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Summary:ASTERISK-11711: Remote Asterisk Does not accept Digits pressed.
Reporter:Rahul Borkar (rahulrborkar)Labels:
Date Opened:2008-03-24 05:57:30Date Closed:2011-06-07 14:02:57
Priority:MajorRegression?No
Status:Closed/CompleteComponents:. I did not set the category correctly.
Versions:Frequency of
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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@sip:1] Answer("SIP/101-081ea6e8", "") in new stack
-- Executing [9009@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@sip:1] Answer("SIP/101-081ea6e8", "") in new stack
-- Executing [9009@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@sip:1] Answer("SIP/101-081ea6e8", "") in new stack
-- Executing [9009@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.
 
Comments:By: Jason Parker (jparker) 2008-03-24 09:43:42

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.
By: Rahul Borkar (rahulrborkar) 2008-03-25 01:47:16

I've given here what I got on the console.
If this is configuration issue.I've tried all possible configurations in all possible files and found no difference in behavior.
I'm again giving Description so that you can get something from this.



I'm using asterisk 1.4.18 on suse linux enterprise 10 and X-Lite as soft phone.

I'm not getting why remote asterisk gives me following console log of sip retries and hangup.

-- Executing [9009@sip:1] Answer("SIP/101-081ea6e8", "") in new stack
-- Executing [9009@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)


In particular this is call to a Festival TTS in extension.conf
as follows,


exten=>9009,1,Answer()
exten=>9009,2,Festival(hi How are you)
exten=>9009,3,Hangup()

but when it tries to connect to festival it just exits.
I think i have configured sip.conf correctly but

After this, when I press any digit from X-Lite soft phone asterisk is not interpreting it and continues to execute call flow.

I have following configuration in sip.conf.

Code:


[general]
bindport=5060  
bindaddr=0.0.0.0
srvlookup=yes  


[101]              
callerid=101 <101>
canreinvite=no    
dtmfmode=rfc2833  
host=dynamic      
nat=yes          
port=5060          
qualify=yes        
record_in=Adhoc    
record_out=Adhoc  
secret=101        
type=friend        
context=sip      
username=101      



My X-Lite is registered with 101 and also I've given stun there as "stun.counterpath.net"

Can anybody please suggest something on this.

I can give any necessary information you want. But please help me out in this.

Thanks and Regards,
Rahul  
By: Dmitry Andrianov (dimas) 2008-03-25 05:31:23

Your issue will most likely be closed again becasue the description is just a mess of information. And it attempts reporting two different issues at the same time.

If the main problem is Asterisk ignoring DTMFs in Background() then forget about Festival and fill ANOTHER issue with clear description what is going on without mixing information with some unsuccessfull attempts of using Festival.

However I would suspect you have problems with firewall/NAT and RTP stream from your X-Lite softphone to Asterisk is not established. This is why RFC2833 DTMF not reaching Asterisk.
By: Rahul Borkar (rahulrborkar) 2008-03-25 06:41:02

So does this is problem with port forwarding and nothing else?
If "yes" then I've forwarded all following ports but found no difference.

5060,5061,8000,8766 to 8800.
for both TCP and UDP

also I'm not getting any help regarding this on portal even after posting a very specific information. what should I do now?
By: Dmitry Andrianov (dimas) 2008-03-25 06:48:22

I'm afraid these ports are not enough because RTP can use high-numbeed ports. http://www.voip-info.org/wiki/view/NAT+and+VOIP says "RTP audio: Ports 8766 to 35000" not "to 8800"


By: Rahul Borkar (rahulrborkar) 2008-03-25 06:53:16

I'll try by forwarding all RTP ports but does that affect security of Asterisk machine?
Also I've specified stun under my x-lite soft phone client. So does it still need all ports to be forwarded?


Please note that this was working when I was using asterisk 1.2.12 and after I formatted machine and installed asterisk 1.4.18 it stopped taking Pressed digits.


By: Donny Kavanagh (donnyk) 2008-03-25 13:37:50

Edit rtp.conf to change the ports asterisk will try to use for rtp.
By: Rahul Borkar (rahulrborkar) 2008-03-26 01:06:21

I've done that. But that does not created much difference.
Also I want to note here that, For same network where asterisk server is located it is working fine.

But if we access asterisk remotely by X-Lite then it has problem.

Guys, I really do not want to make this as a Forum but Problem is I really did not get any response from forums.digium.com.

Really thanks to all of you.
By: Dmitry Andrianov (dimas) 2008-03-26 03:14:38

You can use verious tools like tcpdump/ethereal/wireshark to dump UDP traffic on that box checking if RTP from X-Lite comes in or not.If it does not - you need to deal with your firewall/router/NAT device. Maybe you need to deal with iptables on the Asterisk server as well. And finally, if NAT is involved you need to properly edit sip.conf.

However I'm almost 100% sure it is not a bug - it is just a configuration issue. So you need to seek for help not here but in Google, -users mailing list and on asterisk users IRC channel.
By: Rahul Borkar (rahulrborkar) 2008-03-26 03:34:41

ok! Thanks Folks!
I'll go on there and check

Thanks for all your replies.
I really appreciate that.
By: Jason Parker (jparker) 2008-03-26 09:39:41

Closing, per my original comment and repeated comments in the same regard.