Hi list!
Today I tried to change the NAT-configuration on my Firewall to use
another port for SIP.
I configured it so:
/sbin/iptables -t nat -A PREROUTING -p udp -m udp --dport 1:10100
-j DNAT --to-destination 192.168.20.120
/sbin/iptables -t nat -A PREROUTING -p udp -m udp --dport my
Hi,
I’ve installed Asterisk for use as a SIP server. I can call people, but one
strange thing happens: if I call someone with a SIP account outside my server
(for example, sip:enum-echo-t...@sip.nemox.net) everything is fine, if I call
any Asterisk extension it also works, but the call gets
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky
Sent: Friday, March 13, 2009 9:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] No reply to our critical packet
Hi,
I've
Hi,
thanks for the quick reply.
1. Do you have the incoming 1-2 holes in your firewall so the
remote server can get it's reply back to *?
This was what I checked first. Both firewalls let everything through.
2. If #1 is ok, try putting an Answer command in front of your Dial
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky
Sent: Friday, March 13, 2009 10:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No reply to our
On Friday, 13.03.2009 17:50:57 Danny Nicholas wrote:
Next Step would be to check/update the firmware on your phones or router.
I don’t think the router is to blame, it does deliver all the packets. And
there are no hardware phones, only numerous software SIP clients.
Which (GNU/Linux)
I have the same situation. My scenario is weird:
I have a DID with IPkall that points to my asterisk server, and I have it
play a message with Playback() after about 20 seconds call drops and give
me the same message you get: no reply to our critical packet
BUT
I have a DID from Vitelity, and
--- On Fri, 3/13/09, Pascal Bruno tipas...@gmail.com wrote:
I have the same situation. My scenario is weird:
Well, I've experienced the same symptoms but in a whole different context. It's
happening in my LAN (no firewalls, no NAT) and only with specific IP phones +
early dial +
13, 2009 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No reply to our critical packet
Hi,
thanks for the quick reply.
1. Do you have the incoming 1-2 holes in your firewall so the
remote server can get it's reply back
Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] No reply to our critical packet
I had this issue with SIP Phones (Cisco 7961) to local voicemail; the issue
was resolved by adding a
Ringing() followed by
Wait(1)
before the VoicemailMain() in the dial plan
: Friday, March 13, 2009 2:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] No reply to our critical packet
Not a better hack but perhaps more palatable to the listener
Playback(please-wait)
Wait(1)
-Original Message-
From: asterisk-users
13, 2009 1:42 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] No reply to our critical packet
My best guess at the root cause of the problem after looking at the packet
capture was that the phone was not happy seeing the call connected before
any
Ringing() followed by
Wait(1)
I made it
exten = echo,1,Ringing()
exten = echo,2,Wait(1)
exten = echo,3,Playback(abandon-all-hope)
exten = echo,4,Echo()
to no avail.
This looks like a client issue, though all of my clients fail. Which clients
are the most standards conforming?
Also, maybe
] No Reply to Our Critical Packet SIP Calls
Dropped in Voicemail
Hi Lincoln,
The fact that you can hear and respond to the voice mail (even if its
for the first 20 seconds), means that your phone has received the OK
message properly. The problem is the missing ACK after receiving OK.
When asterisk did
Hi Lincoln,
Asterisk was expecting ACK after sending the 200 OK message. After
repeated attempts at sending the 200 OK message and not receiving ACK,
it terminated the call. Are you able to do a packet capture on the phone
end? Mostly likely the phone is sending the ACK, but its either sent to
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steve J. Douglas
Sent: Tuesday, February 03, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No Reply
Lincoln King-Cliby wrote:
-Original Message-
Then starting at packet 3217 there are a series 6 of ICMP
Destination unreachable (Port Unreachable) messages from the
Asterisk server to the phone, with an RTP packet from the Phone
to the Asterisk server before each Destination
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mark Wiater
Sent: Tuesday, February 03, 2009 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No Reply
I'm not familiar with packets specific to Asterisk, but do have some
familiarity with general Ethernet traffic. The Host unreachable messages you
are getting is from the protocol stack in the Linux computer, and generally
means the traffic is being sent to a port that is not open--i.e. no
:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steve J. Douglas
Sent: Tuesday, February 03, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No Reply to Our
Hi All,
I posted this a couple weeks ago with no response, I'm hoping that someone will
see it this time around and be so kind as to offer advice for resolving this
issue (or point me in the direction of a better place to ask)
Some (but not all) calls on one of our Asterisk boxes are being
On Mon, Feb 2, 2009 at 12:39 PM, Lincoln King-Cliby
linc...@controlworks.com wrote:
Hi All,
I posted this a couple weeks ago with no response, I'm hoping that someone
will see it this time around and be so kind as to offer advice for resolving
this issue (or point me in the direction of a
Thanks to everyone who has replied so far; to answer a few of the follow up
questions that have been posed:
Dave -
Which firmware load? We had all kinds of trouble with 8.4.x, after being
stable for a few months on 8.3.x. Going back to 8.3.x made all of the
weirdness disappear. While we're
Sounds like there's some sort of firewall in place or something else
that is preventing an ACK from being received in response to the 200 OK.
Notice that the 200 OK keeps being retransmitted.
Lincoln King-Cliby wrote:
Hi All,
I posted this a couple weeks ago with no response, I'm hoping
Which firmware load? We had all kinds of trouble with 8.4.x, after being stable
for a few months on 8.3.x. Going back to 8.3.x made all of the weirdness
disappear. While we're on the cisco note, I have script to remotely reboot the
SIP firmware load Ciscos and to provision the phones based on
-- Executing [EMAIL PROTECTED]:2]
VoiceMailMain(SIP/17865221569-b6b03f60, 3523782778|s) in new stack
-- SIP/17865221569-b6b03f60 Playing 'vm-youhave' (language 'en')
app5*CLI
--- SIP read from 74.170.252.213:5060 ---
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail
After about 30 seconds the call drops with these messagess:
[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950
This message is usually caused by Asterisk not receiving an ACK after
about 30 seconds of attempts. There are countless misconfigured UAs and
proxies out there that don't handle ACK well, so it would be nice to be
able to turn this 'feature' off. What's annoying is that the explanation
has
The odd thing is on this particular phone it only happens when you
call voicemail.
It is certainly a bug in Asterisk, not the UA. Asterisk is trying to
send to 192.168.1.x which obviously is not possible. Something in the
NAT support is not working right.
On Mon, Oct 6, 2008 at 3:06 PM, SIP
On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
The odd thing is on this particular phone it only happens when you
call voicemail.
It is certainly a bug in Asterisk, not the UA. Asterisk is trying to
send to 192.168.1.x which obviously is not possible. Something in
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail
After about 30 seconds the call drops with these messagess:
[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950
Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 20
seconds.
This only happens because Im using Asterisk2Billing's AGI (without
A2Billing it doesnt hang up).
does someone knows whats the problem??
Here is my Asterisk debug:
(xxx.xxx.xxx.xxx - the phone's IP)
Apr
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