[asterisk-bugs] [Asterisk 0012288]: Remote Asterisk Does not accept Digits pressed.
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Mar 24 09:43:42 CDT 2008
The following issue has been RESOLVED.
======================================================================
http://bugs.digium.com/view.php?id=12288
======================================================================
Reported By: rahulrborkar
Assigned To: qwell
======================================================================
Project: Asterisk
Issue ID: 12288
Category: . I did not set the category correctly.
Reproducibility: always
Severity: major
Priority: normal
Status: resolved
Asterisk Version: 1.4.18
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
Resolution: no change required
Fixed in Version:
======================================================================
Date Submitted: 03-24-2008 05:57 CDT
Last Modified: 03-24-2008 09:43 CDT
======================================================================
Summary: Remote Asterisk Does not accept Digits pressed.
Description:
Hi Folks,
I have installed asterisk 1.4.18 on REMOTE suse Linux enterprise 10. Also
this machine has Digium TDM2400P card installed.
I have also installed festival 1.95 via rpm on remote suse 10.
after that I have written following code in extensions.conf,
my extension.conf is as follows,
exten => 123,1,Answer()
exten => 123,2,Festival("Hi How are you")
exten => 123,3,Hangup()
when I dial from X-Lite under from my home or office machine, which is not
in the same network where asterisk is running.
I get following response,
-- Executing [9009 at sip:1] Answer("SIP/101-081ea6e8", "") in new stack
-- Executing [9009 at sip:2] Festival("SIP/101-081ea6e8", "hi How are you")
in new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (sip, 9009, 2) exited non-zero on 'SIP/101-081ea6e8'
[Mar 19 10:48:42] WARNING[19352]: chan_sip.c:1947 retrans_pkt: Maximum
retries exceeded on transmission
Y2QxMmY0ODkxZTdmYjY3YTQ3Yjg5NTU3YTljZWZlMzk. for seqno 2 (Critical
Response)
PostPosted: Wed Mar 19, 2008 2:13 am Post subject: Not hearing
festival/Cepstral voices from remote asterisk. Reply with quote
Hi Folks,
I have asterisk 1.4.18 on REMOTE suse Linux enterprise 10. Also this
machine has Digium TDM2400P card installed.
I'm using X-Lite here as a soft phone on windows machine which is under
different LAN. My X-LITE registered to remote machine via it's Public IP.
Now my Problem is,
I have installed festival 1.95 via rpm on remote suse 10.
after that I have written following code in extensions.conf,
exten => 123,1,Answer()
exten => 123,2,Festival("Hi How are you")
exten => 123,3,Hangup()
asterisk shows me following things on console,
-- Executing [9009 at sip:1] Answer("SIP/101-081ea6e8", "") in new stack
-- Executing [9009 at sip:2] Festival("SIP/101-081ea6e8", "hi How are you")
in new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (sip, 9009, 2) exited non-zero on 'SIP/101-081ea6e8'
[Mar 19 10:48:42] WARNING[19352]: chan_sip.c:1947 retrans_pkt: Maximum
retries exceeded on transmission
Y2QxMmY0ODkxZTdmYjY3YTQ3Yjg5NTU3YTljZWZlMzk. for seqno 2 (Critical
Response)
My Festival showed me following log,
$ festival --server
server Wed Mar 19 10:47:29 2008 : Festival server started on port 1314
client(1) Wed Mar 19 10:48:22 2008 : accepted from localhost
client(1) Wed Mar 19 10:48:22 2008 : disconnected
Note: Festival gets disconnected immediately.
My sip.conf is as follows, I've registered x-lite to 101.
[101]
callerid=101 <101>
canreinvite=no
dtmfmode=rfc2833
host=dynamic
nat=yes
port=5060
bindport=5060
bindaddr=0.0.0.0
qualify=yes
record_in=Adhoc
record_out=Adhoc
secret=101
type=friend
context=sip
username=101
allow=all
Problem is :
I'm not able to hear sound "Hi how are you" on x-lite.
when it comes at Festival it directly gives following response.
PostPosted: Wed Mar 19, 2008 2:13 am Post subject: Not hearing
festival/Cepstral voices from remote asterisk. Reply with quote
Hi Folks,
I have asterisk 1.4.18 on REMOTE suse Linux enterprise 10. Also this
machine has Digium TDM2400P card installed.
I'm using X-Lite here as a soft phone on windows machine which is under
different LAN. My X-LITE registered to remote machine via it's Public IP.
Now my Problem is,
I have installed festival 1.95 via rpm on remote suse 10.
after that I have written following code in extensions.conf,
exten => 123,1,Answer()
exten => 123,2,Festival("Hi How are you")
exten => 123,3,Hangup()
asterisk shows me following things on console,
-- Executing [9009 at sip:1] Answer("SIP/101-081ea6e8", "") in new stack
-- Executing [9009 at sip:2] Festival("SIP/101-081ea6e8", "hi How are you")
in new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (sip, 9009, 2) exited non-zero on 'SIP/101-081ea6e8'
[Mar 19 10:48:42] WARNING[19352]: chan_sip.c:1947 retrans_pkt: Maximum
retries exceeded on transmission
Y2QxMmY0ODkxZTdmYjY3YTQ3Yjg5NTU3YTljZWZlMzk. for seqno 2 (Critical
Response)
RTP-stats
* Our Receiver:
SSRC: 0
Received packets: 0
Lost packets: 0
Jitter: 0.0000
Transit: 0.0000
RR-count: 0
* Our Sender:
SSRC: 1443552997
Sent packets: 0
Lost packets: 0
Jitter: 0
SR-count: 0
RTT: 0.000000
My Festival showed me following log,
$ festival --server
server Wed Mar 19 10:47:29 2008 : Festival server started on port 1314
client(1) Wed Mar 19 10:48:22 2008 : accepted from localhost
client(1) Wed Mar 19 10:48:22 2008 : disconnected
Note: Festival gets disconnected immediately.
my sip.conf is as follows, I've registered x-lite to 101.
[101]
callerid=101 <101>
canreinvite=no
dtmfmode=rfc2833
host=dynamic
nat=yes
port=5060
bindport=5060
bindaddr=0.0.0.0
qualify=yes
record_in=Adhoc
record_out=Adhoc
secret=101
type=friend
context=sip
username=101
allow=all
[As my Suse machine is on remote ip I've enabled NAT and Qualify under
this configuration]
Problem 1:
I'm not able to hear sound "Hi how are you" on x-lite.
when it comes to festival asterisk is directly gives following error,
Spawn extension (sip, 9009, 2) exited non-zero on 'SIP/101-081ea6e8'
and it does not go to 3rd priority to hangup.
Problem 2: asterisk does not accept digits pressed from remote x-lite
Also MAIN PROBLEM HERE I WANT TO REPORT IS,
When I dial extension 123 in above example call gets answered also it goes
to second extension and if we have used some ready made file there for
playing and given escape digits.
eg:
exten => 123,1,Answer()
exten => 123,2,Background(bigReadyMadeFileName)
exten => 123,3,Answer()
when I press certain digits Asterisk should stop playing.
But this remote asterisk does not at all accepts Digits pressed.
Note That :
1. I have created a similar setup here on local machine and it works fine
even with Cepstral.
2. I have also tried this with asterisk 1.4.16 and found no difference in
behavior.
Is it a bug in asterisk or does this have solution?
Please help me I've stuck here for more than a month.
======================================================================
----------------------------------------------------------------------
qwell - 03-24-08 09:43
----------------------------------------------------------------------
It was incredibly difficult to read the description here. It made very
little sense, and had very contradicting information.
I am closing this, as we suspect that it is a configuration issue, and
this is not the place to get configuration help.
Please try the asterisk-users mailing list. Be sure to give clear and
concise information, if you want people to help you.
Issue History
Date Modified Username Field Change
======================================================================
03-24-08 09:43 qwell Status new => resolved
03-24-08 09:43 qwell Resolution open => no change
required
03-24-08 09:43 qwell Assigned To => qwell
03-24-08 09:43 qwell Note Added: 0084442
======================================================================
More information about the asterisk-bugs
mailing list