Displaying 20 results from an estimated 2000 matches similar to: "Sip to Sip"
2005 Jun 30
5
Logrotate
I created some scripts to logrotate. I am having a problem. After I do
it, I am sending kill -HUP to the process
its not using the newly created messages file again. Could someone help
me out with how I can rotate asterisk's
log's without killing the process?
..o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL
2004 Nov 22
1
Tinc on OsX, partial success
I have now got the tinc demons (on network OFFICES) on BranchB and
BranchA talking to each other, see below for log from BranchB. For
some trouble shouting issues relating to OsX see at the end of my
e-mail.
However, I have not yet achieved the network connectivity/routing
that I would like.
The aim is:
BranchB is a laptop
I would like to connect it (via tinc) to my office network, so that
2005 Aug 10
2
Help with TNT and Asterisk
Im having some problems with connecting a TNT to asterisk. The problem
is
when the call is sent to asterisk and signaling is done the RTP syncs
however
no audio is produced. Can someone give me some idea of how to
accomplish this?
I am using the standard configs and g711 and 729 do the same. No audio.
Public IPs on both ends. No nat. Any ideas would be appreciated.
2004 Nov 16
1
Tinc on MacOs X
My intention is to set up tinc so that I can connect from home to an
office network. All CPUs are running MacOs X, 10.2.8 or 10.3.5.
I have read the tinc manual, tincd.8 and tinc.conf.5. However, I am
still unclear about a few issues.
First and foremost, how to I setup the VPN interfaces on the hosts,
on MacOs X? Which file do I have to alter and what is the exact
syntax on Os X to setup
2005 Oct 06
14
www.openpbx.org
Hello,
What do you think of this project www.openpbx.org ?
Something like ser and openser !
Kinds Regards
Harry
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
T?l?chargez cette version sur http://fr.messenger.yahoo.com
2005 Jun 30
1
Cisco Voip Question
Does anyone in here know how to setup auto negotiation between g729 and
g711ulaw on
a cisco 5400? I would imagine it would be the same on a 3660.
The problem I am having is natively the call is setup for g729 however
when the call is transferred
to voicemail it uses ULAW so when the cisco tries to connect to the
voice mail I get a SIP error
that the codec couldn't be negotiated. I need
2006 May 17
2
Diverse servers
I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system
2005 Jul 01
1
Got SIP response 481 "Invalid CSeq Number" backfrom X.X.X.X
I had the same problem and I believe it was the payload size of the
codec. What code are you using?
..o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
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 Federico
Alves
Sent: Friday,
2005 Aug 16
1
DTMF, Asterisk, External PSTN gateway, and PAP2 (was: RE: Issue with DTMF Tones - CodecIssues)
I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF.
works very well and have never had a problem with it.
..o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
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
2005 Jul 13
6
OT: DS3 -> VoIP Hardware Recommendations
Hello all,
We are looking for some hardware requirements/recommendations to be
able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would
bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then
need to convert those calls into G729 SIP VoIP calls to send to our
asterisk box over ethernet. Since everything is going in/out of asterisk
is 729, and no features
2004 Nov 24
1
Tinc on OsX, slowly getting there....
I have now got the tinc demons (on network OFFICES) on BranchB and
BranchA talking to each other, and I have been able to access
services (like AFP, Apple file sharing) between the two hosts (in
both directions), but not beyond the local network connected to
BanchA.
I am unclear which routing can be provided by tinc and which routing
would I have to add manually.
The aim is:
BranchB is a
2005 Sep 11
5
rotate * log file?
Running fc3 with current cvs-head...
Is there a nice way to rotate the /var/log/asterisk/messages file without
shutting down asterisk?
I'm currently rotating the log files via cron, however my script requires
asterisk to be shut down, which also kills any outstanding cli sessions
(eg, asterisk -rvvvvv). Would like to rotate the files without killing
the cli session. Any reasonable way to
2006 May 22
10
US telco lingo
Could someone explain to a non-US dummy the following phrases I have seen on
the list.
"I can provide you with tier 1 termination 6/6. I can blend or NPANXX
breakout."
"We provide US48 termination, blended rate for 1 MOU and above is .008 with
6/6."
What is 6/6?
What is US48?
What is blended?
What is MOU?
What is NPANXX breakout?
-------------- next part --------------
2005 Jun 14
6
VOIP-INFO down?
Seems to be all morning. I have not been able to access for several
hours now.
W
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Marcel van
Kaam, Fonetica
Sent: Tuesday, June 14, 2005 7:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] VOIP-INFO down?
Hi
2005 Jul 18
2
Comments on Areski Calling Card Solution plz
Hi,
can anyone who has the Areski Calling Card solution on Asterisk
working comment on it? Is is stable enough for a production system?
Any pros and cons?
thx,
Arnd
2005 Jun 06
2
Variables and status problems in AGI application
I am running a prepaid application with Asterisk. When authentication has to be
done by DTMF everything works fine. However when the user is authenticated
directly from the sip phone, the channel variables seems to disappear.
Trying to retrieve the channel status always returns -1 instead of the 6 that
happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME
variables.
The
2005 May 23
4
Programs to parse queue_log
What third party programs are available for parsing the queue_log file
and CDR file? I know about XC-AST, but management would prefer a php
based solution.
What have other admins done to retrieve detailed call information about
the queue system? Anyone develop their own that they don't mind sharing?
--johann
2005 Jul 05
2
PRI or Trunk monitoring
Did someone monitor the PRI's or trunks some way?
I tried with MRTG and Andrea Fino module but it never worked for me.
Any other experience? I want to track the use of my PRI's and trunks using
graphical as MRTG does each 5 minute, day, week & Year.
But the option of the 5 Minutes I don't think is usefull, We need something
more realtime.
Thanks,
Carlos Alperin
2006 Jun 12
2
Cell gateway for T-Mobile US??
Most gateways I have found are only sold overseas.
Do these work in the US?
My provider is T-Mobile (using their Blackberries).
They support:
GSM (I am pretty sure)
GPRS
EDGE
We get unlimited Cell to Cell minutes and would like to leverage the
possible savings.
Does anyone know of a product that they have been happy with?
SIP or Analog is fine although SIP (or IAX) is preferred for the
2005 Oct 04
2
DPH-140S SIP Phone oddities
Hi, list!
I'm playing on an Asterisk@home installation, since a month or two.
I've had no trouble setting it up 'n running.
I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk.
>From this phones, I can make & receive calls with no trouble, but, when I
try to use some "interactive" function (eg Directory or Voicemail), the
phone seems