Topic: Asterisk - selintra sail

[tias]

Hi, recently got problem with my phones. Since when I worked last time I had done several things. The most recent thing is to change my router to another brand. I have also tried to update my asterisk.

I can make calls but no audio is transfered, just silence.

The package that is installed is smeserver-asterisk-sounds-1.2.2-2, smeserver-asterisk-zappri-MPP-1.2.6-5 and smeserver-asterisk-1.2.10-5.

I also got Selintra-sail installed, version 2.1.14-347.

When I log in to my asterisk with asterisk -r and making a sip show peers, I can see my extensions. And they seem to report in fine (except for the ata-box, but more of it later). But when I try to make a call from two extensions with x-lite, the server makes the call, but no audio.

I have also had some suspect problem with the ata-box. It got unreachable. It was configured to connect to my server over the local lan on port 5060, but the only way to get asterisk to reach it, is it configure the ata box to use uPNP. I had no problem like this before...

My router is configured to forward port 10 000-20 000 UDP and TCP to mine server and port 5060 UDP and TCP.

My system is a VIA, but before I was struggling arround with the upgrade everything worked out well. Saw in a thread that the rpm from Selintra wouldn't work with VIA, but the thread owner had a C3 in the system. Mine CPU is an Eden (EPIA5000)

Does anyone have a suggestion what to do to solve this? Would be nice to have my phones up and running again.

Regards,
Tias

[SARK devs]

Hello tias

Sorry you've been having trouble.

Unfortunately there are quite a few reasons why rtp traffic may not be transferring correctly between phones.   Is your box running server-only or server-gateway?  You don't say and it can be important.

When you changed the router, did it change the IP address of your SAIL box?  If it did then check that you have the correct address set in globals and the correct subnet set in the SIP header - you may be unintentionally spoofing the phones (although it is unlikely if they are registering correctly).

To see what is happening at the packet level, install ethereal on your sail box...

yum install ethereal --enablerepo=base

Once it's installed you can watch the packets leaving and arriving (and  see where they are going) by running tethereal (notice the "t" at the front) at the SME CLI.  You will have to read up on ethereal (it's a big old product and I can't teach you the switches and what they do here).  However, as an example, to watch the rtp packets leaving and arriving for a locally attached phone you can use something like....

tethereal -R rtp -i eth0 -f "host nnn.nnn.nnn.nnn"

where nnn.nnn.nnn.nnn is the address of the phone.  
Similarly, to watch the SIP packets do....

tethereal -R sip -i eth0 -f "host nnn.nnn.nnn.nnn"


Ethereal is great for solving these kinds of problems and its also a superb learning tool if you want to understand SIP packets and how they work.

Let us know how you get on.

Kind Regards

Selintra

[tias]

Hello,

The server got the same IP as before and runs server-only. All traffic are located inside the network. Haven't got time yet to install ethereal but will do that in the evening.

best regards,
/Tias

[tias]

Okey, Installed the package and ran the command:

 tethereal -R rtp -i eth0 -f "host nnn.nnn.nnn.nnn"

where i replaced nnn.nnn....
with a computer with x-lite active. Then I made a call between that extension and another one also running x-lite. No package seems to travel... I got no response at all.

If I instead choose sip tethereal wakes and outputs all kind of stuff.

regards,
Tias

[SARK devs]

OK good,

Now, run the call again but look at the asterisk console (get to it with asterisk - rvvvv).  See what asterisk has to say about the call.  It looks like it  may be a codec problem.

Best

Selintra

[tias]

Hello,
I got the following info from the asterisk CLI.

Connected to Asterisk 1.2.10 currently running on heffa (pid = 8881)
Verbosity was 3 and is now 4
    -- Remote UNIX connection
    -- Executing AGI("SIP/5000-0989e978", "selintra|OutCluster|5001") in new sta
ck
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing AGI("SIP/5000-0989e978", "selintra|InCall|") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (DBget) Options: (dbVal=STAT/OCSTAT)
May  7 19:11:30 WARNING[8952]: app_db.c:226 get_exec: This application has been
deprecated, please use the ${DB(family/key)} function instead.
    -- DBget: varname=dbVal, family=STAT, key=OCSTAT
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=cfimopen/5001)
    -- DBget: varname=dbVal, family=cfimopen, key=5001
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=cfim/5001)
    -- DBget: varname=dbVal, family=cfim, key=5001
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=STAT/OCSTAT)
    -- DBget: varname=dbVal, family=STAT, key=OCSTAT
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=cfimopen/5001)
    -- DBget: varname=dbVal, family=cfimopen, key=5001
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=STAT/OCSTAT)
    -- DBget: varname=dbVal, family=STAT, key=OCSTAT
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=cfim/5001)
    -- DBget: varname=dbVal, family=cfim, key=5001
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=ringdelay/5001)
    -- DBget: varname=dbVal, family=ringdelay, key=5001
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (Dial) Options: (SIP/5001|45|tT)
    -- Called 5001
    -- SIP/5001-09895558 is ringing
    -- SIP/5001-09895558 answered SIP/5000-0989e978

regards,
Tias

[SARK devs]

Hmm,

Call looks normal.

What does /etc/init.d/masq status show you?

Also start tethereal again but without filtering the host.... like this

tethereal -R sip -i eth0

Run your call again and see if any RTP packets are flowing.

S