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:
pbx*CLI> sip show peer sip_proxy
pbx*CLI>
* Name : sip_proxy
Secret : xxxx
MD5Secret : <Not set>
Context : ctc-incoming
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Nat : RFC3581
ACL : No
T38 pt UDPTL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
LastMsg : 0
ToHost : 66.232.80.9
Addr->IP : 66.232.80.9 Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username:
SIP Options : (none)
Codecs : 0x6 (gsm|ulaw)
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
Status : Unmonitored
Useragent :
Reg. Contact :
pbx*CLI>
For the phone:
pbx*CLI> sip show peer office_1
pbx*CLI>
* Name : office_1
Secret : <Set>
MD5Secret : <Not set>
Context : redfish-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 112@redfish,xxxx
VM Extension : voicemail
LastMsgsSent : 0/0
Call limit : 0
Dynamic : Yes
Callerid : "Redfish Solutions" <112>
MaxCallBR : 384 kbps
Expire : 1780
Insecure : no
Nat : Always
ACL : No
T38 pt UDPTL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 192.168.1.7 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: office_1
SIP Options : (none)
Codecs : 0x6 (gsm|ulaw)
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
Status : OK (9 ms)
Useragent : Linksys/SPA942-5.1.15(a)
Reg. Contact : sip:office_1@xxxxxxxxxxx:5060
pbx*CLI>
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