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Important thing to look at if you get one way audio problem with Asterisk 1.4.10 and FreePBX 2.3.0 Imprimer

We've had a weird problem where we though that there must be gremlins in the network or providers playing tricks on us. We've tried everything, reviewing our firewall rules, inserting log statements in our firewall rules, running tcpdump, analyzing STUN server traffic, everything !!


We have lost quite a bit of time looking in the wrong place (e.g. the network). We found out that Asterisk (version 1.4.10 and below) has some problems in filtering invalid characters in CIDNAME (CallerID Name). The problem could have been caused exclusively by our custom CIDNAME lookup application but it could affect other users, we did not take the time to look into it further.

Scenario :

Asterisk send the CIDNUMBER and the CIDNAME to our custom CIDNAME lookup application. If our application doesn't find detailed info related to the CIDNUMBER , it just sends back the CIDNAME name that Asterisk sent to us in the first place. We had to patch our application so it trims invalid chars from the CIDNAME that Asterisk sent us in the first place before returning it to Asterisk to solve our one way audio problem.

The one way audio problem was caused by Asterisk and/or devices sending invalid SIP packets.

Symptom :

Look for open double quotes without a closing one in your SIP packets by running asterisk -r and then sip debug.


SIP/2.0 200 OK
Via: SIP/2.0/UDP;branch=z9hG4bK4a2fdbb1;rport


To: <sip: Cette adresse email est protégée contre les robots des spammeurs, vous devez activer Javascript pour la voir. :61315>;tag=2a078efd6230ec21
Call-ID: Cette adresse email est protégée contre les robots des spammeurs, vous devez activer Javascript pour la voir.

User-Agent: Grandstream GXP2000
Warning: 399 "detected NAT type is symmetric NAT"
Contact: <sip: Cette adresse email est protégée contre les robots des spammeurs, vous devez activer Javascript pour la voir. :60429>
Supported: replaces, timer
Content-Length: 0

That One-Way audio problem with Asterisk 1.4.10 and FreePBX 2.3.0 that we spent so much time on had nothing to do with the network ! Just with stupid malformed headers ;-(





Commentaires / Comments (4)
where can I get the patch
1 lundi, 30 juin 2008 12:38

I am having the same issue it will be great if u let me know where I can get the patch

Re: where can I get the patch
2 lundi, 04 août 2008 22:53
There is no patch available from us. This article was merely a pointer to where to look at for the source of problems.

As stated in the article, we simply fixed our Caller ID lookup script to eliminate invalid characters. (We simply piped all called program output to the "strings" (program /usr/bin/strings).

As mentioned in the article, we could have introduced the problem ourselves but we wanted to point out the fact that asterisk should filter out invalid characters for any fields it sends out in SIP packets. So in the best world, our problem would never have occured.

This is a basic priciple that is applied in every robust piece of software. If a real life use case is needed as example, look at how hibernate filters out/escape invalid characters to avoid SQL injections.

So, the point of the article was to highlight that asterisk would need more robustness at handling CALLER_ID or any other fields provided by custom programs. Asterisk should not blindly forward those fields in the SIP packets it is sending out.

We haven't put any efforts into locating the spot in asterisk source in order to accomplished that, we apologize for that. ;-(
issue over ISDN
3 mardi, 11 novembre 2008 17:01
could this be the same issue over ISDN and not SIP using the Asterisk and FreePBX
Caller ID
4 mercredi, 02 décembre 2009 12:23
Clint Chapman
I had the same problem with one particular caller. I added an entry to the asterisk phonebook so the caller id would come from there instead of the other superfecta sources. Not a complete solution but it fixed the problem. Thanks for the tip!!

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