Displaying 20 results from an estimated 7000 matches similar to: "help with sip registration"
2007 Apr 19
1
Asterisk 1.4.2 connection to Nortel CS1000M -followup with log
Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine.
However, using outgoing call files the CS1000 is hanging up after I answer the call.
I dont know why?
Thanks, for any assistance.
Jerry
my sip.conf entry is:
[Nortel]
type=friend
dtmfmode=rfc2833
username=XXXXXXXXX
disallow=all
allow=ulaw
allow=alaw
2008 Sep 26
2
server and 2 uniden phones no ringing
I have a box running asterisk 1.4.17 that had been working.
it has 2 uniden phones connected on it.
This was working and now the phones dont ring when calling each other.
below is the sip debug. I cant see why the other phone does not ring?
I also tried changing the canreinvite for no to yes but that made no
difference after restarting.
Very simple network. server, linksys router and 2 phones.
2004 May 18
0
No luck using asterisk as proxy...
Still no luck using asterisk as a proxy.
48 hours solid working on this. I'm beginning to think asterisk isn't
going
to be compatible with the provider I'm using :(
Has anyone got *any* clues as to what can cause this message? It's
definately
provider specific (voiptalk works, pipecall doesn't) but confusingly
seems to
be caused by something in the client phone app.
I
2007 Apr 18
2
incoming SIP call
Hello all,
I'm having a quite simple configuration like:
SIP provider <=> asterisk SIP <=> lan
Everythings works fine but sometime I can't get incoming call.
here are some of the logs from set debug 25 set verbosity 25 sip show
debug and sip.conf and a part of extension.conf
thanks in advance
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net
2005 Mar 12
1
Broadvoice outgoing problems
Hello All,
I'm just getting into *, and trying to use a Broadvoice account. It works inbound, but Outbound fails no matter what sip.conf parameters I try. From the recent posts here I think it could be:
A bad CVS release - I will try to download and build from a new one
Broadvoice not challenging and/or Asterisk not responding with an Authorization: in the INVITE header.
I am
2009 Apr 10
0
IVR and DTMF
REPOSTED with MORE Info and Modified Subject Line:
--------------------------------------------------------
I am using one of the Minute Provider to dial out USA numbers.
Now in one of my process, we need to Dial IVR and the enter DTMF digit and
then it connects to the automated IVR.
When I dial out the IVR directly using Xlite and VOIP Mins provider , it
works perfectly. but when In try from
2007 Sep 25
1
Help with Sip Registration
Hi all,
I have installed X-lite client on a windowsXP
machine and asterisk on an enterprise linux m/c.
The client is sending a registration message to asterisk
server. It is able to process the message and sends 200 OK
back. But later it says "Scheduling destruction of sip
dialog xxxx ". Then it says "Really destroying sip
dialog xxxx". What to do for this problem??? I
2009 Jun 25
1
SIP registration fails
Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports
opened and 5060 forwarded to Asterisk (192.168.2.2)
Can someone see why SIP-registration fails ??
register => 092779077:XXXX at 85.119.188.3
[3starsnet]
type=peer
host=85.119.188.3
username=092779077
secret=XXXX
fromuser=092779077
fromdomain=sip.3starsnet.com
dtmfmode=rfc2833
canreinvite=no
insecure=port,invite
qualify=yes
2007 May 04
0
Asterisk registration SIP confusion. Can someone explain this?
We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The
registration succeeds, and is confirmed with SIP SHOW REGISTER. However,
we frequently (every few minutes) see this on our console:
REGISTER attempt 1 to 999@pbx.itsp.com
REGISTER attempt 2 to 999@pbx.itsp.com
Any ideas what is going on? In particular
1. What causes the two register attempt messages above?
2. Why
2009 Sep 12
1
E65 fails registration, soft phone works
Hey folks,
I am trying to get an E65 to connect to asterisk, and I would really
appreciate a second set of eyes. The SIP dialog completes fine, but
the phone subsequently says "Registration failed".
I am in a network that has what seems to be a SIP-capable NAT
gateway, but the asterisk is configured nat=yes anyway. Using
a softphone (twinkle), I can connect just fine, SIP and RTP work.
2009 Sep 30
0
PBXNSIP Registration Issue
I've got PBXNSIP running on a windows box and it is trying to register with
my Asterisk box. I can set up one trunk and it works fine, however if I try
to setup a second trunk from the same box, there is some sort of
authentication issue where Asterisk appears to be confusing which trunk is
which. Here is the chunk from my sip.conf:
[TEST1]
context=STUFF-LD
type=friend
2005 Mar 22
0
Still no Broadvoice Outbound. (Bump)
I'm still not getting my outbound to work. I've seen two patches
relevant to broadvoice for chan_sip.c which apparently have already been
added to CVS. I'm dropping all outgoing calls after ~30 secs. Asterisk
doesn't seem to know they're gone though. I called my cell w/
broadvoice and turned on sip debug AFTer the call had physically dropped:
*CLI> sip show registry
2008 Apr 01
2
help with no audio
I am using asterisk 1.4.18 with a polycom phone.
sip.conf has:
[532]
type=friend
username=532
secret=XXX
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid=532
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
I call into the dialplan and try to play demo-congrats and I hear nothing.
Firewall is disabled.
Everything is on the 192.168.1.X network for this
2008 Mar 03
1
ekiga sip registration fails; externip no help
ekiga registration fails. I've set nat = yes ( also blank ) and i've set
externip. Anybody have a sip.conf that works?
Here's the sip debug:
Reliably Transmitting (NAT) to 86.64.162.35:5060:
REGISTER sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport
Max-Forwards: 70
From: <sip:test at ekiga.net>;tag=as64618445
To: <sip:test at
2006 Oct 17
0
lots of registrations, sip problem
Hello,
I've got a problem with connection to my SIP provider. In general,
everything works, but I get lots of these messages:
Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's
odd... Got a response on a call we dont know about. Cseq 42710 Cmd
SIP/2.0
Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request:
That's odd... Got a response on a call
2004 Jun 15
0
SIP Registration with Entice Softswitch
I'm having problems getting Asterisk SIP to register with an Entice
softswitch SIP Gateway. My provider tells me that all thats needed is a
user name, password and the IP address and to register and it needs to
be using MD5 authentication.
I continualy get a "603 Decline" message. The provider of the gateway
says they are not receiving any authentication information. Registration
2005 Mar 22
0
help with registration
I have a SIP account that I can successfully register with XTEN and a
Sipura-2000. I have yet to be able to get it to authorize with *.
My XTEN looks like:
Username: 001234
Password: xxxx
Authorization Username: 001234
Domain: domain.net
Register with domain:
2006 Jan 25
0
asterisk 1.2 with grandstream ht-496 2nd port registration issues
hi@all
I have the following problem:
With asterisk 1.09 the grandstream's registers fine with both ports,
with version 1.2.1 (the newest port on freebsd) I get "Unauthorized" SIP
messages from the 2nd port. The ports are configured identically, the
only difference is the sip and rtp port. On the first port the sip port
is 5060 on the second 5062. The rtp on the first 5004 on the
2005 Mar 05
0
Asterisk 1.0.3 Periodically Fails Registrations
Asterisk 1.0.3
Sayson 480i running .78 release
(problem may not be Sayson specific, it's just that's what's deployed)
Problem: Asterisk rejects registrations every so often even though
nothing has changed either with Sayson or Asterisk configuration (and
previous registrations have succeeded)
SIP trace of successful registration:
=============================
2008 Nov 07
2
help with dialplan
I have a small system, server, client and 2 phones. Phones are polycom
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that
calls another machine running asterisk - something ODD is happening.
; This is not working....
[smvoice-sip]
exten