-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ugur
GUNCER
Sent: Sunday, 3 April 2005 4:37 p.m.
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W
Hi all,
I bougth zyxel wifi phone but i cant register
when i want to
Hi all,
I bougth zyxel wifi phone but i cant register
when i want to register phone to asterisk i recieve
These errors I spend 6 hours to fix regist problem but i cant find the
solution
[9875]
type=friend
username=9875
secret=5789
host=dynamic
context=default
callerid="Ugur Guncer" <9875>
can
In the Grandstream setup, turn off "subscribe to message waiting
indication".
...or upgrade to CVS head, where I've fixed this problem with SUBSCRIBE.
Best regards,
/Olle
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Hi,
I am adding phones to my asterisk setup, until now i worked with some
softphones, with no problem,
I got some Grandstream BT100 phones, and see something strange in the
log, the on the phone's screen,
This is from the log :
Found peer '122'
Looking for 122 in default
Transmitting (no NAT):
Hi all,
I am using * as a PBX for a Broadvoice VoIP account. It had been working
well since about last November, although not perfectly (similar
disconnection problems, although I am pretty sure it had to do with my
PPPoE setup, but I think these issues were resolved). As of a few weeks
ago, thoug
I have this problem for 2 days and i dont understand
I am behind a nat
my sip.conf is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
disallow = all
allow= gsm
allow= ilbc
allow= ulaw
all
Alberto Martínez wrote:
Hello,
I am trying to register in asterisk with a softphone (x-lite) and I am
getting the following message:
Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request: Registration from 'tito
' failed for '192.168.1.5'
Just a guess, but the ip's don't match up.
[...]
I
I have tried uncommenting the section for xlite included in the sample
configuration file sip.conf and I can't register.
[xlite1]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
regexten=1234
Hello,
I am trying to register in asterisk with a softphone (x-lite) and I am
getting the following message:
Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request: Registration from
'tito ' failed for '192.168.1.5'
In the sip.conf file I have included the following. Does I need to
includ
Is there any benefit of increasing the registration period of a SIP
device ? I've seen periods of between 120 and 3600, and wondered why.
Julian
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trying to set up and configure a polycom soundpoint ip 500 phone, when trying
to get it to register with sip, i get the following message
Sip read:
REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" ;tag=
trying to set up and configure a polycom soundpoint ip 500 phone, when trying
to get it to register with sip, i get the following message
Sip read:
REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58
From: "138polycom" ;tag=
Hi all,
I encounter an annoying problem using Asterisk.
I 'm using SIP. I try to register an Asterisk as a SIP end user with
another Asterisk.
If I put both asterisk in the same local network, no problem to do it.
The asterisk end user registered perfectly with the other (let's call it
the regist
For some time (since pre 1.0), I've been seeing the following messages
fairly regularly from some, but not all, of my SIP devices:
Nov 23 06:37:59 NOTICE[2568]: chan_sip.c:7645 handle_request: Registration
from 'John Doe ' failed for '200.100.50.25'
I have a mix of Sipuras, Grandstreams, ZIPs and
Scott Laird wrote:
First, what's in your extensions.conf? That controls the flow of calls
once they get into the system. There should be a context that has
extensions for 1001 and 1002, and sip.conf should direct calls into that
extension via a 'context =' line.
Indeed, I had not changed the e
On Nov 3, 2004, at 4:16 PM, Ben Greear wrote:
Hello!
I have a Grandstream and a Cisco SIP phone, and I'm trying to make
a call between them. I added this to my sip.conf:
; Grandstream
[1001]
type=friend
host=dynamic
; cisco phone
[1002]
type=friend
host=dynamic
First, what's in your extensions.con
Hello!
I have a Grandstream and a Cisco SIP phone, and I'm trying to make
a call between them. I added this to my sip.conf:
; Grandstream
[1001]
type=friend
host=dynamic
; cisco phone
[1002]
type=friend
host=dynamic
It appears that they register correct:
grok*CLI> sip show peers
Name/usernameH
Hi people,
My asterisk wont register with any sip providers, I have tried three
different but they all end up with:
Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
There is no firewall and my server has a public IP. Cou
I set up my own STUN server and turned reinvite
off.
Lyle
- Original Message -
From:
[EMAIL PROTECTED]
To: '[EMAIL PROTECTED]'
Sent: Tuesday, August 31, 2004 8:53
AM
Subject: [Asterisk-Users] SIP
registration with public dynamic ip address
Hi,
Hi,
I'm trying to configure a natted budgetone
phone to a asterisk server as described in wiki using port forwarding.
I successfully make call from the client
but it seems it does not register the client ip address and when I try
to recall it is not reacheable.
Asterisk can manage natted sip cli
On Tue, 20 Jul 2004 23:50:05 +0200, Andy Powell
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> I've just (earlier today) updated from CVS so that I can apply the dtmf caller id
> patches. Unfortunately this has had an undesired effect.
I'm using * with an IX66 and no issues, with CVS head I suggest you
ha
Hi,
I've just (earlier today) updated from CVS so that I can apply the dtmf caller id
patches. Unfortunately this has had an undesired effect.
I have an intertex ix66 which up until the CVS update allowed me to register my *
server with the ix66 for my local domain (eg sip.mydomain.com). Now it
>From the wiki...
http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone
"If you are having problems with the phone losing registration periodically,
make sure that "SUBSCRIBE for MWI" is set to "No" in the phone's
configuration. This applies to at least version 1.0.4.55, possibly othe
Hi Folks,
I'm having problem with GS registering in Asterisk.
My setup is the following:
[1755]
type=friend
incominglimit=10
qualify=no
nat=yes
insecure=no
secret=X
dtmfmode=rfc2833
username=1755
host=dynamic
canreinvite=no
defaultip=192.168.0.1
context=sip-incoming
I have dozens of ph
Hello all,
I am very new to asterisk I have been using it for sometime
but now I want to maintain it myself I have built my own server and am trying
to get my cisco ata 186 to register I am having a problem I get this un 17
17:34:54 NOTICE[1116941120]: chan_sip.c:6715 handle_request: R
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
Hi All,
I am trying to Register Asterisk PBX to a SIP Server. But SIP Server
gives the following response to Asterisk: “400 Bad Request” .
Asterisk sends the Register Message to SIP server with the
URI: “sip:
domain_name_sip-server”. BUT
URI should be of the format: “sip:
EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Richard Neese
Sent: Wednesday, June 09, 2004 7:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP Registration seems to timeout
try changing your codec to ilbc and make sure that his gs has the latest
flash
to suppo
try changing your codec to ilbc and make sure that his gs has the latest flash
to support it.
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Hi,
I have an * server on a routable (public) IP address and a sip client behind
NAT using a Grandstream phone. He is connected through a bi-directional
satellite so he has a bit of latency involved. Usually I can dial this
extension and them to me. But I keep getting a registration failed mess
Hello everyone,
I'm currently attempting to get Asterisk properly registering through a NAT
proxy.
Here's the twist, the provider does not permit direct SIP messages to the
sip registry, instead they want registration to be done by their nat
traversal proxy, and when you send-out the registration
Brian Rathman wrote:
I am using snom200 phones registering with Asterisk via SIP. I can see
where the phone registers without a problem, and then when you try and
make a call I get a proxy authentication required message on the phone
and failed to authenticate user error in the Asterisk messages
: Friday, May 28, 2004 11:28
AMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] SIP Registration Problem
I am
using snom200 phones registering with Asterisk via SIP. I can see where the
phone registers without a problem, and then when you try and make a call I get
a proxy authentication
Title: Re: [Asterisk-Users] Wiki TOS - worrying for an open sourceproject?
I am
using snom200 phones registering with Asterisk via SIP. I can see where the
phone registers without a problem, and then when you try and make a call I get a
proxy authentication required message on the phone and f
Karl Brose wrote:
This is also closely related to Asterisk SIP's lack of proper [user
section] authentication/recognition for incoming calls. We've seen a lot
of posts here where new users have problems with this, but the real
problem is usually not acknowledged.
So tell me what's wrong with th
No and Yes, Olle. But mostly NO.
What Asterisk is doing actually depends on how it is configured. If you
are, by design, accepting calls for a particular [user] through the
default context from the general section in sip.conf it will generate
the correct response, but this is not because aster
Karl Brose wrote:
If the response to an OPTIONS is generated by a proxy server, the
proxy returns a 200 (OK), listing the capabilities of the server.
The response does not contain a message body.
Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
fields SHOULD be presen
for those who want to patch their SIP, here is a quck fix to make
Asterisk do a little better:
--- chan_sip.c 2004-05-16 01:33:06.0 -0400
+++ chan_sip.c_OPTIONS 2004-05-17 14:30:36.0 -0400
@@ -5916,6 +5916,7 @@
/* Initialize the context if it hasn't been already */
t: Re: [Asterisk-Users] Sip Registration Problem
Karl Brose wrote:
Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
not, Asterisk doesn't do it correctly either.
The host should respond with 200/OK if the call >could< succeed
theoretically if it were an INVITE or else
>>>I removed the qualify lines and sip reload [ed]. The extension still
>>>showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a
>>>full restart to get it to stop sending the OPTIONS messages.
>>>What did I do wrong here? How can I make a change to qualify without
>>>restarting?
> If a
D] On Behalf Of Olle E.
Johansson
Sent: Tuesday, May 25, 2004 1:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sip Registration Problem
Karl Brose wrote:
> Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
> not, Asterisk doesn't do it correctly either.
&
Karl Brose wrote:
Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
not, Asterisk doesn't do it correctly either.
The host should respond with 200/OK if the call >could< succeed
theoretically if it were an INVITE or else it should send a
404 or maybe a 487(? hmm, have to look)
It's a bug in Asterisk.
I believe it's still open also on the bugtracker. There are a few
reported senarios with these kind of problems.
Some of them where solved with the recent 'ast_gethostbyname' fix. Are
you running a recent version?
Btw, Ignoring OPTIONS is not a valid option (:-) whether s
Title: Message
Hi
All,
I had an unusual
problem today; I'm sure it's a configuration problem.
I had 2 phones
behind a nat device and I had qualify=300 in both extensions config. The device
I was talking to was an edgewater traffic shaper,/Sip Proxy. Since it is acting
as a sip proxy, it
Hi!
Registration only works if you have set "host=dynamic" for the client! In
case of a static host registration makes no sense, anyway! The only
purpose of registration is to tell the server at which IP address the
phone can be found.
Cheers, Philipp
Larry Keyes wrote:
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below. Anyo
Larry Keyes wrote:
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below. Any
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below. Anyone know what is
goi
Thanks for the info so far. I am still trying to asterisk'ize my ML9.2
firewall box and can't get the external SIP registration to work. If I
hook up my Sipura directly to the WAN it registers OK.
This is the message I get from asterisk:
Mar 8 21:03:07 NOTICE[196621]: chan_sip.c:3140 sip_reg_t
I have a question on SIP devices that are setup and working but you
change the login name and contents to them why does asterisk need to be
shut down and restarted for them to work? I have reloaded extensions
and done a reload command. But the updated sip phones do not work until
I shut down and
- Original Message -
From: "SW" <[EMAIL PROTECTED]>
To: "[EMAIL PROTECTED] Digium. Com" <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 1:47 PM
Subject: [Asterisk-Users] sip registration send out by asterisk
> Hi friends,
>
> I've notice
Hi friends,
I've noticed that first register message sent by * always get rejected by
the destination sip server. Then * sends a second registration message (
with Autherization section, and that get accepted by the destination host).
Why is this ?
Isnt there a way to tell * to send with Autotho
Hello All,
I am trying to get some ATA 188 units to register
with my Asterisk box over SIP. I continue to get the same "401
Unauthorized" Error when they try to register. If I turn Sip
registration off, I can use the phones without any problems with a static IP
assigned in my sip.conf fil
Hi!
I want to accept all the incoming calls (SIP) and
redirect them to the good extensions. How do I do
that? (Asterisk is acting as a SIP server then...
isn't it?
Thanks.
Best regards,
Mireia
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On Tue, 2003-09-30 at 20:21, Brian Capouch wrote:
> Does this imply that it will work even in a NAT environment?
>
> I have watched the list like a hawk for evidence of FWD working for
> machines placed behind NAT, but so far haven't seen that anyone could
> actually get it going.
>
> If so, t
Dave Cotton wrote:
I have SIP registrations working correctly for FWD and Sipphone, but it
is impossible to connect to Sipcall or Nikotel, I saw that someone on
the list has problems with ICH.
Does this imply that it will work even in a NAT environment?
I have watched the list like a hawk for ev
I have SIP registrations working correctly for FWD and Sipphone, but it
is impossible to connect to Sipcall or Nikotel, I saw that someone on
the list has problems with ICH.
To try and sort out the problem I tried to register to Sipcall with
Linphone and sent the dialogs to tech support of the eq
Hi all,
when I try register my netergy SIP Phone with *, I can't do it
due to the next message:
1 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a
From: "asterisk" ;tag=as34fa433f
To:
Contact:
Call-ID:
MAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, September 19, 2003 12:12 PM
Subject: Re: [Asterisk-Users] SIP registration between *'s
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th another SIP server.
>
> That's the matter.
> - Original Message -
> From: "Jamie Carl" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, September 19, 2003 12:12 PM
> Subject: Re: [Asterisk-Users] SIP registration between *&
>
Sent: Friday, September 19, 2003 12:12 PM
Subject: Re: [Asterisk-Users] SIP registration between *'s
> Why?
>
> Use IAX2, it is s much better...
>
> J
>
> On Fri, 19 Sep 2003 11:54:23 +0200
> "Xisco" <[EMAIL PROTECTED]> wrote:
> >Hi ever
Why?
Use IAX2, it is s much better...
J
On Fri, 19 Sep 2003 11:54:23 +0200
"Xisco" <[EMAIL PROTECTED]> wrote:
Hi everybody,
I'm trying to SIP register between two asterisk, each one
have a Public IP. Asterisk told me that Unathorizae
In * one sip.conf
register =>usuario1:pass1@
In
Hi everybody,
I'm trying to SIP register between two asterisk,
each one have a Public IP. Asterisk told me that Unathorizae
In * one
sip.conf
register
=>usuario1:pass1@
In * two
sip.conf
[usuario1]
type=friendusername=usuario1
secret=pass1host=dtmfmode=i
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 18 September 2003 19:04, Hielke Christian Braun wrote:
> try to change [siptestphone] to [atrg613test] in sip.conf. Maybe
> that helps.
It didn't. And now something else is weird. Asterisk fails sending audio to my
SIP phone. Found this
>
> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de Jan Janak
> Enviado el: viernes, 19 de septiembre de 2003 8:59
> Para: [EMAIL PROTECTED]
> Asunto: Re: [Asterisk-Users] SIP registration
>
>
> Hello,
>
> I don&#
PROTECTED]
Asunto: Re: [Asterisk-Users] SIP registration
Hello,
I don't know if it is the problem, but the message below is
syntactically invalid, there must be space between the name token in
>From and To (704) and the URI, i.e. correct From should look like this:
From: 704 ;tag=
PROTECTED]
Asunto: Re: [Asterisk-Users] SIP registration
Hello,
I don't know if it is the problem, but the message below is
syntactically invalid, there must be space between the name token in
>From and To (704) and the URI, i.e. correct From should look like this:
From: 704 ;tag=230b0-e0
nreinvite=no
> qualify=300
> nat=1
>
>
> ANY IDEA ABOUT THIS?
>
>
>
> srsergio
>
>
>
>
> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de Hielke
> Christian Braun
> Enviado el: jueves, 18 de sep
do el: jueves, 18 de septiembre de 2003 19:05
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] SIP registration
Hello,
try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that
helps.
Regards,
Christian.
On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote:
&
Hello,
try to change [siptestphone] to [atrg613test] in sip.conf. Maybe
that helps.
Regards,
Christian.
On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote:
> Hi,
>
> I'm having problems letting a SIP endpoint register at Asterisk. Here's the
> debug output from Asterisk:
>
>
>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm having problems letting a SIP endpoint register at Asterisk. Here's the
debug output from Asterisk:
Sip read:
REGISTER sip:s.s.s.s;transport=UDP SIP/2.0
User-Agent: ATI-RG613/1-1-0_8
From: atrg613test ;tag=AABcMQAMRhB0AAxx
To: atrg613test
C
sip show registry is when asterisk registers with some gateway.
you want to look at sip show peers or sip show users.
regards
Martin
On Thu, 31 Jul 2003, Steve Woolley wrote:
> I am trying to get SIP registrations to work within Asterisk. From my
> snom 200 phone (and on my SJPhone soft client)
I am trying to get SIP registrations to work within Asterisk. From my
snom 200 phone (and on my SJPhone soft client) I can dial via extension.
Example:
To Dial extension 1110 on my asterisk1 server:
I can simply enter SIP:[EMAIL PROTECTED] and the call goes through just
like it should.
As I unde
Not 100% sure here but its probably somthing to do with the fact that MS doesn't
support MD5 and I think * makes use of md5 password hashing during authentication..
Maybe you can try adding auth=plaintext to that account in the sip.conf I know this
option works in the iax.conf..
Later..
> I u
I use Windows Messenger ( I duck as to let the hurled
penguins barely miss my head J ) and I am able to
place and receive calls. So what is the problem you ask??? If I specify a
password in the password field of WM I get a Proxy Authentication Error during
SIP debug and I am not able to c
Remove the "secret=" lines for SIP peers that do not have passwords.
Here is an example of a host that sends us calls but no password:
[foo1]
host=192.168.200.160
type=friend
dtmfmode=inband
That's it; very simple. If you discover that SIP messages seem to be
"ignored" in one direction, se the
You could try placing the password after the username in the URI:
sip:username:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cerrajetto
Sent: Friday, June 20, 2003 11:17 AM
To: Asterisk Users
Subject: [Asterisk-Users] SIP
Hello,
I'm trying to registrate a Nuance Server in Asterisk (using SIP) with no
success.
It seems that Nuance does not send any secret/password (there is no way to
define it!), this is the list of parameters that Nuance provides for
registration:
audio.sip.UserAgentURI=sip:[EMAIL PROTECTED]
audi
> > Instead, I sugest implementing it this way:
> >
> > start-of-uri[:password[:[EMAIL PROTECTED]/contact]
> >
> > with the []'s being used solely to indicate optional fields.
>
> fine with me.
Okies, it's in CVS as of this morning.
> > If we're doing to/from that way, then agreed. In CVS
. I will try
to find out what the registration server is in the morning and that may help
as well.
Bob
From: Masakazu Nakano <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP registration
Date: Thu, 13 Mar 2003 09:21:21 +0900
vers
version is 'Asterisk CVS-03/11/03-09:57:33'
we can regist to wcom in two ways.
first.
register => masakazu:[EMAIL PROTECTED]
* send REGISTER, but no response from wcom.
second. quit * and change the way with number. like this.
register => 9706052:[EMAIL PROTECTED]
and REGISTER again.
in thi
> **ASTERISK SIP PACKET
>
> XXX Need to handle Retransmitting XXX:
> REGISTER sip:166.60.255.41 SIP/2.0
> Via: SIP/2.0/UDP 68.86.118.111:5060;branch=12720924
> From: ;tag=08e71f4b
> To: ;tag=08e71f4b
> Contact:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 113 REGISTER
> User-Agent: As
I have a test SIP account set up with WorldCom and I have been trying to
have Asterisk register to the WorldCom server with no luck. It appears
that the SIP headers are different coming from Asterisk. I have included
a packet capture from a successful login with a Windows Messenger client
for refer
qualify=1000 in sip.conf in the phone config entry
regards
Martin
On Wed, 5 Mar 2003, Mark Spencer wrote:
> > But if I close my sip phone and a call goes through it will still wait
> > the 25 seconds before it goes to voice mail even though my Sip phone is
> > not even on. If I restart Asterisk
> But if I close my sip phone and a call goes through it will still wait
> the 25 seconds before it goes to voice mail even though my Sip phone is
> not even on. If I restart Asterisk and do not register my sip phone it
> will go straight to voice mail after no one picks up on Zap/2. Is there
> a w
Hello,
I have my sip stuff seemingly working fine as well as my zaptel stuff
working great... But I have a problem with sip registration timers (I'm
guessing here).
In my extensions.conf file I have this...
exten => 2244,1,Dial,Zap/2|25
exten => 2244,2,Dial,Sip/brian|25
exten => 2244,3,VoiceMa
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