnormally
during a conversation.
I'm including a text file with a subset of the data collected by Wireshark
that shows the problem (I have the complete packet capture if anybody