[sip-comm] Asterisk Connection Problem


#1

Hi,

I seem to have a problem with using Sip Communicator with Asterisk. I'm able to register alright with Asterisk, but when I make a call from the Sip Communicator to another phone (Grandstream 100 or X-Lite soft phone, both registered with Asterisk), the receiver rings, but when it picks up the dialer (Sip Communicator) doesn't receive a message that the other side has picked up. So Sip Communicator still displays the "dialing" message, and the other side thinks there's already a connection set up. When I make calls INTO Sip Communicator, Sip Communicator doesn't detect that there is a call coming in.

I tried not registering with Asterisk, and making sip calls directly from Sip Communicator to another phone, or the other way around, and the problem still occurs.

In addition to this, when I make calls with Asterisk there is a voice message that plays back which tells me which extension I've dialed. I can't hear this message at all in Sip Communicator, but I can hear the "ringing" sound when someone is Dialing in.

I've attached my property settings and stack log from the log directory. For some reason it does not generate another log file.

Here are my property settings:

Thanks!

#SipCommunicator Properties

#Media properties
net.java.sip.communicator.media.IP_ADDRESS=
net.java.sip.communicator.media.AUDIO_PORT=
net.java.sip.communicator.media.MEDIA_SOURCE=
net.java.sip.communicator.media.PREFERRED_AUDIO_ENCODING=PCMU
net.java.sip.communicator.media.PREFERRED_VIDEO_ENCODING=
net.java.sip.communicator.media.VIDEO_PORT=

#SIP properties
net.java.sip.communicator.sip.STACK_PATH=gov.nist
net.java.sip.communicator.sip.PUBLIC_ADDRESS=sip:103@192.168.0.41
net.java.sip.communicator.sip.DISPLAY_NAME=phone3
net.java.sip.communicator.sip.TRANSPORT=UDP
net.java.sip.communicator.sip.PREFERRED_LOCAL_PORT=5060
net.java.sip.communicator.sip.REGISTRAR_ADDRESS=192.168.0.41
net.java.sip.communicator.sip.REGISTRAR_TRANSPORT=UDP
net.java.sip.communicator.sip.REGISTRATIONS_EXPIRATION=3600
net.java.sip.communicator.sip.REGISTRAR_PORT=5060

#JAIN SIP defined properties
javax.sip.IP_ADDRESS=192.168.0.13
javax.sip.STACK_NAME=sip-communicator
javax.sip.ROUTER_PATH=
javax.sip.RETRANSMISSON_FILTER=
javax.sip.OUTBOUND_PROXY=
javax.sip.EXTENSION_METHODS=

#NIST SIP logging properties
gov.nist.javax.sip.TRACE_LEVEL=32
gov.nist.javax.sip.SERVER_LOG=log/sip-communicator.stack.log

#log4J

#log4j.logger.net.java.sip.communicator=WARN
#log4j.logger.net.java.sip.communicator.sip=WARN
#log4j.logger.net.java.sip.communicator.media=WARN
#log4j.logger.net.java.sip.communicator.gui=WARN

log4j.rootLogger=net.java.sip.communicator.common.Console.TraceLevel, RFLogger

#log4j.appender.stdout=org.apache.log4j.ConsoleAppender
#log4j.appender.stdout.layout=org.apache.log4j.PatternLayout

# Pattern to output the caller's file name and line number.
#log4j.appender.stdout.layout.ConversionPattern=%5p [%t] (%F:%L) - %m%n

log4j.appender.RFLogger=org.apache.log4j.RollingFileAppender
log4j.appender.RFLogger.File=log/sip-communicator.app.log

log4j.appender.RFLogger.MaxFileSize=256KB
# Keep one backup file
log4j.appender.RFLogger.MaxBackupIndex=1

log4j.appender.RFLogger.layout=org.apache.log4j.PatternLayout
log4j.appender.RFLogger.layout.ConversionPattern=%r [%t] %p %c{2} %x - %m%n

sip-communicator.stack.log (17.7 KB)


#2

Hello Jason,

There are some known problems with Asterisk. There will be someone
looking at them till the end of next week. In the mean time could you
please send me the contents of your sip-communicator/log/ directory so
that I could add it to the bug report list?

Thanks
Emil

Hi,

I seem to have a problem with using Sip Communicator with Asterisk. I'm
able to register alright with Asterisk, but when I make a call from the
Sip Communicator to another phone (Grandstream 100 or X-Lite soft phone,
both registered with Asterisk), the receiver rings, but when it picks up
the dialer (Sip Communicator) doesn't receive a message that the other
side has picked up. So Sip Communicator still displays the "dialing"
message, and the other side thinks there's already a connection set up.
When I make calls INTO Sip Communicator, Sip Communicator doesn't detect
that there is a call coming in.

I tried not registering with Asterisk, and making sip calls directly
from Sip Communicator to another phone, or the other way around, and the
problem still occurs.

In addition to this, when I make calls with Asterisk there is a voice
message that plays back which tells me which extension I've dialed. I
can't hear this message at all in Sip Communicator, but I can hear the
"ringing" sound when someone is Dialing in.

I've attached my property settings and stack log from the log directory.
For some reason it does not generate another log file.

Here are my property settings:

Thanks!

#SipCommunicator Properties

#Media properties
net.java.sip.communicator.media.IP_ADDRESS=
net.java.sip.communicator.media.AUDIO_PORT=
net.java.sip.communicator.media.MEDIA_SOURCE=
net.java.sip.communicator.media.PREFERRED_AUDIO_ENCODING=PCMU
net.java.sip.communicator.media.PREFERRED_VIDEO_ENCODING=
net.java.sip.communicator.media.VIDEO_PORT=

#SIP properties
net.java.sip.communicator.sip.STACK_PATH=gov.nist
net.java.sip.communicator.sip.PUBLIC_ADDRESS=sip:103@192.168.0.41
net.java.sip.communicator.sip.DISPLAY_NAME=phone3
net.java.sip.communicator.sip.TRANSPORT=UDP
net.java.sip.communicator.sip.PREFERRED_LOCAL_PORT=5060
net.java.sip.communicator.sip.REGISTRAR_ADDRESS=192.168.0.41
net.java.sip.communicator.sip.REGISTRAR_TRANSPORT=UDP
net.java.sip.communicator.sip.REGISTRATIONS_EXPIRATION=3600
net.java.sip.communicator.sip.REGISTRAR_PORT=5060

#JAIN SIP defined properties
javax.sip.IP_ADDRESS=192.168.0.13
javax.sip.STACK_NAME=sip-communicator
javax.sip.ROUTER_PATH=
javax.sip.RETRANSMISSON_FILTER=
javax.sip.OUTBOUND_PROXY=
javax.sip.EXTENSION_METHODS=

#NIST SIP logging properties
gov.nist.javax.sip.TRACE_LEVEL=32
gov.nist.javax.sip.SERVER_LOG=log/sip-communicator.stack.log

#log4J

#log4j.logger.net.java.sip.communicator=WARN
#log4j.logger.net.java.sip.communicator.sip=WARN
#log4j.logger.net.java.sip.communicator.media=WARN
#log4j.logger.net.java.sip.communicator.gui=WARN

log4j.rootLogger=net.java.sip.communicator.common.Console.TraceLevel, RFLogger

#log4j.appender.stdout=org.apache.log4j.ConsoleAppender
#log4j.appender.stdout.layout=org.apache.log4j.PatternLayout

# Pattern to output the caller's file name and line number.
#log4j.appender.stdout.layout.ConversionPattern=%5p [%t] (%F:%L) - %m%n

log4j.appender.RFLogger=org.apache.log4j.RollingFileAppender
log4j.appender.RFLogger.File=log/sip-communicator.app.log

log4j.appender.RFLogger.MaxFileSize=256KB
# Keep one backup file
log4j.appender.RFLogger.MaxBackupIndex=1

log4j.appender.RFLogger.layout=org.apache.log4j.PatternLayout
log4j.appender.RFLogger.layout.ConversionPattern=%r [%t] %p %c{2} %x - %m%n

Cheers
Emil

http://www.emcho.com

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#3

Emil,

Also, I don't know if this makes a difference, but with all other SIP clients I've found that in the "contact" field they included the full sip address of the client, i.e.

Contact: <sip:103@192.168.0.41>

But about half the time Sip Communicator puts in everything but the user name, which in this case is "103":

Contact: "phone3" <sip:192.168.0.13:5060;transport=udp>

Could this be the cause of some trouble?

Thanks,
Jason

Emil Ivov wrote:

···

Hello Jason,

There are some known problems with Asterisk. There will be someone
looking at them till the end of next week. In the mean time could you
please send me the contents of your sip-communicator/log/ directory so
that I could add it to the bug report list?

Thanks
Emil

Hi,
   
I seem to have a problem with using Sip Communicator with Asterisk. I'm
able to register alright with Asterisk, but when I make a call from the
Sip Communicator to another phone (Grandstream 100 or X-Lite soft phone,
both registered with Asterisk), the receiver rings, but when it picks up
the dialer (Sip Communicator) doesn't receive a message that the other
side has picked up. So Sip Communicator still displays the "dialing"
message, and the other side thinks there's already a connection set up.
When I make calls INTO Sip Communicator, Sip Communicator doesn't detect
that there is a call coming in.
   
I tried not registering with Asterisk, and making sip calls directly
from Sip Communicator to another phone, or the other way around, and the
problem still occurs.
   
In addition to this, when I make calls with Asterisk there is a voice
message that plays back which tells me which extension I've dialed. I
can't hear this message at all in Sip Communicator, but I can hear the
"ringing" sound when someone is Dialing in.
   
I've attached my property settings and stack log from the log directory.
For some reason it does not generate another log file.
   
Here are my property settings:
   
Thanks!
   
#SipCommunicator Properties
   
#Media properties
net.java.sip.communicator.media.IP_ADDRESS=
net.java.sip.communicator.media.AUDIO_PORT=
net.java.sip.communicator.media.MEDIA_SOURCE=
net.java.sip.communicator.media.PREFERRED_AUDIO_ENCODING=PCMU
net.java.sip.communicator.media.PREFERRED_VIDEO_ENCODING=
net.java.sip.communicator.media.VIDEO_PORT=
   
#SIP properties
net.java.sip.communicator.sip.STACK_PATH=gov.nist
net.java.sip.communicator.sip.PUBLIC_ADDRESS=sip:103@192.168.0.41
net.java.sip.communicator.sip.DISPLAY_NAME=phone3
net.java.sip.communicator.sip.TRANSPORT=UDP
net.java.sip.communicator.sip.PREFERRED_LOCAL_PORT=5060
net.java.sip.communicator.sip.REGISTRAR_ADDRESS=192.168.0.41
net.java.sip.communicator.sip.REGISTRAR_TRANSPORT=UDP
net.java.sip.communicator.sip.REGISTRATIONS_EXPIRATION=3600
net.java.sip.communicator.sip.REGISTRAR_PORT=5060
   
#JAIN SIP defined properties
javax.sip.IP_ADDRESS=192.168.0.13
javax.sip.STACK_NAME=sip-communicator
javax.sip.ROUTER_PATH=
javax.sip.RETRANSMISSON_FILTER=
javax.sip.OUTBOUND_PROXY=
javax.sip.EXTENSION_METHODS=
   
#NIST SIP logging properties
gov.nist.javax.sip.TRACE_LEVEL=32
gov.nist.javax.sip.SERVER_LOG=log/sip-communicator.stack.log
   
#log4J
   
#log4j.logger.net.java.sip.communicator=WARN
#log4j.logger.net.java.sip.communicator.sip=WARN
#log4j.logger.net.java.sip.communicator.media=WARN
#log4j.logger.net.java.sip.communicator.gui=WARN
   
log4j.rootLogger=net.java.sip.communicator.common.Console.TraceLevel, RFLogger
   
#log4j.appender.stdout=org.apache.log4j.ConsoleAppender
#log4j.appender.stdout.layout=org.apache.log4j.PatternLayout
   
# Pattern to output the caller's file name and line number.
#log4j.appender.stdout.layout.ConversionPattern=%5p [%t] (%F:%L) - %m%n
   
log4j.appender.RFLogger=org.apache.log4j.RollingFileAppender
log4j.appender.RFLogger.File=log/sip-communicator.app.log
   
log4j.appender.RFLogger.MaxFileSize=256KB
# Keep one backup file
log4j.appender.RFLogger.MaxBackupIndex=1
   
log4j.appender.RFLogger.layout=org.apache.log4j.PatternLayout
log4j.appender.RFLogger.layout.ConversionPattern=%r [%t] %p %c{2} %x - %m%n
   
Cheers
Emil

http://www.emcho.com

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