Displaying 20 results from an estimated 800 matches similar to: "FW: problems with asterisk and the IAX protocol"
2004 Aug 05
0
problems with asterisk and the IAX protocol
Hello group,
I wanted to try out the asterisk iax protocol between two asterisk
machines but have several problems with it.
My scenario looks like follows. I am using asterisk 0.9.0 on both machines.
SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2
Both SER and asterisk run on a machine with a public IP address. When
the telephone on one side makes a call the telephone
2004 Jun 07
2
AGI + g729A
Hello....
I have the follow situatuion:
< ISDN >
|
|
V
E100P
|----------------| IAX2 / g729A |----------------| T100P
| Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - -
-> |--------------|
| | | | | Zhone |
----------------- ----------------- ---------------
Here's the situation: I receive calls from the PSTN
2011 Jul 25
0
Registration problems, Linksys SPA 3102 on Asterisk 1.4.20
Sorry, I am resending this, I tried earlier, but I
couldn't see it appear on the archives -
apologogies if it appears double!
--------------------------------------------------
My Sipura 3000 ATA died on me this morning. I had
a Linksys SPA 3102 available which I would like to
use as a replacement. Unfortunately, the SPA3102
is not able to register with the asterisk server -
I am
2008 Dec 18
1
Ghost in the Channel-Banks
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
I've been struggling with an ongoing problem the last month.
Here is the layout of the wiring:
T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank
(same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server
zap card > fax channel bank
(same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1
2003 Sep 05
1
ISDN Primary Rate Interface (PRI) - 2B Transfer
Does * support ISDN Primary Rate Interface (PRI) - 2B Transfer Capability
for T-1/PRI?
In other words the ability to take a call and join it to another call and
then drop off letting the CO-switch take over.
-Kevin
Kevin Fjelsted, President
AltiCom CTI, Inc.
Track Me Down!
One number Access, Press 11# during the voice mail message greeting
to have me F-O-U-N-D!
Phone: 612.259.0722
Fax:
2013 Mar 29
0
Getting Unknown Error while configuring Asterisk with Linux HA
Hi,
I recently configured Linux HA for Asterisk service (using Asterisk
resource agent downloaded from link:
https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk
).
As per configuration it is working good but when I include "monitor_sipuri="
sip:42 at 10.3.152.103" " parameter in primitive section it is giving me an
errors like listed below;
root at
2008 Dec 03
0
chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to
Hello,
I need help for that error message:
?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE
to?
My network is:
Client1--
-----------asterisk1------asterisk2
Client2--
? With client1, I do a call
? Asterisk1 forward the call to asterisk2
? Asterisk2 forward the call to asterisk1
? Asterisk1 forward the call to
2008 Dec 03
0
problem with RTP
Hello,
My network is:
Client_SS7_1--
-----------asterisk1------asterisk2
Client_SS7_2--
? I receive a fax from Client_SS7_1
? Asterisk1 forward the call to asterisk2
? Asterisk2 forward the call to asterisk1
? Then, asterisk2 forward the fax to Client_SS7_2
I want that the SIP signaling go to asterisk2,
But, I need that the RTP don?t go
2005 Oct 06
0
Issue with trunking
Hi all.
Ive recently setup two Asterisk boxes (running Asterisk@Home to be specific), and Im trying to get a trunk going between them.
So far I have tried a combination of IAX and SIP configuring them through AMP and writing the config files manually, but I cant seem to get calls going between the two.
I have named each box asterisk1 and asterisk2.
Does anyone have some working SIP and/or IAX
2013 Oct 07
1
Dahdi not detecting hangup when analog forwarding
Hello,
I've got a test setup with 2 asterisk boxes:
Asterisk1 with:
asterisk 11.5.1
dahdi 2.7.0.1
Digium TDM400 with 2 FXO ports
Asterisk2 with:
asterisk 11.5.1
dahdi 2.7.0
Digium TDM400 with 2 FXS ports
Asterisk1 has the following AEL Dialplan:
context remote {
s => {
Answer();
Dial(DAHDI/g1/7005);
};
};
When a call from Asterisk2 comes in, it is correctly
2014 Sep 24
0
Identifying frequency tone in Asterisk
Hi,
I have 2 Asterisk systems and a unique scenario where I need to play a
particular tone on Asterisk1 and identify the same tone on Asterisk2.
Following is my call flow,
Asterisk1(Plays audiofile1,Wait for 2 Sec,Plays a tone,Plays audiofile2) ->
PSTN -> 3rd Party CONFERENCE SYSTEM <- PSTN <- Asterisk2(Record
audiofile1,Wait for a tone,Record audiofile2).
A few points to keep in
2007 Apr 24
0
3 way calls and meetme problem
Hello,
I have a problem with the meetme application, but I'm not sure if it's a
bug or just a misuse.
I'm trying to get a 3 way call system working as follow :
A calls C
B calls C
C who's speaking with A or B, presses one keypad (only one)
and the 2 incoming SIP (A, B) and C are redirected into a conference room.
Therefore, I created an entry in the applicationmap
2010 Feb 19
1
transcoding with TC400P
Hello,
I have transcoding card TC400P installed in server running Debian with
Asterisk 1.4.23. Everything seams to be fine and after I boot up
server I see in dmesg:
7.590966] Zapata Telephony Interface Registered on major 196
[ 7.590966] Zaptel Version: 1.4.12.1
[ 7.590966] Zaptel Echo Canceller: MG2
[ 7.610963] zttranscode: Loaded.
[ 7.618969] wctc4xxp: tc400b0: Attached to
2006 Jun 01
1
audio streaming points different with VRRP
Hi!I've a question:
I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5
second, using the VRRP protocol, where must I set the IP for the
connection goes on the second asterisk?
I want this:
I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the
other asterisk but not the audio streaming...the callers are always pointed
to asterisk1, but for the
2007 Jul 31
1
g729 setup help
Hi
I am trying to make this setup work
phone1---g729---asterisk1---sip---asterisk2---g729---phone2
I have tried several configurations but none worked
I keep getting transcoding errors
I have installed one g729 licence on each asterisk, but I can't verifiy
because the show g729 command is not available,
I use 1.2.17
Do I need 2 g729 licences per asterisk ?
Do I need to register
2012 Sep 17
1
iax2 trunks between asterisk servers
Hi,
I am using iax2 trunks between asterisk servers and am having a callerid
problem. We are using realtime sip clients distributed between multiple
servers. Only in test now but have run into a calleeid problem - the
name of the called party is not displayed if the called party is on a
different server, it works if the called party is on the same server.
On each server sip clients show calleeid
2013 Feb 21
1
CDR direct executed failed
Hi,
I have configured the cdr throught ODBC with this files:
/etc/cdr_odbc.conf
[global]
dsn=asterisk2
;loguniqueid=yes
dispositionstring=yes
table=cdr ;"cdr" is default table name
usegmtime=no ; set to "yes" to log in GMT
hrtime=yes ;Enables microsecond accuracy with the billsec and duration fields
/etc/cdr.conf
[general]
enable=yes
unanswered =
2006 Apr 07
0
Dial Plan Problem with extensions ringing multiple phones connected on different * servers
Hi all
I wonder how to solve this issue:
Asterisk1: 2 BRI Cards, TE and NT Mode.
- ISDN In (From telco)
- ISDN out (to a phone) (Zap/g6)
exten => 999999,1,Dial(IAX2/key@asterisk2/999999&Zap/g6/999999)
Asterisk2: Just different kind of SIP Connections.
exten => 999999,1,Dial(SIP/999999,20,r)
exten => 999999,n,Voicemail(u999999)
exten => 999999,n,Hangup
Now when a call commes
2003 Mar 12
4
Printers Icon
I have installed Samba2.2.7a on an Ultra5 Solaris 8 Operating System. The
smb.conf file has been configured and all appears to work fine. However,
along with my shared resource, I am seeing the Printers icon. I only want
to use Samba to share resources, and not for printing. Is there a way to
make the Printers Icon invisible? I already tried the hide files, and
veto files parameters in
2004 Apr 26
0
Help with connecting 2 servers via iax
I have successfully configured two servers and I am now trying to connect
via iax. When I attempt to call from one ext, 2006(server viop1) to
extension 3006 (server voip2) I receive a timeout or "call failed 403
forbidden.
The information I am receiving from the console is below.
Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type
registered for 'IAX'