Displaying 20 results from an estimated 1000 matches similar to: "SIP Errors"
2003 Aug 02
1
SIP app_queue
I noticed a few issues with app_queue just wanted to know if its sip
related or ata186 related:
Ext 111 and Ext 112 are dynamically loged into the queue via
AddQueueMember.
Call hits queue with fewestcalls routing.
Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some
reason ext 112 doesn't answer it rings back to 111. Again at this point
ext 111 isn't answered it
2004 Feb 02
1
Voicetronix Audio Problems when making two or more simultanoues calls
Hi there,
Besides the problem of Voicetronix dialing too early before the carrier gives a dial tone, there also appears to be issues with the audio quality when more than 1 channel is utilized.
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
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 Oct 04
2
300 extensions on Asterisk?
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello
I am running an * box with just 8 extensions connected to our old Alcatel BCN
5200 PABX.
The requirement is that we now scale it up to handle about 300 lines and get
rid of our old PABX. Is there a way of hooking up 300 phones to asterisk
without going via the PABX. I am more of a network person than a telecomms
one so i may not fully
2005 Jan 13
2
SMS Gateway
Does anyone know of any companies where I can interconnect with for SMS?
?
.o-------------------------------------------------------o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
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
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323connections
are you giving answer()?
..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
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,
2004 Sep 28
1
Looking for whoever wrote cdr_mysql
I don't completely understand this.. Lemme try it out..
[default]
exten => 1112223333,1,Macro(happy-did)
[macro-happy-did]
exten => s,1,Goto(${MACRO_EXTEN},1)
exten => _XXXXXXXXXX,1,NoOp(Normal "s" exten stuff here)
So when this is ran it will cut the cdr and the s will show the actual
DID not the s correct? But then the NoOp would be something like:
....
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
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 May 20
3
Help with follow me
I hope someone can help me with this. This is what I want to happen.
Someone dials in and goes to my extension.
First, the phone on my desk rings
If there is not an answer, I would like to have the dialplan call my cell
phone. If I answer my cell phone, speak the incomming number to me. I
press one of the buttons on my cell phone to accept the call.
If I don't answer, or I don't
2005 Jan 12
2
Trouble building appradius
I am having trouble building appradius from http://appradius.minitelecom.org/
I configure, make, make install cpprad-1.0, but when I configure, then
make appradius I get :-
obelix:/usr/src/appradius/appradius1.0 # make
make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory
2005 Jul 14
1
PSTN to SIP gateway
I've been looking through the examples and docs, but haven't yet quite
figured out how to do what I want.
We've got a T1 coming in carrying a block of telephone numbers,
terminating on an Asterisk box. Any call that comes in needs to get
sent to a SIP proxy, with a particular extension format:
*ANI*DNIS*@sipproxy.address
The closest I can see to do this with the Dial() command is:
2009 Dec 27
2
Call ends when picked up
Hello list.
My phone rings, I pick up, and the conversation is terminated. Every
time.
The setup :
Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server
--> ITSP
Could it be the SIP proxy of my Endian firewall ??
I have 4 accounts on the Grandstream which listen on port 5060 --> 5063.
They have a proxy defined namely my Endian firewall.
On this SIPproxy I have a
2005 Jul 11
2
Unable to dial certain calls
To begin with, I am new to both asterisk and VOIP and although I've
gotten pretty far with my Asterisk setup and have two different sip
accounts working fine for outgoing calls I am having trouble with one
issue.
My problem is that I have another provider who uses IAX2 that I wish
to use for calling various countries, including local (The
Netherlands) calls and calls to the US to
2004 Apr 30
1
Error compiling asterisk-oh323-0.6.0
Hi together,
i try to compile astrisk-oh323 like described in the Readme
- pwlib V1.6.6 (janus)
- openh323 V1.13.5 (janus) with make-patch
- asterisk V0.9.0
i got the following error
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE
-I/usr/src/redhat/BUILD/asterisk/include -I../wrapper -g -c -o chan_oh323.o
chan_oh323.c
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 Oct 26
6
voicemail.conf
I have delete=yes and attach=yes. But my messages are not getting
deleted after they're sent. I'm running asterisk as root so it can't be
a permission issue. Any ideas?