From: Patrick McHardy <kaber@trash.net>
To: Philip Prindeville <philipp_subx@redfish-solutions.com>
Cc: netfilter-devel@vger.kernel.org
Subject: Re: More nf_conntrack_sip questions
Date: Fri, 05 Dec 2008 17:21:01 +0100 [thread overview]
Message-ID: <493954ED.9070002@trash.net> (raw)
In-Reply-To: <4935B885.8030107@redfish-solutions.com>
Philip Prindeville wrote:
> I did a little investigation into my one-way voice issue, and noticed
> that if I don't do voice-menus (i.e. where the Asterisk box itself
> generates the first outbound INVITE, then passes-through the 2nd INVITE
> once a handset picks up) then I get two-way voice (i.e. with sending the
> call directly to the phone). (In this topology, my Asterisk box is also
> my firewall/NATting router...)
>
> If I enable the voice menus in the inbound dialplan, however, it can
> hear the voice menus, but not the called-party when they pick up their
> phone (extension).
>
> So someone (either the SIP conntrack module on the Asterisk border
> firewall or else the SBC at the ILEC) is failing to look into the 2nd
> INVITE (i.e. we're not rewriting it properly as it goes by, or the SBC
> is failing to see it).
What module options are you using for the SIP helper and how is call
setup in asterisk configured (directrtpsetup, canreinvite, ...)?
next prev parent reply other threads:[~2008-12-05 16:21 UTC|newest]
Thread overview: 14+ messages / expand[flat|nested] mbox.gz Atom feed top
2008-12-02 22:36 More nf_conntrack_sip questions Philip Prindeville
2008-12-05 16:21 ` Patrick McHardy [this message]
2008-12-05 18:00 ` Philip Prindeville
2008-12-05 18:07 ` Patrick McHardy
2008-12-05 18:08 ` Patrick McHardy
2008-12-05 18:10 ` Philip Prindeville
2008-12-05 18:13 ` Patrick McHardy
2008-12-05 18:16 ` Philip Prindeville
2008-12-05 18:19 ` Patrick McHardy
2008-12-05 18:32 ` Philip Prindeville
2008-12-05 18:39 ` Patrick McHardy
2008-12-07 0:31 ` Philip Prindeville
2008-12-07 16:06 ` Patrick McHardy
2008-12-08 0:28 ` Philip Prindeville
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