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Whenever I call someone using my workplace's Asterisk VOIP server there's a 2-3 second delay before I can hear them so I always miss them saying "hello". This happens with any SIP client running on windows, Mac, or linux. After that the call goes fine. It only happens for outgoing calls - the audio for incoming calls is fine.

Is this likely to be something broken on our work server or is it something most or all Asterisk users have to put up with?

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I've never noticed this myself - to help narrow down possible causes, is your system IP-only (outgoing calls over a SIP trunk) or a hybrid (outgoing calls via Zaptel/DAHDI cards)? – voretaq7 Feb 5 '10 at 4:01
up vote 3 down vote accepted

I've used many versions of asterisk, with several different VoIP providers (SIP and one or two IAX2), physical cards on analog lines (Sangoma/Digium/Pika), and many different PC-based and hardphones (Aastra/Polycom/Sipura/Linksys/Snom/Mitel) and have never had that problem.

It's almost certainly your PSTN gateway. If it's an analog card, there's probably an extra wait at the end of the dial string, or a delay setting specific to the manufacturer. These cards mute the line while dialing, and that's probably what is happening here.

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thanks, I'll get our asterisk guy to check this out – Joe Mahoney Feb 8 '10 at 20:07

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