Displaying 20 results from an estimated 40000 matches similar to: "Feature ver 1/2 Questions"
2005 May 26
1
How do I diagnose the problem in this Asterisk test session with FWD?
=============
SJphone Log
============
Outgoing SIP session
Respondent: (sip:8612@192.168.2.2)
Remote client:
Started: May 26 16:33
Accepted: no
Ended: May 26 16:34
End reason: Call rejected: 503 Service Unavailable
===============
Asterisk Debug
================
Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r")
in new stack
--
2003 Jul 03
3
Using switch =>
hello,
I have a test setup with 2 asterisk servers, each having a one snom 100
via sip using it. I`m experimenting on how trunking between them would
work. I have them setup for RSA authentication which I plan to use in
the future.
So I`ve setup the keys and servers seem authenticate to each other. One
is named phila and other hurricane.
Here is what I see on phila:
-- Registered
2004 Apr 23
1
IAXPHONE failures in calls to Cisco Phones
2004 Dec 22
1
Problem ringing simultaneous channels
Russell,
What kind of zap cards do you have??
If T1, is it PRI or RBS
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Russell
Horn
Sent: Wednesday, December 22, 2004 6:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Problem ringing simultaneous channels
I have a
2004 Jan 31
0
Dial app does not indicate ringing to calling party
I hope somebody has seen this before...
I'm trying to use a Dial command on a inbound call to ring multiple
destinations. The calls come in to me from the provider on IAX2, and one
of the destinations I try to ring is a IAX2 to call to my cell phone.
When I add the IAX2 destination into the Dial command, the setup I am trying
to achieve works (i.e. my Zap, SIP, and cell phone all ring) but
2005 May 18
2
Call forwarding...
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based -
2005 May 30
0
IAX2 to H323
Hi all,
I'm using following software and equipment and I have very strange behavior:
Asterisk CVS-NHEAD-05/30/05-16:42:41
H323 gatekeeper - GnuGK 2.2.2
IAX2 client - Firefly 1.9.8 build 3945
H323 client - SJPhone Build 1.50.271d
H323 gateway - Welltech Wellgate 3504A
When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected.
When I dial from SJPhone (H323) ->
2003 Sep 16
1
Using IAXTEL with RSA authentication. MD5 works, RSA not. [2]
[ Sorry, I incorrectly copied some Reference headers into this post
and tacked it onto the wrong thread. -Steve ]
So far, I have been able to receive incoming iaxtel calls via my
assigned 1-700-xxx-xxxx number, but only when using md5
authentication in iax.conf:
[iaxtel]
type=user ; Incoming calls only
context=incoming
auth=md5
secret=<mysecret> ; Required for
2005 Mar 15
1
Asterisk retains DTMF Control Even when an External IVR System is dialed
I am using Asterisk 1.06 Stable.
When I dial my Mobile Number to check Voice Mail or my Bank Account
Phone Access Number, the IVR System on the other end asks me to enter
*2378 to transfer to an attendant.
But When I press *2378, Asterisk tells me that it cannot transfer the
calls and gives an error on CLI saying Extension '' does not exist in
the dial plan.
What is the trick to make
2003 Jul 26
0
app_voicemail2 became a bit silent, lately...
Hi,
after cvs upgrading my * installation yesterday, the prompts in both
VoiceMail2 and VoiceMailMain2 have become silent. All I get is the initial
"blip" followed by the Voice taking breath and being cut off before she
has a chance to say "Comedian Mail".
All other prompts (ie the Playback application) seem to work fine.
I can still login to VoiceMailMain2, however, each
2006 Jun 02
0
OT recommend an IAX phone or IAX softphone+USB handset?
Hi Yall,
I would love to put a very compact phone on my wife's desk at work...
Ideally this would be a very small IAX phone with 2 RJ-45's so I could
drop it in without much notice and only have to beg for 1 port from the
sysadmin.
I have looked around and I don't see such an item in existence? What I
would love is a "home" phone looking handset (wired is fine), which
2005 Aug 27
2
Problems with registration
My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP
phone.
Here is my sip.conf file:
;
; SIP Configuration
;
[general]
context=default ; Default context for incoming calls
port=5060 ;added
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ;
2007 Jul 17
0
ASA-2007-015: Remote Crash Vulnerability in IAX2 channel driver
Asterisk Project Security Advisory - ASA-2007-015
+------------------------------------------------------------------------+
| Product | Asterisk |
|--------------------+---------------------------------------------------|
| Summary | Remote Crash Vulnerability in IAX2 channel driver |
2007 Jul 17
0
ASA-2007-015: Remote Crash Vulnerability in IAX2 channel driver
Asterisk Project Security Advisory - ASA-2007-015
+------------------------------------------------------------------------+
| Product | Asterisk |
|--------------------+---------------------------------------------------|
| Summary | Remote Crash Vulnerability in IAX2 channel driver |
2003 Dec 18
6
G729 question
I am thinking about using the G729 codecs on my endpoint devices and
purchasing some G729 licenses for Asterisk but I have several questions:
1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I?
2. If I have G729A on one end and G729B on the other, are they compatible?
I have looked all over the place for question 2, but without buying the
ITU docs
I cannot seem to find this
2003 Jul 07
0
Problems with Hangup Detection in VoiceMail2.
Hi.
Has anyone experienced hangup detection problems with the VoiceMail2 app?
I have a console phone on the FXS port. When I call a SIP phone, and get
its voicemail greeting, I can enter the VoiceMail2 app, leave a message,
and then hit # to stop message recording.
Recording does stop, but the channel stays up inside the VoiceMail2 app
(as shown by a "show channels" command) for about
2003 Jul 09
0
SUMMARY: Problems with Hangup Detection in VoiceMail2.
Many thanks to Martin Pycko and Mark Spencer.
Mark's suggestion below was correct:
"Maybe it's stuck trying to send the e-mail notification. If you take
the e-mail address out of /etc/asterisk/voicemail.conf does that speed
it up?"
Indeed it did!
The problem turned out to be a 60second delay while invoking mail,
caused by a mis-configuration of my hostname and
2005 Feb 16
0
More jitter buffer questions
I've been trying to resolve some quality issues and I was hoping
someone might be able to provide some insight.
To give you an idea the calls are coming in via a SIP DID and sent out
via an IAX2 connection. Latency to both the SIP equipment and IAX
equipment are around 80ms with 0 packet loss accoridng to ping tests.
The server is located in a data centre so bandwidth is not an issue.
Most
2005 Jan 12
0
IAX2 dropped calls: need debug suggestions
Hi,
I'm trying to determine why IAX2 calls are getting dropped after a 4-24 hours of continuous connect time. My project requires that calls stay up for days at a time. When I turn on IAX2 debugging, I see "max retries exceeded" for control frames just before the connection is dropped.
My test setup is:
Zap Phone => Local Asterisk Server/IAX2/GSM => NAT => Internet =>
2006 Mar 28
1
Redirect problem/bug/feature
I have a major problem with SIP redirects, or maybe just do not understand
how they are supposed to work. We are using Cisco 7940s and 7960s with SIP
version 6.3. Asterisk 1.2.5.
A call come in to extension 944 over the IAX trunk. Extension 944 has
forward all to extension 904 set on the phone. According to the dialplan.
extension 904 should ring for 90 seconds, then ring another extension, and