Displaying 20 results from an estimated 10000 matches similar to: "Two sip extensions"
2005 Mar 04
2
Broadvoice + incoming call works only for ~2 minutes
Hi, all.
The asterisk setup is working fine, receiving calls via broadvoice "initially". ?
When call comes in via broadvoice number, asterisk picks it up and routes
correctly, as long as the call came in within ~2 min from the previous one.
In other words, as long as a call comes in within ~2 min since the previous one,
asterisk will answer the call. However, if the call comes in
2004 May 09
2
Help with initial setup
Hi,
I've have followed through the help docs in trying to get an initial setup
going with two phones and the asterisk server. Firstly, all I'm trying to
do is get the two phones actually talking to one another VIA asterisk..
I've added this to sip.conf:
[phone1]
type=friend
host=dynamic
defaultip=192.168.1.106
;username=blah
;secret=blah
dtmfmode=rfc2833 ; Choices are inband,
2019 Jul 18
3
Two sip extensions
It looks like moving both to the general section got it working.
Never new that was a requirement. :)
Thanks,
Jerry
>
>
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2010 Apr 16
2
Testing a sip call through Asterisk?
I'm looking to find a test tool that will register with our Asterisk
(Trixbox) server here at work and place an outgoing call via our main SIP
trunk (BroadVoice) to confirm that things are working. I've looked around
but I can't seem to find any tools that will do what I'm looking for.
I can't just monitor the status of the trunk inside Asterisk, as this is the
normal status:
2005 Mar 02
1
Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
> Hmmm... I have this aweful feeling that I'm choosing the
> exact wrong time to ask a "newbie question" :) Oh well, here
> it goes.
>
> The quick question is : "How do I dial an extension?"
> (answer is probably - "you don't" in which case:) "How do I
> dial my asterisk box?" - I have no outside line, I just want
>
2007 Feb 03
3
error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
The following strange conditions is happening while I try to dial a
SIP user from another SIp user.
SIP to Zap dialing is fine, as all 4 users can call PSTN.
I'm using Asterisk SVN-branch-1.2-r51359M
Example: extension 3210 calls extension 3213. They are all registered properly:
chrom01*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
3213/3213
2004 Nov 03
5
FireFly Problems
How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and
Info. RFC2833 is the default, so I left it that way. I also unchecked all
the codecs except g711ulaw to force that codecs usage. However, when I go to
place a call, I get this:
Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF
is not supported on codec G.711 u-law. Use RFC2833
Nov 3 13:18:44
2003 Aug 19
1
Problem with * server and FWD
I have a small HUGE problem with *.
I have installed * but I have 2 problems.
1 - Making call to FWD.
2 - Receiving call from FWD
More info of the problem at the end.
Here is the sip.conf file.
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = sip ;default Default for incoming calls
register =>
2005 Mar 08
13
Broadvoice latest changes and still not working
I have added the three lines to the sip.conf file based on the latest
changes
from broadvoice. I can receive incoming calls but cannot place any
outgoing calls.
The error I get is:
*CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
-- Attempting call on SIP/Broadvoice/5068012 for application
Playback(demo-congrats) (Retry 1)
Mar 8 08:35:21 NOTICE[29290]:
2005 May 17
4
multiple sip accounts from same sip registrar
Dear all,
I have an asterisk sip issue which I don't believe is unique.
I use a registrar (sipgate.co.uk) where I have 3 different accounts.
These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users.
By using just one of these accounts I can set asterisk up to send and receive calls no problem.
However, when I start to introduce an
2006 Apr 07
1
wellgate registration 3802
I have a new wellgate 3802 unit. I have not gotten it to
register with asterisk 1.2.6.
My proxy setting is the correct IP in the 3802.
My security config is 1001/1001 and 1002/1002 on the wellgate (simple at
this time).
My sip.conf has:
[wellgate3802L1]
type=friend
dtmfmode=inband
username=1001
secret=1001
host=dynamic
canreinvite=yes
nat=no
context=wellgate
[wellgate3802L2]
type=friend
2012 Jul 12
1
Asterisk with OpenBTS and mobile phone
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile
phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
OpenBTS works and I can send sms from the OpenBTS server to the
mobile phone. What I also need is a call between Asterisk and OpenBTS.
I have also two soft phones which works with Asterisk. And also
2007 Aug 29
2
sip authorization problem
Hi,
I am trying to setup a simple home voip service w/ *
I have compiled and installed the svn source
as a first step I am trying to configure SIP for inside my network.
I have a handful of softphones and a few hardphones that I want to all be
able to call each other
I have configured users.conf with a single softphone(kphone) and have tried
calling itself (ext 6000) and the demo
from the
2005 Jan 26
4
A working BroadVoice config example
I finally got my incoming and outgoing to work on Broadvoice with a
configuration file that is no where close to the one given by them.
Here it Is (sip.conf). For others who have a working config could u please
share so that I could compare. Thank You
[broadvoice]
type=friend
username=[number]
fromuser=[number]
secret=[password]
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
2003 Mar 09
6
DTMF detection on SIP provider ?
Hi..
I just wondering why DTMF are not recognized by aterisk on incoming calls
from my SIP provider ...
ANy suggesteions ?`
/Mike
2010 Jun 15
4
can't seem to register, status unmonitored
Hi everybody,
I am trying to register my softphone(twinkle) on an asterisk server.
Everything seems to be fine.
Here is the output on show registrations in twinkle:
Tue 18:57:51
nikhil: you have the following registrations
<sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013
208 is ip of the asterisk server.
on the server on doing 'sip show peers' , it
2004 Jan 25
2
Incoming SIP matching
Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to
have dtmfmode=rfc2833. However, incoming FWD calls from the dialup
access numbers (such as libretel) need to have dtmfmode=inband. To
solve this problem, I created a second FWD account and configured
sip.conf as follows, in order to match the incoming number to the proper
dtmfmode:
[fwd-rfc]
type=friend
secret=*****
2004 Nov 29
1
NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap'
Hi Asterisk-ians!
Need all of your help. I am stuck with this issue for last few days. I have
one X100P installed in my system. My Asterisk is registered with another
Asterisk Server/SIP provider as client and the call is successfully received
by my Asterisk server (named as starwars).
Now, the extentions.conf file states, the incoming INTO * should go out to
fxo as below:
exten =>
2004 Jun 11
11
Broadvoice and DTMF
I understand there has been some issues sending DTMF tone through
Broadvoice. Can some provide me with symptoms?
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004
2005 Jan 05
4
Broadvoice / * re-register issues
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =>
##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234
[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com