[asterisk-users] Problem with rport (CGNAT) going from Linux kernel 3.16 to 4.9

martin f krafft madduck at madduck.net
Thu Feb 2 21:20:02 CST 2017


Hello,

I operate an Asterisk server (v11.13.1) on Debian stable, and it's
rock-solid. The other day, however, I accidentally upgraded the
kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped
working.

Below you can find my analysis while running the 4.9.0 kernel. 888
is a simply Echo() extension. I am calling it from a phone behind
carrier-grade NAT ("mtvic-main"). The problem is that the Asterisk
server sends RTP to the 100.64.0.0/10 address I have on the internal
side of NAT, even though the Asterisk server correctly (?)
transports the actual socket on the outside via rport (cf. the 401
Unauth response).

Once I boot back into 3.16.0, it all works again. I didn't capture
any logs yet, but since audio works, I am led to believe that the
100.64.0.0/10 address is not being used.

Right now it works, but eventually, the kernel upgrade will be
required. It's possible that a newer Asterisk will work with the v4
kernel, but in any case I'd be interested in finding out the root of
the problem at hand.

Any hints appreciated. Thank you!


>>> sip.conf <<<
[general]
nat=auto_force_rport,auto_comedia

[mtvic-main]
md5secret=xxx
context=mtvic-in-main
callerid="Martin in windy Wellington <60>"
dtmfmode=rfc2833
context=from-office
type=friend
directmedia=no
host=dynamic
nat=force_rport,comedia

# sip show peer output below
>>> /sip.conf <<<



>>> debug output <<<
[Feb  2 08:35:24] <--- SIP read from UDP:219.88.239.74:43525 --->
[Feb  2 08:35:24] INVITE sip:888 at madduck.net;user=phone SIP/2.0
[Feb  2 08:35:24] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bK2c95e270486c659f91f1baa7712ebc80;rport
[Feb  2 08:35:24] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:24] To: <sip:888 at madduck.net;user=phone>
[Feb  2 08:35:24] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:24] CSeq: 2 INVITE
[Feb  2 08:35:24] Contact: <sip:mtvic-main at 100.64.45.19:5865>
[Feb  2 08:35:24] Max-Forwards: 70
[Feb  2 08:35:24] User-Agent: S685IP/022270000000
[Feb  2 08:35:24] Supported: replaces
[Feb  2 08:35:24] Allow-Events: message-summary, refer
[Feb  2 08:35:24] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
[Feb  2 08:35:24] Content-Type: application/sdp
[Feb  2 08:35:24] Content-Length: 375
[Feb  2 08:35:24] 
[Feb  2 08:35:24] v=0
[Feb  2 08:35:24] o=mtvic-main 8602 68 IN IP4 100.64.45.19
[Feb  2 08:35:24] s=Mapping
[Feb  2 08:35:24] c=IN IP4 100.64.45.19
[Feb  2 08:35:24] t=0 0
[Feb  2 08:35:24] m=audio 8602 RTP/AVP 9 8 0 96 97 2 18 101
[Feb  2 08:35:24] a=rtpmap:9 G722/8000
[Feb  2 08:35:24] a=rtpmap:8 PCMA/8000
[Feb  2 08:35:24] a=rtpmap:0 PCMU/8000
[Feb  2 08:35:24] a=rtpmap:96 G726-32/8000
[Feb  2 08:35:24] a=rtpmap:97 AAL2-G726-32/8000
[Feb  2 08:35:24] a=rtpmap:2 G726-32/8000
[Feb  2 08:35:24] a=rtpmap:18 G729/8000
[Feb  2 08:35:24] a=fmtp:18 annexb=no
[Feb  2 08:35:24] a=rtpmap:101 telephone-event/8000
[Feb  2 08:35:24] a=fmtp:101 0-16
[Feb  2 08:35:24] <------------->
[Feb  2 08:35:24] --- (14 headers 16 lines) ---
[Feb  2 08:35:24] Sending to 219.88.239.74:43525 (NAT)
[Feb  2 08:35:24] Sending to 219.88.239.74:43525 (NAT)
[Feb  2 08:35:24] Using INVITE request as basis request - 4239363066 at 192_168_15_112
[Feb  2 08:35:24] Found peer 'mtvic-main' for 'mtvic-main' from 219.88.239.74:43525
[Feb  2 08:35:24] 
[Feb  2 08:35:24] <--- Reliably Transmitting (NAT) to 219.88.239.74:43525 --->
[Feb  2 08:35:24] SIP/2.0 401 Unauthorized
[Feb  2 08:35:24] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bK2c95e270486c659f91f1baa7712ebc80;received=219.88.239.74;rport=43525
[Feb  2 08:35:24] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:24] To: <sip:888 at madduck.net;user=phone>;tag=as39e92fd2
[Feb  2 08:35:24] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:24] CSeq: 2 INVITE
[Feb  2 08:35:24] Server: Asterisk PBX
[Feb  2 08:35:24] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Feb  2 08:35:24] Supported: replaces, timer
[Feb  2 08:35:24] WWW-Authenticate: Digest algorithm=MD5, realm="madduck.net", nonce="2a4c925b"
[Feb  2 08:35:24] Content-Length: 0
[Feb  2 08:35:24] 
[Feb  2 08:35:24] 
[Feb  2 08:35:24] <------------>
[Feb  2 08:35:24] Scheduling destruction of SIP dialog '4239363066 at 192_168_15_112' in 32000 ms (Method: INVITE)
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <--- SIP read from UDP:219.88.239.74:43525 --->
[Feb  2 08:35:25] ACK sip:888 at madduck.net;user=phone SIP/2.0
[Feb  2 08:35:25] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bK2c95e270486c659f91f1baa7712ebc80;rport
[Feb  2 08:35:25] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:25] To: <sip:888 at madduck.net;user=phone>;tag=as39e92fd2
[Feb  2 08:35:25] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:25] CSeq: 2 ACK
[Feb  2 08:35:25] Contact: <sip:mtvic-main at 100.64.45.19:5865>
[Feb  2 08:35:25] Max-Forwards: 70
[Feb  2 08:35:25] User-Agent: S685IP/022270000000
[Feb  2 08:35:25] Content-Length: 0
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <------------->
[Feb  2 08:35:25] --- (10 headers 0 lines) ---
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <--- SIP read from UDP:219.88.239.74:43525 --->
[Feb  2 08:35:25] INVITE sip:888 at madduck.net;user=phone SIP/2.0
[Feb  2 08:35:25] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bK30820c8abce160b81ba532563001d99;rport
[Feb  2 08:35:25] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:25] To: <sip:888 at madduck.net;user=phone>
[Feb  2 08:35:25] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:25] CSeq: 3 INVITE
[Feb  2 08:35:25] Contact: <sip:mtvic-main at 100.64.45.19:5865>
[Feb  2 08:35:25] Authorization: Digest username="mtvic-main", realm="madduck.net", algorithm=MD5, uri="sip:888 at madduck.net;user=phone", nonce="2a4c925b", response="xxx"
[Feb  2 08:35:25] Max-Forwards: 70
[Feb  2 08:35:25] User-Agent: S685IP/022270000000
[Feb  2 08:35:25] Supported: replaces
[Feb  2 08:35:25] Allow-Events: message-summary, refer
[Feb  2 08:35:25] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
[Feb  2 08:35:25] Content-Type: application/sdp
[Feb  2 08:35:25] Content-Length: 375
[Feb  2 08:35:25] 
[Feb  2 08:35:25] v=0
[Feb  2 08:35:25] o=mtvic-main 8602 68 IN IP4 100.64.45.19  --HERE--
[Feb  2 08:35:25] s=Mapping
[Feb  2 08:35:25] c=IN IP4 100.64.45.19  --HERE--
[Feb  2 08:35:25] t=0 0
[Feb  2 08:35:25] m=audio 8602 RTP/AVP 9 8 0 96 97 2 18 101
[Feb  2 08:35:25] a=rtpmap:9 G722/8000
[Feb  2 08:35:25] a=rtpmap:8 PCMA/8000
[Feb  2 08:35:25] a=rtpmap:0 PCMU/8000
[Feb  2 08:35:25] a=rtpmap:96 G726-32/8000
[Feb  2 08:35:25] a=rtpmap:97 AAL2-G726-32/8000
[Feb  2 08:35:25] a=rtpmap:2 G726-32/8000
[Feb  2 08:35:25] a=rtpmap:18 G729/8000
[Feb  2 08:35:25] a=fmtp:18 annexb=no
[Feb  2 08:35:25] a=rtpmap:101 telephone-event/8000
[Feb  2 08:35:25] a=fmtp:101 0-16
[Feb  2 08:35:25] <------------->
[Feb  2 08:35:25] --- (15 headers 16 lines) ---
[Feb  2 08:35:25] Sending to 219.88.239.74:43525 (NAT)
[Feb  2 08:35:25] Using INVITE request as basis request - 4239363066 at 192_168_15_112
[Feb  2 08:35:25] Found peer 'mtvic-main' for 'mtvic-main' from 219.88.239.74:43525
[Feb  2 08:35:25]   == Using SIP RTP CoS mark 5
[Feb  2 08:35:25] Found RTP audio format 9
[Feb  2 08:35:25] Found RTP audio format 8
[Feb  2 08:35:25] Found RTP audio format 0
[Feb  2 08:35:25] Found RTP audio format 96
[Feb  2 08:35:25] Found RTP audio format 97
[Feb  2 08:35:25] Found RTP audio format 2
[Feb  2 08:35:25] Found RTP audio format 18
[Feb  2 08:35:25] Found RTP audio format 101
[Feb  2 08:35:25] Found audio description format G722 for ID 9
[Feb  2 08:35:25] Found audio description format PCMA for ID 8
[Feb  2 08:35:25] Found audio description format PCMU for ID 0
[Feb  2 08:35:25] Found audio description format G726-32 for ID 96
[Feb  2 08:35:25] Found audio description format AAL2-G726-32 for ID 97
[Feb  2 08:35:25] Found audio description format G726-32 for ID 2
[Feb  2 08:35:25] Found audio description format G729 for ID 18
[Feb  2 08:35:25] Found audio description format telephone-event for ID 101
[Feb  2 08:35:25] Capabilities: us - (ulaw|alaw|g726), peer - audio=(ulaw|alaw|g726|g729|g726aal2|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
[Feb  2 08:35:25] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Feb  2 08:35:25] Peer audio RTP is at port 100.64.45.19:8602
[Feb  2 08:35:25] Looking for 888 in mtvic-in-main (domain madduck.net)
[Feb  2 08:35:25] list_route: hop: <sip:mtvic-main at 100.64.45.19:5865>
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <--- Transmitting (NAT) to 219.88.239.74:43525 --->
[Feb  2 08:35:25] SIP/2.0 100 Trying
[Feb  2 08:35:25] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bK30820c8abce160b81ba532563001d99;received=219.88.239.74;rport=43525
[Feb  2 08:35:25] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:25] To: <sip:888 at madduck.net;user=phone>
[Feb  2 08:35:25] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:25] CSeq: 3 INVITE
[Feb  2 08:35:25] Server: Asterisk PBX
[Feb  2 08:35:25] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Feb  2 08:35:25] Supported: replaces, timer
[Feb  2 08:35:25] Contact: <sip:888 at 188.174.253.168:5060>
[Feb  2 08:35:25] Content-Length: 0
[Feb  2 08:35:25] 
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <------------>
[Feb  2 08:35:25]     -- Executing [888 at mtvic-in-main:1] Gosub("SIP/mtvic-main-00000010", "subDebugging,echo,1") in new stack
[Feb  2 08:35:25]     -- Executing [echo at subDebugging:1] NoOp("SIP/mtvic-main-00000010", "") in new stack
[Feb  2 08:35:25]     -- Executing [echo at subDebugging:2] Answer("SIP/mtvic-main-00000010", "") in new stack
[Feb  2 08:35:25] Audio is at 10454
[Feb  2 08:35:25] Adding codec 100004 (alaw) to SDP
[Feb  2 08:35:25] Adding codec 100003 (ulaw) to SDP
[Feb  2 08:35:25] Adding codec 100011 (g726) to SDP
[Feb  2 08:35:25] Adding non-codec 0x1 (telephone-event) to SDP
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <--- Reliably Transmitting (NAT) to 219.88.239.74:43525 --->
[Feb  2 08:35:25] SIP/2.0 200 OK
[Feb  2 08:35:25] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bK30820c8abce160b81ba532563001d99;received=219.88.239.74;rport=43525
[Feb  2 08:35:25] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:25] To: <sip:888 at madduck.net;user=phone>;tag=as28bdecb0
[Feb  2 08:35:25] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:25] CSeq: 3 INVITE
[Feb  2 08:35:25] Server: Asterisk PBX
[Feb  2 08:35:25] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Feb  2 08:35:25] Supported: replaces, timer
[Feb  2 08:35:25] Contact: <sip:888 at 188.174.253.168:5060>
[Feb  2 08:35:25] Content-Type: application/sdp
[Feb  2 08:35:25] Content-Length: 307
[Feb  2 08:35:25] 
[Feb  2 08:35:25] v=0
[Feb  2 08:35:25] o=root 1024396389 1024396389 IN IP4 188.174.253.168
[Feb  2 08:35:25] s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
[Feb  2 08:35:25] c=IN IP4 188.174.253.168
[Feb  2 08:35:25] t=0 0
[Feb  2 08:35:25] m=audio 10454 RTP/AVP 8 0 2 101
[Feb  2 08:35:25] a=rtpmap:8 PCMA/8000
[Feb  2 08:35:25] a=rtpmap:0 PCMU/8000
[Feb  2 08:35:25] a=rtpmap:2 G726-32/8000
[Feb  2 08:35:25] a=rtpmap:101 telephone-event/8000
[Feb  2 08:35:25] a=fmtp:101 0-16
[Feb  2 08:35:25] a=ptime:20
[Feb  2 08:35:25] a=sendrecv
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <------------>
[Feb  2 08:35:25] Retransmitting #1 (NAT) to 219.88.239.74:43525:
[Feb  2 08:35:25] SIP/2.0 200 OK
[Feb  2 08:35:25] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bK30820c8abce160b81ba532563001d99;received=219.88.239.74;rport=43525
[Feb  2 08:35:25] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:25] To: <sip:888 at madduck.net;user=phone>;tag=as28bdecb0
[Feb  2 08:35:25] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:25] CSeq: 3 INVITE
[Feb  2 08:35:25] Server: Asterisk PBX
[Feb  2 08:35:25] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Feb  2 08:35:25] Supported: replaces, timer
[Feb  2 08:35:25] Contact: <sip:888 at 188.174.253.168:5060>
[Feb  2 08:35:25] Content-Type: application/sdp
[Feb  2 08:35:25] Content-Length: 307
[Feb  2 08:35:25] 
[Feb  2 08:35:25] v=0
[Feb  2 08:35:25] o=root 1024396389 1024396389 IN IP4 188.174.253.168
[Feb  2 08:35:25] s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
[Feb  2 08:35:25] c=IN IP4 188.174.253.168
[Feb  2 08:35:25] t=0 0
[Feb  2 08:35:25] m=audio 10454 RTP/AVP 8 0 2 101
[Feb  2 08:35:25] a=rtpmap:8 PCMA/8000
[Feb  2 08:35:25] a=rtpmap:0 PCMU/8000
[Feb  2 08:35:25] a=rtpmap:2 G726-32/8000
[Feb  2 08:35:25] a=rtpmap:101 telephone-event/8000
[Feb  2 08:35:25] a=fmtp:101 0-16
[Feb  2 08:35:25] a=ptime:20
[Feb  2 08:35:25] a=sendrecv
[Feb  2 08:35:25] 
[Feb  2 08:35:25] ---
[Feb  2 08:35:25]     -- Executing [echo at subDebugging:3] Wait("SIP/mtvic-main-00000010", "2") in new stack
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <--- SIP read from UDP:219.88.239.74:43525 --->
[Feb  2 08:35:25] ACK sip:888 at 188.174.253.168:5060 SIP/2.0
[Feb  2 08:35:25] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bK7be5b674593511a99c805f11852c560;rport
[Feb  2 08:35:25] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:25] To: <sip:888 at madduck.net;user=phone>;tag=as28bdecb0
[Feb  2 08:35:25] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:25] CSeq: 3 ACK
[Feb  2 08:35:25] Contact: <sip:mtvic-main at 100.64.45.19:5865>
[Feb  2 08:35:25] Authorization: Digest username="mtvic-main", realm="madduck.net", algorithm=MD5, uri="sip:888 at madduck.net;user=phone", nonce="2a4c925b", response="9d6abe6c9f46c7801b50679e2721ab05"
[Feb  2 08:35:25] Max-Forwards: 70
[Feb  2 08:35:25] User-Agent: S685IP/022270000000
[Feb  2 08:35:25] Content-Length: 0
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <------------->
[Feb  2 08:35:25] --- (11 headers 0 lines) ---
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <--- SIP read from UDP:219.88.239.74:43525 --->
[Feb  2 08:35:25] 
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <------------->
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <--- SIP read from UDP:219.88.239.74:43525 --->
[Feb  2 08:35:25] ACK sip:888 at 188.174.253.168:5060 SIP/2.0
[Feb  2 08:35:25] Via: SIP/2.0/UDP 100.64.45.19:5865;branch=z9hG4bKf83c093eb7a06025aa4157c41f0e4304;rport
[Feb  2 08:35:25] From: "Penny & Martin / Wellington" <sip:mtvic-main at madduck.net>;tag=4132889942
[Feb  2 08:35:25] To: <sip:888 at madduck.net;user=phone>;tag=as28bdecb0
[Feb  2 08:35:25] Call-ID: 4239363066 at 192_168_15_112
[Feb  2 08:35:25] CSeq: 3 ACK
[Feb  2 08:35:25] Contact: <sip:mtvic-main at 100.64.45.19:5865>
[Feb  2 08:35:25] Authorization: Digest username="mtvic-main", realm="madduck.net", algorithm=MD5, uri="sip:888 at madduck.net;user=phone", nonce="2a4c925b", response="9d6abe6c9f46c7801b50679e2721ab05"
[Feb  2 08:35:25] Max-Forwards: 70
[Feb  2 08:35:25] User-Agent: S685IP/022270000000
[Feb  2 08:35:25] Content-Length: 0
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <------------->
[Feb  2 08:35:25] --- (11 headers 0 lines) ---
[Feb  2 08:35:27]     -- Executing [echo at subDebugging:4] Set("SIP/mtvic-main-00000010", "JITTERBUFFER(adaptive)=default") in new stack
[Feb  2 08:35:27]     -- Executing [echo at subDebugging:5] Playback("SIP/mtvic-main-00000010", "vm-from-phonenumber") in new stack
[Feb  2 08:35:27] Sent RTP packet to      100.64.45.19:8602 (type 08, seq 028319, ts 000160, len 000160)
[Feb  2 08:35:27]     -- <SIP/mtvic-main-00000010> Playing 'vm-from-phonenumber.slin' (language 'en')
[Feb  2 08:35:27] Sent RTP packet to      100.64.45.19:8602 (type 08, seq 028320, ts 000320, len 000160)

# Note the invalid target address of the RTP packet




swan*CLI> sip show peer mtvic-main
  * Name       : mtvic-main
  Description  : 
  Secret       : <Not set>
  MD5Secret    : <Set>
  Remote Secret: <Not set>
  Context      : mtvic-in-main
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     : en
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Named Callgr : 
  Nam. Pickupgr: 
  MOH Suggest  : 
  Mailbox      : 
  VM Extension : 99
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "Penny & Martin in windy Wellington" <60>
  MaxCallBR    : 384 kbps
  Expire       : 186
  Insecure     : no
  Force rport  : Yes
  Symmetric RTP: Yes
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 
  Addr->IP     : 219.88.239.74:43525
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: mtvic-main
  SIP Options  : (none)
  Codecs       : (ulaw|alaw|g726)
  Codec Order  : (alaw:20,ulaw:20,g726:20)
  Auto-Framing : No
  Status       : Unmonitored
  Useragent    : S685IP/022270000000
  Reg. Contact : sip:mtvic-main at 100.64.45.19:5865
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   : 
  Use Reason   : No
  Encryption   : No

-- 
@martinkrafft | http://madduck.net/ | http://two.sentenc.es/
 
a Hooloovoo is a superintelligent shade of the color blue.
            -- douglas adams, "the hitchhiker's guide to the galaxy"
 
spamtraps: madduck.bogus at madduck.net
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