Thanks. I will give that a try. Do you know if removing that line will
affect
other phones I might have?
If so, maybe I am better off getting someone else's phone.
ACT's support seems a bit problematic. They responded to my first email right
away,
but never, so far, to my second.
joe, at j4computers wrote:
Thanks. I will give that a try. Do you know if removing that line will affect
other phones I might have?
If so, maybe I am better off getting someone else's phone.
ACT's support seems a bit problematic. They responded to my first email right away,
but
I saw this problem before... to solve that, I needed to hack asterisk
to remove a header SIP field.
Check your ACT phone log, and you can figure out which filed is that.
Then, comment that filed from your chan_sip.c and recompile asterisk..
and that's it.. it only happens with ACT phones.
I
I have a setup with a polycom 601 and an act p160s. All on local segment, no
NAT.
Can call the act p160s, from the polycom, rings, connects, and a conversation
can take place. The reverse is not true, Dialing from the act to the polycom
does not work. SIP debug shows, at the end, Incoming
Using a SIP connection with a CLEC, the downstream (received) audio is
perfect when the mute button is activated on the phone. However, when
there is upstream audio (i.e., talking or even breathing into the
microphone), the downstream audio is cut off. It's kinda like having a
half-duplex
On Mar 23, 2006, at 8:00 AM, Michael Welter wrote:
Using a SIP connection with a CLEC, the downstream (received) audio is
perfect when the mute button is activated on the phone. However, when
there is upstream audio (i.e., talking or even breathing into the
microphone), the downstream audio
CLEC = Competitive Local Exchange Carrier
ILEC = Incumbent Local Exchange Carrier (aka the telephone company)
On Mar 23, 2006, at 8:52 AM, Martin Joseph wrote:
On Mar 23, 2006, at 8:00 AM, Michael Welter wrote:
Using a SIP connection with a CLEC, the downstream (received)
audio is perfect
On Mar 3, 2006, at 1:46 PM, Gavin Adams wrote:
Hi All,
I'm stumped on a weird problem. I have an * server working fine for
local
SIP phones and IAX2 connections. We just provisioned a second Ethernet
port to attach to a local SIP provider.
PSTN calls incoming work fine:
PSTN - SIP
Hi All,
I'm stumped on a weird problem. I have an * server working fine for local
SIP phones and IAX2 connections. We just provisioned a second Ethernet
port to attach to a local SIP provider.
PSTN calls incoming work fine:
PSTN - SIP Provider - SIP - *
but outgoing calls are not. Call setup
Try turning off iptables (firewall)
service.
MD
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abhimanyu
RapriaSent: Friday, February 17, 2006 2:19 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP Problem
Fedora Core 4 and Asterisk 1.2.4
Fedora:Linux
Fedora:Linux abcde 2.6.11-1.1369_FC4 #1 Thu Jun 2 22:55:56 EDT 2005 i686 i686 i386 GNU/LinuxAsterisk: 1.2.4SIP Problem1. Asterisk sends SIP messages to Softphone.
2. Softphone receives SIP messages and replys back.3. Asterisk doesn't receive these replies and keeps on sending.Asterisk:Reliably
sip debug
rtp debug
enable all the log levels in console in logger.conf
regards
On 1/20/06, RumaTech [EMAIL PROTECTED] wrote:
Hi, all
I am trying to call to particular destination via SIPNET (one of the VoIP
providers).
I can succesfully dial and I can hear waiting tone, however nothing
Hi, all
I am trying to call to particular destination via SIPNET (one of the VoIP
providers).
I can succesfully dial and I can hear waiting tone, however nothing happens
beyond it.
Here is what asterisk shows:
Executing Dial(SIP/phone2-fa85, SIP/sipnet/84959951017) in new stack
--
On Upgrading do 1.2 I do have an PRoblem with my VOIP-Provider.
Making outbound calls does result in Error 400 - exept if I do call my
own phonenumber.
I dind find the solution th this problem in current CVS source,
chan_sip.c has to be updated.
Elmar
Hi,
On Upgrading do 1.2 I do have an PRoblem with my VOIP-Provider.
Making outbound calls does result in Error 400 - exept if I do call
my own phonenumber.
Usimg my SNOM190 directly or reverting to 1.0.9 does resolve the
problem immediately.
What has to be changed in SIP config to move
i have configured a sip phone to make calls through a
sip server but when i make call through the sip phone
to the sip server every thing goes well and the call
is done perfectly but on sip server it gives me these
messages(i have 2 pc with different ips one with a sip
phone and the another with
hi,
I am not registered my SIP Phone with Asterisk i spend almost one day
but find no luck.I know very well this is not kind a problem discussed
in this group but i try my best and all in vein so finally i am here
hoping you ppl helping me out.I discussed this problem in
asterisk's-users
and then tries to connect the incoming sip call
to 101, hence the loop :)
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Adnan Ahmed
Verzonden: maandag 22 november 2004 21:34
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] SIP
I am a newbie to Asterisk (though not to SIP) . I am trying to setup a
pure SIP environment for some testing. Here is my SIP.CONF file:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
[247417]
type=friend
host=dynamic
dtmfmode=inband
secret=xyz123
context=default
And my EXTENSION.CONF
When calling from Zap (E100P) to ATA186 (SIP) *
hanged up...
below is 'show channels' command
output:
Channel (Context Extension Pri
) State Appl.
Data SIP/565-adc3
(voip
1 ) Up
AppDial (Outgoing
Line) Zap/31-1
(incoming 565
2 ) Ringing
Dial
SIP/565|60|r
Hi, I'm testing Nikotel with Asterisk.
Sound quality is Ok, but I can´t manage to have a call longer then 1 minute
After 1 minute or so, my * exchanges some SIP messages with Nikotel and the
call ends with maximum retries error.
Debugging the SIP messages, I see 2 IP´s in the VIA header, the
Turn sip debug on and forward the logs.
A 481 means that a dialog was not correctly established.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista
Sent: Thursday, February 12, 2004 6:28 PM
To: Asterisk User List
Subject: [Asterisk-Users] Sip
I have one of my IpDialog phones giving this error about once an hour.
On the Asterisk server CLI I get this message.
Got SIP response 481 Call Leg/Transaction Does Not Exist back from
204.241.XXX.XXX
If I go to the phone and dial out it works and I no longer get the
message. Also if I check
HI List,
I have two SIP phones. one is 6002, w:
which is behind a NAT, and another is 5009 which has public IP .
When a call between 6002 to 5009, the 6002 cannot hear any from 5009, and
5009 did hear from 6002.
And in the sip debug, I see following message
Sip read:
SIP/2.0 481
I have two SIP phones. one is 6002, w:
which is behind a NAT, and another is 5009 which has public IP .
When a call between 6002 to 5009, the 6002 cannot hear any from 5009, and
5009 did hear from 6002.
[6002]
type=friend
host=dynamic
nat=1
qualify=yes
[5009]
type=friend
, 2003 9:33 AM
Subject: [Asterisk-Users] SIP problem with asterisk
HI List,
I have two SIP phones. one is 6002, w:
which is behind a NAT, and another is 5009 which has public IP .
When a call between 6002 to 5009, the 6002 cannot hear any from 5009, and
5009 did hear from 6002
I am having a problem when a SIP registration fails. I get the following
messages in the log:
Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874
(sip_reg_timeout): Registration for 'user@[EMAIL PROTECTED]'
timed out, trying again
Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c,
regarding my previous post about SIP outgoing calls
dropping with an error 481 ..
this is my output from a SIP debug.
the call dropped occurs at the end.
Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my
control.
help :) please!!
Dave
Signal=0
Duration=250
(no NAT)
28 matches
Mail list logo
