Displaying 20 results from an estimated 1100 matches similar to: ""username mismatch, have <x>, digest has <y>""
2011 Jan 10
3
sendrpid does not work!
Hello,
I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work!
I placed this in my peer: (sip.conf)
sendrpid=yes
trustrpid=yes
or
sendrpid=yes
trustrpid=no
(and restarted Asterisk)
and the line "Remote-Party-ID" does not appear in my sip debug!
Please help me,
Mickael.
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2010 Feb 20
1
Fax, T38 and NAT
Gentlemen,
I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk.
0851711201 and 0851711290 is on our WAN, no NAT.
0197673581 is outside our WAN and needs to be NAT'ed.
Sending a fax from 0851711201 to 0851711290, no problem, switches to T38
and fax goes through.
Sending a from 0197673581 to 0851711201, no problem as long as i dont
enable T38 on 0197673581.
But, if i enable T38
2008 Feb 01
0
Bypassing a Auth on Invite or Forbiden?
Hello,
I have 2 asterisk servers that are not working well together. One is
acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX
devices. And the other is acting like my sip gateway (PBX02) to
various providers. They are both on a private network and should be
trusting each others IP 100%. But the PBX02 challenges PBX01's
requests all the time even though
2015 Jun 26
0
Asterisk dialplan best practices syntax
On Fri, 26 Jun 2015, Ludovic Gasc wrote:
> 1. What's the "official" notation of each line: "=>" or "=" ? In the
> wiki of Asterisk, I see very often "=>", however, what's the reason for
> both syntaxes authorized ? Historical ?
I'm not 'official,' but I have a strong preference for just '=.' Using
2010 May 06
2
problem with trustrpid
Hi everyone,
I am trying to figure out the behavior of trustrpid
Basically its not behaving the way I expected it to or maybe I am
missing a configuration option or something else.
When a call from a phone is sent to the * box it has the following sip
headers:
From: "From Phone" <sip:1001 at 10.0.0.29>;tag=4bf4bb4e11e92476.
Remote-Party-ID: "Cloutier"
2013 Feb 15
6
Cisco 7942 Connected line ID
Hi,
Is it working for anyone?
I have tried with
trustrpid=yes
sendrpid=yes/pai
but can not get it working, Asterisk cli shows prevented message like this.
Connected line update to SIP/1231-00000200 prevented
Regards,
Zohair Raza
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2009 Jul 26
3
Not getting inbound CallerID name on Asterisk
We have an inbound PRI connected to our Cisco 3825 router which is then
passing the calls to Asterisk as SIP calls. We're getting the CallerID
number but not the CallerID name. We are seeing the name in the RPID field
with a SIP trace on the Asterisk box but don't understand why it's not
registering as the CallerID name.
Here is a link to pastebin with the Sip trace. In it you
2015 Jun 28
2
Asterisk dialplan best practices syntax
2015-06-26 17:11 GMT+02:00 Steve Edwards <asterisk.org at sedwards.com>:
> On Fri, 26 Jun 2015, Ludovic Gasc wrote:
>
> 1. What's the "official" notation of each line: "=>" or "=" ? In the wiki
>> of Asterisk, I see very often "=>", however, what's the reason for both
>> syntaxes authorized ? Historical ?
>>
2015 May 28
3
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb:
> Ahh. Seen that before! That suggests to me that you don't have your
> sip.conf records setup right.
>
> What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register =>
2015 May 28
0
Peer is UNREACHABLE
I think your phone may be trying to register with the username '1234',
while your sip configuration is expecting 'luca'. Can you try changing
your phone registration credentials to use 'luca'? Can you give us a sip
transcript when you try to place a call from it?
On 15-05-28 05:09 PM, Luca Bertoncello wrote:
> Darryl Moore <darryl at moores.ca> schrieb:
>
2014 Apr 16
1
Connecting 2 asterisks, one with PJSIP and other SIP returning 401
It's my first post here, so I'll cut to the chase
I have 2 Asterisk servers and want to connect them using sip on one and
pjsip on the other one. One is running at home and another at a VPS. The
first one will be the client (with dynamic ip) and the 2nd the server.
The client uses sip and the server pjsip.
This is the client's sip.conf
[general]
context = default
allowguest = no
2012 Aug 17
0
Trouble with call pickup using RPID with Cisco
I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to
asterisk 1.8.15.0.
imagining in extensions.conf:
exten => 1,1,Dial(SIP/121)
exten => 2,1,Dial(SIP/121&SIP/122)
When a caller dials extension 2 /and/ I have
trustrpid=yes
generaterpid=yes
sendrpid=yes
in sip.conf and I use the pickup exten, the caller is disconnected.
see:
2014 Aug 13
0
SRTP only from asterisk to extention possible
Hello,
trying to implement srtp with already working tls i somehow stuck with
srtp. If the extension has successfully registered a call from asterisk
to that extension works fine. But the other way round nothing happens.
[Aug 13 14:54:16] WARNING[31053]: chan_sip.c:3906 __sip_xmit: sip_xmit
of 0x7fc8880467e0 (len 609) to 123.456.789:36785 returned -2: Success
[Aug 13 14:54:20] NOTICE[31053]:
2014 Dec 10
0
PJSIP configuration question
On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp <dan at amtelco.com> wrote:
> Not sure why, but Vitelity changed the settings to IP based authentication
> on me. Here's the new sip.conf settings they sent me.
>
> type=friend
> dtmfmode=auto
> host=64.2.142.93
> allow=all
> nat=yes
> canreinvite=no
> trustrpid=yes
> sendrpid=yes
>
> When I use these
2015 Jun 04
0
Differences between blind or attended transfer and impact on CDR entries
Hello,
Sorry for a bit of a newbie post but we all had to start somewhere right ..
I'm wondering if someone can briefly explain the difference between blind and attended transfers and why they would generate two very different CDR entries.? From my own research, it seems that transfers are both ultimately a SIP REFER and thus seeing two different CDR entries just confuses me further.
2012 Aug 02
1
Originate call from cli does not work for SIP line...
I have a SIP line that is working fine when I make calls from IP
phones. I can send and receive calls. The problem is that if I try to
dial from the CLI using the originate command or use an AMI connection
to originate a call I get the following error:
originate SIP/protel-out/0445540881644 application playback tt-monkeys
WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received
2008 Dec 29
0
SIP host=dynamic help needed for CCME
Hi,
I'm trying to get a remote Cisco Call Manager Express (CME) system behind
a dynamic IP address routing both inbound and outbound calls via SIP to my
local asterisk server. I've got a local CME system working fine on the LAN,
where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't
figure out how to get it working with host=dynamic, even locally on a test
2010 Sep 12
1
username mismatch with 1.6.2.11
Hello,
everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I
get the following :
[Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username
mismatch, have <329909006666>, digest has <3291119600>
[Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite:
Failed to authenticate device "0473990000"
<sip:0473990000 at
2011 Apr 11
1
Asterisk codec negotiation and canreinvite=no
Hi all,
I realise that asterisk's codec negotiation has been discussed in
the past multiple times. What I haven't been able to understand is
how asterisk decides which video codecs to advertise to the other
end when canreinvite=no in sip.conf and the initial caller
doesn't support video.
My tests are quite simple, I use an asterisk with 4 peers all on the
same LAN. My sip.conf
2014 May 12
1
new install: no re-invite and unwanted transcoding
I am unable to get re-invite to work on a new system. Also, unwanted
transcoding is occurring on PSTN calls.
The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will
eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2,
CentOS 5.8) currently in production. Both systems are on VPS with public
IP addresses. Goals for the new system include: HD (g722) connections on