Displaying 20 results from an estimated 9000 matches similar to: "ArrayVox problems"
2003 May 20
0
FW: SIP-firewall problem?
Thanks for the firewall hint. Actually, most of my UDP and TCP port are open.
I believe the problem is related to NAT. Here is the reasoning:
- I call from SJPhone using a real IP address (e.g. 200.22.33.44)
- The call reaches my router (with a real IP address of 200.33.44.55), which
routes the packets to my * server at the fake IP 192.168.0.200
- My * server places a call through my TDM
2003 May 07
1
SIP-firewall problem?
Hi,
I have asterisk working well on my intranet, using SJPhone to make SIP calls.
Everything works fine except when I try to connect from outside my intranet.
I've opened the 5060 UDP port on my D-link firewall/router and, although I can
dial, complete calls and hear the other party, my voice gets lost (the other
party does not hear me).
I've took a look on the >sip debug output
2003 Oct 17
2
Beta testers for visual configuration tool f or asterisk
Count me in too.
-----Original Message-----
From: sip [mailto:sip@intology.com]
Sent: Friday, October 17, 2003 1:56 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for
asterisk
count me in
----- Original Message -----
From: "Paulo Mannheimer" <paulohm@instant.com.br>
To: <asterisk-users@lists.digium.com>
2003 Dec 11
0
FW: Iax, Iax2 and Iaxcomm
Talking to myself ... ;-)
Solved this by ...
disallow=all
allow=gsm
;allow=ulaw
;allow=alaw
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paulo
Mannheimer
Sent: quinta-feira, 11 de dezembro de 2003 09:02
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Iax, Iax2 and Iaxcomm
Hi,
I'm trying
2003 Dec 09
2
Erratic DTMF on E1/PRI (continuation of Strage bip on ISDN/PRI)
At the same site, DTMF recognition is functioning badly, sometimes
duplicating digits and sometimes totally missing others.
We have checked already /proc/interrups, there is no interrupt being
shared.
Our zaptel has .. span=1,1,0,ccs,hdb3
On zapata we have ...
switchtype=euroisdn
signalling=pri_cpe
relaxdtmf=no (yes doesn't seem to help)
-----Original Message-----
From:
2003 Sep 04
1
Arraycom voip phone
Hi All,
Does anyone have any experience with the ArrayCom VoIP phone?
I bought one a couple of weeks ago, it used to work quite well with *
until I misconfigured one option.
I now cannot make it work anymore, because the phone boots up, doesn't
find a valid SIP gateway, resets itself and keeps rebooting indefinetely
;-( Their technical support refuses to answer my questions.
Any hint on a
2003 Jun 23
5
dynamic queue channels
Hi, I'm trying to build a call center application that allows attendants
to come in the morning and dial a certain extension to make their
extension available.
I wouldn't like to use the AgentLogin app because their line would need
to stay off-hook (is this correct?)
Is there any SET channel status command that would allow me to do
something like this?
PauloHM
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2003 Jun 05
1
answering calls with SIP phones
Hi,
I have an incoming call that I would like answered every time by a
different SIP phone (out of 50).
Also, some of the phone may not be available (may be turned off and thus
unregistered with Asterisk).
Any way of doing this?
Paulo H. Mannheimer
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2003 Oct 17
5
Beta testers for visual configuration tool for asterisk
Hi All,
We've been developing for a while an IDE for Asterisk, and the time has
come to open it for beta testers.
You can check at www.instant.com.br/viv.html for a snapshot of the
application.
Current modules are Dialplan and VoiceMail configuration. As you may
see, it is all-visual, with drag and drop support and integrated sound
recording, saving and cross-checking, so you dialpland
2003 Jul 30
4
SCO/Linux concerns
Hello
Since I am getting a bit concerned about the SCO vs IBM issue, I was
wondering if can I can setup Asterisk on FreeBSD is it supported ?
Are drivers for Digium cards available on FreeBSD ?
Thanks
Ajit
----- Original Message -----
From: <asterisk-users-request@lists.digium.com>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, July 30, 2003 3:05 PM
Subject: Asterisk-Users
2003 May 29
0
Large Asterisk installations
Hi,
Could anyone elaborate on using * on medium/large sized installations
(more than 50 channels)?
I'm interested in learning more about performance, stability,
scaleability, fault tolerance, and what hardware are you using.
Thanks!
Paulo H. Mannheimer
Instant Solutions
+55 21 2512.7999
+55 21 9225.9395
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2003 Jun 30
0
stuck channel
I'm getting this intermittent problem, sometimes a zap channel gets
stuck after a call. Below is a snapshot of the channel. Any ideas what
can be happening?
Name: Zap/1-1
Type: Zap
UniqueID: 1056988772.10
Caller ID: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 0
NativeFormat: 68
WriteFormat: 4
ReadFormat: 4
1st File Descriptor:
2003 Jul 22
0
new voicemail messages
Hi,
I'm localizing the voicemail messages to Portuguese. To make it possible
for another person to translate it, I've set up a couple of extensions
that call the following macro for each message on the system. After
recording, I can perfectly hear each message using Playback.
When I try to play the new recorded message using VoiceMailMain, I can't
hear the new message (line goes
2003 Jul 22
3
busydetect and random hangups
Hi,
I'm having random hangup problems with zap channels.
If I turn busydetect off in Zapata.conf, * fails completely to detect a
user hangup in the middle of a script.
On the other hand, if I turn it on, everything works much better, but
long calls tend to be hung up without a motive.
Any other parameter that I can try? Any #define that I can tweak and
recompile?
Will callprogress
2003 Dec 10
3
pridump
Hi All,
Can anyone tell me what are the <dev1> <dev2> parameters that I should
use to run pridump? I took a look at the source code but couldn't figure
this one out.
Best,
PauloHM
2003 Jun 03
0
Asterisk localization
Hi All,
I've been working with asterisk for about two months, and I would like
to contribute to the project on the localization side, mostly making it
easier to translate text output and pre-recorded messages.
My goal is to discuss with you guys a framework for
localization/translation, and then slowly start to implement it. I
believe it's important to discuss this before any code is
2003 Sep 11
3
SIP busy
Hi,
I would like * to treat a SIP extension as a normal extension, when it
comes to the busy functionality. In other words, if someone tries to
call the SIP phone and there is already an ongoing conversation, the new
caller should get a busy message/tone
Is there any parameter that I can set? Is this something that should be
configured at my softphone?
Best,
PHM
2003 Dec 11
1
Iax, Iax2 and Iaxcomm
Hi,
I'm trying to use iaxcomm. I can place a call from the softphone, but
when I place a call to it, when I answer I get ...
NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping
incompatible voice frame on IAX2[paulohm]/3 of format GSM since our
native format has changed to ALAW
My iax.conf looks like this ..
[paulohm]
type=friend
host=dynamic
username=...
secret=...
2003 Sep 03
2
E1 problems
Hi,
I'm testing an E1 with E&M signaling. Some of the problems I'm running
into are the following:
1) if I try to configure any channel above channel 15, I start
getting a "multiframe alignment error" on my telco test equipment. So I
have my zaptel file only configured for 15 channels, like this
span=1,1,0,cas,hdb3
e&m=1-15
2) When the test equipment tries to send me
2003 Nov 26
1
Pbx / channel bank install
Hi all,
We are about to make our first channel bank install. This will be a one
PRI outside connection and up to 70 extensions.
As the schedule (and the budget) is pretty tight, I would like to learn
a little bit more about general experiences with channel banks, like
echo cancellation problems, Caller ID usage, etc.
TIA,
Paulohm