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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