no voice passes between 2 SIP phones call

[wiener]

Hello,
When I make a call from SIP agent '9999' to SIP agent '7649', the destination rings, when it answers, the Asterisk CLI shows
a message saying that call was answered, but I hear no voice. If I do a call from 7649 toward 9999, the call is good and voice passes through.
The CLI output (for the bad call) is:
== Using SIP RTP CoS mark 5
-- Executing [7649@phones:1] Dial("SIP/9999-00000069", "SIP/7649") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/7649
-- SIP/7649-0000006a is ringing
-- SIP/7649-0000006a answered SIP/9999-00000069
-- Remotely bridging SIP/9999-00000069 and SIP/7649-0000006a
== Spawn extension (phones, 7649, 1) exited non-zero on 'SIP/9999-00000069'

As I'm completely new to Asterisk, I do not know where to start looking for a solution.
I would be grateful if anybody could share a piece of knolage.

[david55]

Set directmedia to no.

This is not the most efficient option, but is likely to avoid problems. If it works, you then need to provide information about your firewall and NAT configuration, in order to work out how to re-enable it.

[wiener]

Set directmedia to no.

This is not the most efficient option, but is likely to avoid problems. If it works, you then need to provide information about your firewall and NAT configuration, in order to work out how to re-enable it.

Thanks forthe advice. I'll try it tomorrow morning (it's night now at my timezone).
Could you make thing even easier for me and advice where do I configure the 'directmedia' item?

[david55]

grep the sample configuration files.

[wiener]

grep the sample configuration files.

Well, I tried the directmedia option but it did not help. Still, dialing from 7649 toward 9999 gets a good call, and dialing from 9999 toward 7649 produces a call in which voice can not be heared.
I looked at the SIP messages that run between the Asterisk and the 2 SIP agents, and I see that in the good call, the 7649 sends an INVITE for 9999 (message sent to Asterisk), the Asterisk sends an INVITE to the 9999, and when done with the OK and ACK messages, it sends re-invite messages to both sides, so they send the RTP to each other.

In the bad call, the 9999 invites 7649 (message goes to Asterisk), the Asterisk sends an INVITE to the 7649 agent, and when done with the OK and ACK messages, the Asterisk doesn't send re-invite messages to the 2 parties.
When I looked at the messages, I saw that in the good call, the 9999 responds to the INVITE it got with a RINGING+Session Descrition message, while in the bad call, when the 7649 gets the INVITE, it responds with a plain RINGING message.
Could that be the origin of the failure?

Many thanks,
Moshe.

[david55]

Early media is more likely to cause problems that solve them.

[wiener]

Thanks for the help.
I have found the problem. It was a problem with my User Agent which had a bug in handling voice. Nothing to do with the SIP signaling. I appreciate all the effort!!

Early media is more likely to cause problems that solve them.