[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
-------------- next part --------------
A non-text attachment was scrubbed...
Name: digital_signature_gpg.asc
Type: application/pgp-signature
Size: 1118 bytes
Desc: Digital GPG signature (see http://martin-krafft.net/gpg/sig-policy/999bbcc4/current)
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170203/8bd7371e/attachment.pgp>
More information about the asterisk-users
mailing list