Displaying 20 results from an estimated 10000 matches similar to: "2 Sip phones behind un-natted Asterisk"
2004 May 24
2
testing asterisk on FXS lines
I am configuring an asterisk server and I want to test the incoming
configuration with my FXS handsets.
I have the FXS lines able to call eachother and they can connect out
the FXO lines.
I changed the context for the FXS lines to "incoming" so that they
would be able to test the setup for incoming calls.
For the incoming context I have:
[incoming]
exten => s,1,Wait(1)
exten
2004 May 14
3
SoftPhone to SoftPhone with No Voice
Hello
I Installed Asterisk on RedHat 9. I am currently try to configure minimum with
two softphone talking to each other over the LAN. I am using X-Lite softphones
from xten.com site. I defined 3 phones in sip.conf and also specifies in
extensions.conf file. I am able to ring other computer but there is no voice
exchange ( i can't hear any think except ring). Here is the portion of sip.conf
2005 Mar 02
3
More NAT questions
> Still trying to get NAT working.
Try adding a canreinvite=no.
Nabeel
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......
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,
2006 Nov 04
1
Pass through
Hi!
I want to tell asterisk to simply pass-through any codecs that my phones
support. I have to use codecs that are not popular and implemented by a
third-party, asterisk has nothing to do with them.
I've made a test with g722 (that asterisk doesn't support), i've set all my
two snom 300 phones to support only g722 and asterisk declined the sip
invitation. That is bad for me. Is it
2003 Nov 06
3
Grandstream problem
Hi,
I installed Asterisk an all works fine exept for Grandstream.
When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok
When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so)
It's the same when I
2006 Dec 13
3
Multi Operator
Hi,
Actually on my setup all outgoing calls are going trhu a SIP unique account
A have a second SIP account with another operator and I would like my setup
to use alternatively each of the two accoutns
Call 1=> Dial SIP/phone1
Call 2=> Dial SIP/phone2
Call 3=> Dial SIP/phone1
<...>
If you have an sample please let me know
2003 Oct 29
1
Host unspecified ??
Dear,
When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field.
Name = phone1 and phone2
Host=unspecified
mask 255.255.255.255
port = 0
status = unmonitored
I can ping the two phone's and get a reply (also from the laptop)
phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14)
2008 Oct 31
1
Monitor group calls (recording calls)
Hello there,
I appreciate any help about this problem that I can't figure out...
I need to record all my calls: this is pretty easy using Monitor() before
the Dial().
eg:
exten =>
425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb)
exten => 425,n,Dial(${PHONE1},10)
Now, I want to create a call group: I mean, I want a number (eg 800) that
makes
2006 Oct 18
2
random one way audio and noise between SIP phones on same LAN
Hi,
sometimes I have one way calls and noise between sip phones connected to
the same LAN so no nat/firewall is involved. I tried with different sip
phone models soft phones and the result is the same. I searched inside
every log file but found nothing. I made different PBX with different
hardware but the result is always the same.
Is there anybody experiencing this terrible problem?
2007 Jun 07
1
DUNDi and reinvites...
I don't know if this is possible, and I can't quite get my head around
how to do it...
If I am using DUNDi for redundancy in a cluster, when Phone1 makes a
call to Phone2, both Asterisk A and B will be in the RTP stream:
+---+ +---+
| A |-----| B |
/+---+ +---+\
/ \
Phone1 Phone2
Is there a way configure re-invites
2006 Jul 26
4
Dropdown with concatenated columns.
What is the best way to create a drop down where the viewable text in a
concatenation of 2 or more columns?
For instance, I hane a lookup table with these columns.
Model FOO
columns: id , name, phone
In my drop select tag, I''d like the user to see:
"name1 phone1"
"name2 phone2"
etc..
I know I can do this using find_by_sql . ..
But, isn''t there a more
2004 Dec 19
1
sip phones in different private networks have one way audio
Hello
I have one phone (phone1) in one network, the other (phone2) in public
network. both can call the other side; phone1 can be heard by phone2, phone2
can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER.
Is NAT still necessary to be set on both phones?
Thank you!
steven
2005 Feb 28
1
call from two sip phones registered in different asterisk server
Hi all
I have registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones.
i configured the extensions.conf file in both the server.
the extensions.conf file on server 192.168.0.9 is
exten=>301,1,Dial(SIP/301@192.168.0.6,20,tr)
exten=>401,1,Dial(SIP/phone1,20,tr)
301 is the extension number for phone 2 in asterisk server
2006 Jan 20
2
How to have a phone ring another extension as soon as off-hook?
I am seeking to implement the following behavor:
When a headset on phone1 is picked up, phone2 rings right away, without any
need to dial numbers on phone1. Is this possible to implement?
ScriptHead
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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
2006 Jan 14
3
Reducing echo on FXS port
Hello everybody,
I am sorry to bring this up again if this kind of echo issue has ever discussed.
Phone2 in below call path experiences quite annoying echo:
Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2
It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with
2003 Sep 17
1
A WORKING EXAMPLE
Hello!
I've looked at the reference examples they are all for SIP. I have two
X100p and a TDM400P. Can someone send me a working example so I can
receive calls and make them. I'm stuck at first base. [I'm using standard
phones - not SIP] Help please!
Thanks,
Bill Flood
2018 Aug 27
2
feeling n00b again
Retrying, falling of the list some how :-(
-------- Original Message --------
Subject: feeling n00b again
Date: 2018-08-20 09:51
From: asterisk at a-domani.nl
To: asterisk-users at lists.digium.com
Hi all,
Long time ago, I followed a Asterisk training, and both at work and at
home, was able to deploy Asterisk,
make all sorts of internal call (hard/soft voip-phones,
incoming/outgoing,