Solved

Call Progress Ring Tones not being heard over WAN with Linksys SPA942 and Asterisk server

Posted on 2009-05-16
13
681 Views
Last Modified: 2013-11-12
I am not able to hear any call progress rings over the WAN on my Linksys spa942 IP Phone, to an asterisk 1.4.23 server.  I am able to hear the person who picks up and they are also able to hear me.  I am able to hear the dial tone before I dial.  The problem is while the call is being placed, until the other person picks up the phone, I do not hear any ringing in the head set.  I have proxy and outbound proxy enabled on the phone and have also tried the STUN server setting.  The phone rings normally when someone calls me on the Linksys spa942.

The problem does not occur if the Linksys phone is on the same LAN as the asterisk box.

There are NAT firewalls at both locations.

I have ports 5060 open on both firewalls, and have also tried opening 10000-20000

Any ideas??
0
Comment
Question by:jkockler
[X]
Welcome to Experts Exchange

Add your voice to the tech community where 5M+ people just like you are talking about what matters.

  • Help others & share knowledge
  • Earn cash & points
  • Learn & ask questions
  • 6
  • 4
  • 3
13 Comments
 
LVL 32

Expert Comment

by:harbor235
ID: 24412478


what codec are you using? Have you tried G729?

harbor235 ;}
0
 
LVL 4

Author Comment

by:jkockler
ID: 24413013
Have not tried that yet... Where do I specify G729?  On the Linksys PAP2?
0
 
LVL 32

Expert Comment

by:harbor235
ID: 24414385


You configure it on the Asterisk box

http://www.sureteq.com/blog/?p=65


harbor235 ;}
0
Industry Leaders: We Want Your Opinion!

We value your feedback.

Take our survey and automatically be enter to win anyone of the following:
Yeti Cooler, Amazon eGift Card, and Movie eGift Card!

 
LVL 4

Author Comment

by:jkockler
ID: 24414399
I tried different codecs but I do not think that is the issue.

I am almost certain the issue has something to do with NAT.  It only seems to effect the call progress tones but none the less here is what I found.

Originally this problem was only over the WAN but when I specified external and lan ip schemes in sip.conf , suddenly I was no longer able to hear call tones when calling from the same LAN as the asterisk server.  Remember this problem did not occur on the LAN with asterisk.  I am thinking I have something wrong in the external and internal ip settings in sip.conf
Here is what I have

externip = xx.xxx.xx.221 ;(substitute your public ip address)
localnet = 192.168.0.0/255.255.255.0 ;(substitute your lan subnet address)
nat=yes

I
0
 
LVL 32

Expert Comment

by:harbor235
ID: 24414448
0
 
LVL 19

Accepted Solution

by:
feptias earned 500 total points
ID: 24415659
Is this question relevant?
http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Asterisk_/Q_24415478.html

If not, can you collect a sip trace for the packets exchanged during call setup, until the point where the call is answered and speech is heard. I suggest using the "sip debug" mechanism in Asterisk. At the Asterisk CLI switch on the trace with "sip set debug" and switch it off with "sip set debug off". Post it here as a code snippet please.
0
 
LVL 4

Author Comment

by:jkockler
ID: 24416998
Thanks.  I will get that debug snippet.

Is there a way to get the sip debug to show a time-stamp?  I usually keep it on all the time, but it would be great if I could see what time everything happened.
0
 
LVL 19

Expert Comment

by:feptias
ID: 24419348
Edit /etc/asterisk/logger.conf and add the following line (it may already be there, but commented out):
dateformat=%F %T
0
 
LVL 4

Author Comment

by:jkockler
ID: 24422727
What do you mean by commented out?  Here is what I have in the logger.conf for the date

[general]
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS)
; see strftime(3) Linux manual for format specifiers
;dateformat=%F %T
0
 
LVL 19

Expert Comment

by:feptias
ID: 24422849
"commented out" is programmer-speak for lines that have the semi-colon in front of them. In Asterisk conf files, the semi-colon marks that line as a comment line, and it is ignored by the Asterisk parser. What I meant was "remove the semi-colon" from that line:

[general]
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS)
; see strftime(3) Linux manual for format specifiers
dateformat=%F %T
0
 
LVL 4

Author Comment

by:jkockler
ID: 24426494
Gotcha.  I commented it out but it still does not show the time stamp.  I reloaded and restarted asterisk after.  Any other ideas?

I actually was able to get this issue with the ringing resolved through opening up the rtp port range 10000-20000 on the asterisk side again.  That fixed it this time, maybe last time I did not wait long enough.

Is there anything wrong with keeping sip debugging enabled all the time?  Thanks!
0
 
LVL 19

Assisted Solution

by:feptias
feptias earned 500 total points
ID: 24429443
I only enable sip debug during testing or when solving a problem. It is strongly recommended that any type of debug setting be disabled on production systems. The disadvantages of leaving sip debugging enabled all the time are:
1. Extra work for Asterisk
2. If you open the CLI (using asterisk -r) then the screen will just constantly be scrolling with "stuff" which makes it very difficult to run diagnostic commands such as "core show channels" or "sip show peers" etc.
3. If you are writing it to the log files then it will fill up disk space like crazy and you will have to use logrotate or something similar to keep the log files under control.

When debugging a problem I would try to focus it on a specific peer - sip set debug peer <peer_name>
I also often use a tool called "sipgrep" which runs from the Linux command prompt, not from the Asterisk CLI. sipgrep reports the SIP packets that match a specific string in the To or From headers so you don't get overwhelmed with data related to other calls on the system. It is also easy to pipe the output to a file.
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/utils/sipgrep/

In logger.conf you can configure it so the sip debug is sent to the /var/log/messages file and this must be the place where the time/date gets written (perhaps it does not get written to the console). To make it send sip debug to your log file, add the word "verbose" to the list of notifications for the messages log file:
[general]
dateformat=%F %T
...
[logfiles]
console => notice,warning,error
messages => notice,warning,error,verbose
...
0
 
LVL 4

Author Closing Comment

by:jkockler
ID: 31582287
Thanks for all the extra info fepitas.  In case someone is having the same problem, this issue was actually resolved by opening ports 10000-20000 on the firewall where asterisk resides.
0

Featured Post

Don't Miss ATEN at InfoComm 2017!

Visit booth #2167 to see the  new ATEN VM3200 32 x 32 Modular Matrix Switch. Other highlights include the VE8950 4K HDMI Over IP Extender, VS1912 12-Port DP Video Wall Media Player  and VK2100 ATEN Control System. Register now with Free Pass Code ATEN288!

Question has a verified solution.

If you are experiencing a similar issue, please ask a related question

If your business is like most, chances are you still need to maintain a fax infrastructure for your staff. It’s hard to believe that a communication technology that was thriving in the mid-80s could still be an essential part of your team’s modern I…
When you try to share a printer , you may receive one of the following error messages. Error message when you use the Add Printer Wizard to share a printer: Windows could not share your printer. Operation could not be completed (Error 0x000006…
Here's a very brief overview of the methods PRTG Network Monitor (https://www.paessler.com/prtg) offers for monitoring bandwidth, to help you decide which methods you´d like to investigate in more detail.  The methods are covered in more detail in o…
This video gives you a great overview about bandwidth monitoring with SNMP and WMI with our network monitoring solution PRTG Network Monitor (https://www.paessler.com/prtg). If you're looking for how to monitor bandwidth using netflow or packet s…
Suggested Courses

690 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question