Re: More nf_conntrack_sip questions

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Patrick McHardy wrote:
Philip Prindeville wrote:
Patrick McHardy wrote:
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, ...)?


For the PSTN's switch:

 CanReinvite  : Yes

I vaguely recall some problem in the implementation of this feature,
something with missing bridging of the RTP streams that was still
necessary under some circumstances. Might be worth to try turning
it off.

What about the module options?

pbx*CLI> sip show settings
pbx*CLI>
Global Settings:
----------------
 SIP Port:               5060
 Bindaddress:            0.0.0.0
 Videosupport:           No
 AutoCreatePeer:         No
 Allow unknown access:   Yes
 Allow subscriptions:    Yes
 Allow overlap dialing:  No
 Promsic. redir:         No
 SIP domain support:     Yes
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Our auth realm          asterisk
 Realm. auth:            No
 Always auth rejects:    No
 Call limit peers only:  No
 Direct RTP setup:       No
 User Agent:             Asterisk PBX
 MWI checking interval:  10 secs
 Reg. context:           (not set)
 Caller ID:              asterisk
From: Domain: Record SIP history: Off
 Call Events:            Off
 IP ToS SIP:             CS3
 IP ToS RTP audio:       EF
 IP ToS RTP video:       AF41
 T38 fax pt UDPTL:       No
 RFC2833 Compensation:   No
 SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
 Codecs:                 0x6 (gsm|ulaw)
 Codec Order:            ulaw:20,gsm:20
 T1 minimum:             100
 Relax DTMF:             No
 Compact SIP headers:    No
 RTP Keepalive:          0 (Disabled)
 RTP Timeout:            0 (Disabled)
 RTP Hold Timeout:       0 (Disabled)
 MWI NOTIFY mime type:   application/simple-message-summary
 DNS SRV lookup:         Yes
 Pedantic SIP support:   No
 Reg. min duration       60 secs
 Reg. max duration:      3600 secs
 Reg. default duration:  120 secs
 Outbound reg. timeout:  20 secs
 Outbound reg. attempts: 0
 Notify ringing state:   Yes
 Notify hold state:      No
 SIP Transfer mode:      open
 Max Call Bitrate:       384 kbps
Auto-Framing: No
Default Settings:
-----------------
 Context:                INVALID
 Nat:                    RFC3581
 DTMF:                   rfc2833
 Qualify:                0
 Use ClientCode:         No
 Progress inband:        Never
 Language:               (Defaults to English)
 MOH Interpret:          default
MOH Suggest: Voice Mail Extension: asterisk

----
pbx*CLI>
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