Displaying 20 results from an estimated 26 matches for "mysipprovider".
2004 Nov 27
0
Failed to WWW-authenticate on INVITE
...to a SIP Express router.
Inbound calls to my asterisk server works just fine, but when i try to
make outbound calls I get the following error message:
Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to
WWW-authenticate on INVITE to '"username"
<sip:username@mysipprovider>;tag=as5399a078'
I'm currently running Asterisk CVS-HEAD-11/27/04-15:07:27 and chan_sip2.c
downloaded today. I had the problem using chan_sip as well, but the
error message was slightly different.
Below is a packet dump from the SIP session.
My current sip.conf configuration is:
---...
2005 Sep 14
2
Starting From Scratch
...k box with SER (which I don't begin to understand
yet) at the office. In order to try to understand how all this works,
I have stripped my extensions.conf down to almost nothing. I am
building it up piece by piece. This is the entirety of my
extensions.conf file:
[globals]
OUTBOUNDTRUNK=SIP/mysipprovider.com
[from-internal]
exten => 105,1,Answer()
exten => 105,2,Playback(abandon-all-hope)
exten => 105,3,Hangup()
exten => 106,1,Dial(${OUTBOUNDTRUNK}/916xxx6000)
exten => 107,1,Dial(${OUTBOUNDTRUNK}/916xxx2128)
This is all just testing. When I dial 105 from either of my
softphones, i...
2003 Nov 24
0
SIP channel modification
If you update your source from the CVS, you'll get a new SIP channel
that supports a new syntax for SIP calls in extensions.conf
If you define a SIP peer in sip conf, like
[mysipprovider]
...
You can now use
dial(SIP/mysipprovider/extension)
Where the part "mysipprovider" is related to the sip.conf section.
Also, you can dial any SIP URL by
dial(SIP/extension@domain)
like
dial(SIP/oej@edvina.net)
In this case, "domain" referes to a Interne...
2004 Jun 16
5
Failed to authenticate on INVITE
Hi,
I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
These two boxes talk to eachother via sip, not iax. Since the upgrade, I
get the error "Failed to authenticate on INVITE" trying to make calls to/from
either box. Removing the secret from each box's sip config seems to work but
is utterly braindead.
Has anyone seen this?
- Eric
2006 Mar 29
5
Asterisk Between PBX and FXS
Hi guys,
I''m setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure out
in asterisk which extension was dialed before the call came to asterisk?
2003 Mar 14
3
SIP registrations
...m". Rather
than the entry/username/password that is setup in the sip.conf file.
That way a user could log into any SIP enable client and their calls
would follow them around.
I have read the sip.conf man pages and have noticed the following
commented lines in sip.conf:
;
;register => 1234@mysipprovider.com ; Register with a SIP provider
;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider
as 1234 here.
;
But I am a little confused how this should be implemented.
--
Steve Woolley
ADS Telecom, Inc.
59 Skyline Drive #1250
Lake Mary, FL 32746
Phone: (407) 682-6226 x...
2003 Mar 09
6
DTMF detection on SIP provider ?
Hi..
I just wondering why DTMF are not recognized by aterisk on incoming calls
from my SIP provider ...
ANy suggesteions ?`
/Mike
2003 Sep 18
2
SIP, X-Lite
...ti
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; Disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
;
;register => 1234@mysipprovider.com ; Register with a SIP provider
;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as
123
;
[sip7101]
context=sip
type=friend
secret=blah
auth=md5
; defaultip=192.168.5.10
host=dynamic
dtmfmode=inband
mailbox=7101
--
Using M2, Opera's revolutionary e-mail cl...
2004 Nov 26
1
direct asterisk to asterisk SIP calls without external SIP provider
...ncluding an external SIP provider, where
both RGs would register to that provider with a telephone number and
they could call each other via that telephone number. Each RG had a line
register => <telephone number>:<password>@sip.myprovider.com
in sip.conf. I also included a section
[mysipprovider]
type=peer
context=fromINTERNET
host=sip.myprovider.com
and used Dial(SIP/<telephone number>@mysipprovider) in my dialplan.
Context fromINTERNET only consisted of
exten => s,1,Dial(FXSport/0,,tH)
This setup was working great, but now I want to have the two RGs
communicate directly to eac...
2010 Dec 14
1
Asterisk + VOSP account working configuration?
...context = dummy
deny=0.0.0.0/0
permit=<IP address of VOSP server>
externip=<public IP address of NAT router>
localnet=192.168.0.0/24
disallow=all
allow=ulaw
allow=alaw
allow=gsm
;all RTP packets go through Asterisk
canreinvite=no
;incoming calls from VOSP
register => me:mypasswd at mysipprovider.com
;for outgoing calls to VOSP
[vosp]
;friend = peer+user
type=friend
username=me
fromuser=me
fromdomain=mysipprovider.com
authname=me
secret=mypasswd
host=mysipprovider.com
insecure=very
qualify=yes
context=outgoing
;Since VOSP is on the Net, nat=no or nat=yes?
nat=no
;extension for XLite
[6011...
2006 Apr 19
1
Fwd: sip.conf and jump from register to the extension
Hi,
the documentation of sip.conf is telling me this:
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
In reality it jumps to the extension 1234 in the context and not to s
So it is much more complicate to write an proper dialplan.
Is this an bug or is the documentation not up to date?
best regards
Thomas
2011 Jun 06
0
About Asterisk SIP NAT Config
...-------------------------------------------------------------------------------------
[My sip.conf in Asterisk]
[general]
; SIP Client Config - Start
externip=*.*.*.*
localnet=192.168.100.0/255.255.255.0
register => 1000:jrcyagi at 192.168.100.1/1234
; To make a call to a external SIP server
[mysipprovider-out]
type=friend
secret=jrcyagi
username=1000
host=jrc.nwt.com
fromuser=1000
fromdomain=jrc.nwt.com
canreinvite=no
insecure=very
qualify=yes
nat=yes
context=from-mysipprovider
; is further defined in extensions.conf
; SIP Client Config - End
; SIP Server Config - Start
[1000]
type=friend
secret=j...
2004 Nov 22
0
How to configure the Asterisk server such that a FXS phone can talk to SIP client?
...e type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
disallow=all ; Disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
;VoWLAN testing!
allow=ulaw
allow=alaw
allow=gsm
;allow=all
;
;register => 1234@mysipprovider.com ; Register with a SIP provider
;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here.
;
;[snomsip]
;type=friend
;secret=blah
;host=dynamic
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59
;mailbox=1234,2345...
2004 Apr 23
6
Polycom registration
I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone
registered on an asterisk box but am having no luck. I get the
following errors 192.168.22.196 being the phone and 22.254 being the
asterisk box..
Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request:
Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for
2006 Jan 28
3
Multiple Subscriptions to SIP accounts at Same Domain
Sorry not to have observed etiquet and lurked here for a bit before
wading in with a question but I have an issue that may well be because
I dont know enough about what asterisk is actually doing under the hood
to understand why I cant do what I want with asterisk.
Im hoping that someone can point me in the right direction :-)
This is what I have:
Mandrake 2006 running Asterisk 1.2.3 - no
2003 May 13
1
beginner's question!
...nstalled for testing. I don't have any extra hardware installed so far, was attempting to just try out connectivity. I am having some probs with the configuration, maybe someone out there can give me some tips :
firstly on modifying the sip.conf file I got stuck at the line
register => 1234@mysipprovider.com
What exactly is a SIP provider? is this essential? Leaving the line as it was in the
sample config file, asterisk crashes the machine after trying to read the SIP.conf
(Crashes to the extent that the machine freezes .. )
What I would *like* the system to do is as follows :
for now, just take...
2009 Dec 30
1
Monitoring SIP & Skype connections
I have an Asterisk 1.4.2 server with 3 different SIP providers and
Asterisk for Skype gateway installed. Periodically the SIP providers go
offline for some reason, or the Skype connection fails.
When this happens, I lose my SIP registration to the provider.
Unfortunately I don't know this has happened until someone eventually
contacts me to say, "I tried to call you but it
2005 Aug 20
2
Realtime sip_buddies "register=>" how?
Hi all
I've been doing some testing on realtime using mysql, an have a little
question that could not find the answer to or maybe its not posible at this
time.
Is there a way use "register=>......" on a DB using realtime. For the moment I
use it in sip.conf. It will help me a lot if this could be store on a DB
somehow.
commets or sugestions .... ?
thanks
Billy
2003 Aug 26
0
TDM10M && Siemens Euroset 2015
...; Max length of incoming registration we
allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in
NOTIFY
;
;register => me@mysipproxy.org ; Register with a SIP provider
;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider
as 1234 here.
;
[snom1]
type=friend
host=129.26.10.121
dtmfmode=rfc2833
mailbox=2101
reinvite=no
context=local
callerid="Studenten 1" <2101>
[snom2]
type=friend
host=129.26.10.122
dtmfmode=rfc2833
mailbox=2102
reinvite=no
context=local
call...
2006 Dec 18
0
pap2/wrt54gs/asterisk
...peer/user also
;notifyringing = yes ; Notify subscriptions on RINGING state
GNU nano 1.3.8 File: sip.conf
; (provider).
;
; host is either a host name defined in DNS or the name of a section defined
; below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;registertimeout=20 ; retry registration calls every 20
seconds (default)
;registerattempts=10 ; Number of registration attempts before
we give up...