ers@lists.digium.com
Sent: Saturday, May 05, 2007 4:08 PM
Subject: [asterisk-users] SIP registration problem
I've reposted with a more meaningful subject - hopefully someone will
replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP.
The registration succeeds, and is confir
I've reposted with a more meaningful subject - hopefully someone will
replyWe 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 attem
Thanks Andrew,
I see the resolved bug report. I'll get the patch fix.
Sorry for the unnecessary mail.
-Tom
On 1/20/07, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
http://www.google.com/search?q=423+%22Interval+Too+Brief%22&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:of
http://www.google.com/search?q=423+%22Interval+Too+Brief%22&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official
Hint: Who develops Asterisk?
On 1/20/07, Thomas Madler <[EMAIL PROTECTED]> wrote:
Hi,
I'm trying to get my * server connected to a softswitch through an SBC. I
Hi,
I'm trying to get my * server connected to a softswitch through an SBC. I
get the following error when * trys to register.
Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx
Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:--
Registration for '[EMAIL PROTEC
I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone.
I can't find/replicate when exactly its happends but sometimes after server restart or phone restart one of the phone can't register and I get this in the server:
Transmitting (no NAT) to 10.1.1.152:5060:SIP/
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):
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=
>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
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
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