Displaying 20 results from an estimated 2000 matches similar to: "SIP info"
2003 Jul 18
0
FW: Sip codec preferences
Did anyone have a way to make codec negotiation work with Asterisk?
This is something I would love to have working as well.
I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711.
-----Original Message-----
From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it]
Sent: Wednesday, July 16, 2003 11:32 AM
To:
2003 Nov 23
2
SIP Express Router & Asterisk
Greetings...
We've been having some interoperability issues between Asterisk and an
AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000
somewhere. So, I've been pondering using iptel.org's SIP server (SIP
Express Router) as a "front end" for PSTN calls going out to the Mediant,
while using Asterisk for everything else.
Has anyone done something similar, or
2003 Oct 12
2
INFO method and DTMF translation
Hello guys,
I have searched high and low, but not found any information about
rules of using DTMF in SIP INFO method. Cisco has described something with
Signal=, but it look like this feature is dependent on implementors?
The problem is chan_sip.c cannot correctly translate received DTMF
digits, especially #,*. At least with my Antek EGW-804 gateway.
Looking into chan_sip.c, I found this code:
2003 Nov 05
12
Mediatrix 1204
I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following.
PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0
2003 Jul 14
3
New budgetone firmware
Hi.
Has anyone experienced with the new firmware .77 ?
There's Day Light Saving time now, but haven't
time to play with it, till now.
Matteo.
--
Matteo Brancaleoni
Espia System Administrator - IT services
Website : http://www.espia.it
Email : mbrancaleoni@espia.it
2003 Jul 16
0
Sip codec preferences
Hi.
I'm experiencing a issue (not big, but important)
I have an asterisk installation with a buch of sip
phones & analog ones.
I have 2 1 sip phone that's outside in the "world",
and is nat'ed. I'm using g.729 with it.
I wanna use g.729 only for the remote phone, and ulaw
for the local ones, since they're on a lan.
What happens? when I call the remote phone, g.729
2003 Nov 07
2
Callgroups and Pickupgroups in Console/dsp
Hi all.
I've made a patch for chan_oss.c to enable
callgroups and pickupgroups in it (since wasn't enabled).
I needed it for a special use of the console (pickup
calls arriving to the console from another phone)
btw, If someone is interested, I can submit a patch
to the bugtracker. I won't do it until
that's usefult for someone... since is a very special
features that probably no
2003 Oct 07
3
Line going to Zombie
I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says:
astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r
I have tried to release it with soft hangup Zap/1
& also soft hangup
2003 Jun 22
2
How can I log SIP debug messages to a file?
Hi everybody,
I want to read to debug messages and try to interpret them but they happen
too fast, how can I log these guys to a file, or is there a file like this
already?
I checked the /var/log/asterisk but there isn't much interesting there yet?
How can i turn on logging for SIP,IAX and other things?
Thanks,
Umut
2003 Jul 11
1
SIP call from one extention to another
Hi
I am trying to call from Linphone on extention 109 to Xlite on extention 108
and I get this error
----------------------
to 216.75.167.18:5068
WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application
'Dial ' for extension (sip, 108, 1)
== Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43'
---------------------
Can you tell me what
2003 Oct 16
1
VoIP Monitor
Hi all!
I am looking for some free software to monitoring all the calls that are being
done in my network. Which telephone are connected, how long are the calls,
quality of service, bandwidht,etc.
If someone knows about a good one, plesea tell me.
Regards,
Mireia
2003 Oct 24
1
Questions about Zapateller and Privacy Manager
Hey all...I'm just getting my * setup and right now all I have is an FXO
but no FXS. I wan't to get rid of telemarketers by having * pick up the
phone if there is no CID present, give the caller the Zapateller tones
and then ask the user to input their phone number via Privacy Manager
(yes I realize that this won't get us any where given that I can't
re-ring the phones without FXS
2003 Jul 23
1
Cisco 7960 upgrade from SKINNY load
Here's a clip of comments lifted from a Cisco bug list. This will
be perhaps useful to those of you who have just purchased a Cisco
phone off eBay.
JT
-------------
(1) Short problem description:
Documentation on how to load SIP image on phone with skinny software
(2) Longer problem description (what happens):
If the phone is loaded with the Cisco Skinny code, then there is a
small
2003 Oct 17
4
Using channel banks
Hello Everyone,
What kind of hardware setup would I need to do if I want a T1 connection to PSTN
and have 48 users in office with analog phones. Will something work if I have a
T410P card in asterisk and have one T1 going to PSTN and other two to a channel
bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks.
Deepak
2003 Jun 13
3
Call queues for phone operator
Hi.
I was wondering how can I make incoming calls to wait if the phone
operator is busy. I've 8 incoming lines, with 30 extensions.
What I need is if the operator is busy with call nr #1 , the new
incoming call waits until the op. is free.
Looking into app_queue seems the way to go.
So I want to ask if I'm right or wrong:
I set up only a queue , is to say operatorq, where
the only member
2004 Feb 03
2
Playing announcement to called user prior toConfirmation
I wish 'A(x)' was available with AgentCallBackLogin!! :(
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matteo
Brancaleoni
Sent: Tuesday, February 03, 2004 6:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Playing announcement to called user prior
toConfirmation
show application
2003 Apr 08
1
Wiki for the * community.
Hi 2 all.
I was thinking to start a little web site with phpwiki,
to let the * community build a sort of shared
documentation 'bout * & related.
That because in a wiki "place" all grows faster,
and is also the right place to share experiences.
For example it's right to have documentation
about * installations, ie who has done what with asterisk
Till now we don't know
2003 May 25
2
Message Waiting and VoiceMail 2
Hi.
I noticed that if new messages are recorded
with voicemail2 , they're not detected by
the message waiting indicator, so
the mailbox=XXXX param has no effect, and
no message waiting is sent to the phone
(sip & adsi, or stutter dialtone)
Any hint?
--
Brancaleoni Matteo <mbrancaleoni@espia.it>
Espia - Emmgi Srl
2003 May 01
2
Max number of connection in IAX ?
Hi.
I was wondering if there's a parameter to limit
the number of concurrent sessions in IAX, globally or
on a per-user basis.
That could be needed for security purposes
(to prevent dos attacks), to limit bandwidth / cpu usage, or
to not allow more than N guest connections, for example.
Any other VoIP channel support that?
(like SIP, MGCP)
Matteo.
--
Brancaleoni Matteo
2003 Apr 18
1
Account code on SIP
I was wondering if the accountcode flag works
with sip channels. I was looking into the
debug and ,even if I have the line accountcode=XXX
into the users sections of my sip.conf, I don't see
it logged into the cdr.
Matteo Brancaleoni
mbrancaleoni@espia.it
Emmegi System Administrator
EspiA - EMMEGI Srl - e*solution provider
Uffici: Via Pascoli, 37
20129 Milano - Italy
Sede Legale: Corso