[Asterisk-Dev] Fake channels stucked in Asterisk.

Fri Sep 17 02:27:41 MST 2004

Hello,

I believe this is potentially a bug.  I have a very strange problem that
is replicable when using Grandstream Budgetone 101 (Polycom and other phones
as well) as end-point.  
There are some fake channels stucked when I type "sip show channels".  First
let me describe my setup

Grandstream Budgetone 101 ----> Asterisk SIP (A) ------> Asterisk SIP &
ZAP (B)

This is what happens, when you use Grandstream Budgetone 101 Latest
firmware (5.11) speaker phone mode and make a call say 852-12345678 then
hangup immediately when the timer is 01sec on the phone.  What would
happen is Between GSBT101 and Asterisk SIP channel is hung up, but
when I type "sip show channels", the channels will never hangup.  It just
stucks and uses the rtp port.

This is a ngrep result showing that between the phone and Asterisk,
channel has hungup but between the two Asterisk, they haven't hung up yet.

Grandstream -> Asterisk (A)
 INVITE sip:*12345 at testsip.deltapath.com SIP/2.0..Via: SIP/2.0/UDP
192.168.11.2;branch=z9hG4bK810e5617b13c591d..From:
 "Kenny Lam" <sip:4006 at testsip.deltapath.com>;tag=6278425a9ac48e96..To:
<sip:*12345 at testsip.deltapath.com>..Contact
 : <sip:4006 at 192.168.11.2>..Call-ID:
88dcf1dbcf9a4c8b at 192.168.11.2..CSeq: 11089 INVITE..User-Agent:
Grandstream BT100 1.0
 .5.11..Max-Forwards: 70..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
application/sdp
 ..Content-Length: 214....v=0..o=4006 8000 8000 IN IP4
192.168.11.2..s=SIP Call..c=IN IP4 192.168.11.2..t=0 0..m=audio 50
 04 RTP/AVP 98 18 4..a=rtpmap:98 iLBC/8000..a=fmtp:98
mode=20..a=rtpmap:18 G729/8000..a=rtpmap:4 G723/8000..a=ptime:40..
#

Asterisk (A) to Grandstream
 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
192.168.11.2;branch=z9hG4bK810e5617b13c591d;rport=5
 060;received=202.64.51.68..From: "Kenny Lam"
<sip:4006 at testsip.deltapath.com>;tag=6278425a9ac48e96..To: <sip:*12345 at y
 ellowtail.deltapath.com>..Call-ID:
88dcf1dbcf9a4c8b at 192.168.11.2..CSeq: 11089 INVITE..Server: Sip EXpress
router (0.8.12
 -1rc6 (i386/linux))..Content-Length: 0..Warning: 392 202.83.198.5:5060
"Noisy feedback tells:  pid=23302 req_src_ip=202.
 64.51.68 req_src_port=5060 in_uri=sip:*12345 at testsip.deltapath.com
out_uri=sip:*12345 at 202.83.198.5:5070 via_cnt==1"..
 ..

Asterisk (A) to Grandstream
 SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.11.2;received=202.64.51.68;branch=z9hG4bK810e5617b13c591d..Record-Route:
<sip:*
 12345 at 202.83.198.5;ftag=6278425a9ac48e96;lr=on>..From: "Kenny Lam"
<sip:4006 at testsip.deltapath.com>;tag=6278425a9ac48
 e96..To: <sip:*12345 at testsip.deltapath.com>;tag=as618334c8..Call-ID:
88dcf1dbcf9a4c8b at 192.168.11.2..CSeq: 11089 INVIT
 E..User-Agent: SIP-By-Deltapath..Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER..Contact: <sip:*12345 at 202.83.198.5:5070
 >..Content-Type: application/sdp..Content-Length: 266....v=0..o=root
2572 2572 IN IP4 202.83.198.5..s=session..c=IN IP4
 202.83.198.5..t=0 0..m=audio 11238 RTP/AVP 18 98 0 101..a=rtpmap:18
G729/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:0 PCMU/80
 00..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - - -..

Grandstream to Asterisk (A)
 ACK sip:*12345 at 202.83.198.5:5070 SIP/2.0..Via: SIP/2.0/UDP
192.168.11.2;branch=z9hG4bK53d821b9ed5bf082..Route: <sip:*123
 45 at 202.83.198.5;ftag=6278425a9ac48e96;lr=on>..From: "Kenny Lam"
<sip:4006 at testsip.deltapath.com>;tag=6278425a9ac48e96
 ..To: <sip:*12345 at testsip.deltapath.com>;tag=as618334c8..Contact:
<sip:4006 at 192.168.11.2>..Call-ID: 88dcf1dbcf9a4c8b@
 192.168.11.2..CSeq: 11089 ACK..User-Agent: Grandstream BT100
1.0.5.11..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NO
 TIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0....

Grandstream to Asterisk (A)
 BYE sip:*12345 at 202.83.198.5:5070 SIP/2.0..Via: SIP/2.0/UDP
192.168.11.2;branch=z9hG4bKca89177a7ec72aa8..Route: <sip:*123
 45 at 202.83.198.5;ftag=6278425a9ac48e96;lr=on>..From: "Kenny Lam"
<sip:4006 at testsip.deltapath.com>;tag=6278425a9ac48e96
 ..To: <sip:*12345 at testsip.deltapath.com>;tag=as618334c8..Contact:
<sip:4006 at 192.168.11.2>..Call-ID: 88dcf1dbcf9a4c8b@
 192.168.11.2..CSeq: 11090 BYE..User-Agent: Grandstream BT100
1.0.5.11..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NO
 TIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0....

Asterisk (A) to Grandstream
 SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.11.2;received=202.64.51.68;branch=z9hG4bKca89177a7ec72aa8..From:
"Kenny Lam" <s
 ip:4006 at testsip.deltapath.com>;tag=6278425a9ac48e96..To:
<sip:*12345 at testsip.deltapath.com>;tag=as618334c8..Call-I
 D: 88dcf1dbcf9a4c8b at 192.168.11.2..CSeq: 11090 BYE..User-Agent:
SIP-By-Deltapath..Allow: INVITE, ACK, CANCEL, OPTIONS, BY
 E, REFER..Contact: <sip:*12345 at 202.83.198.5:5070>..Content-Length: 0....

----- Now after the phone hung up, Asterisk A still initiates a channel
with Asterisk B-----------

Asterisk (A) to Asterisk (B)
 INVITE sip:*12345 at 210.94.74.4 SIP/2.0..Via: SIP/2.0/UDP
210.94.74.5:5070;branch=z9hG4bK79b69dfd..From: "Kenny Lam" <s
 ip:4006 at 210.94.74.5:5070>;tag=as469cf124..To:
<sip:*12345 at 210.94.74.4>..Contact: <sip:4006 at 210.94.74.5:5070>..Call-I
 D: 761165010967f26137fb2b33579499ca at 210.94.74.5..CSeq: 102
INVITE..User-Agent: SIP-By-Deltapath..Date: Fri, 17 Sep 2004
  03:37:40 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER..Content-Type: application/sdp..Content-Length: 214....v=
 0..o=root 2572 2572 IN IP4 210.94.74.5..s=session..c=IN IP4
210.94.74.5..t=0 0..m=audio 16500 RTP/AVP 0 101..a=rtpmap:
 0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - - -..
#
U 210.94.74.5:5070 -> 210.94.74.4:5060
 CANCEL sip:*12345 at 210.94.74.4 SIP/2.0..Via: SIP/2.0/UDP
210.94.74.5:5070;branch=z9hG4bK79b69dfd..From: "Kenny Lam" <s
 ip:4006 at 210.94.74.5:5070>;tag=as469cf124..To:
<sip:*12345 at 210.94.74.4>..Contact: <sip:4006 at 210.94.74.5:5070>..Call-I
 D: 761165010967f26137fb2b33579499ca at 210.94.74.5..CSeq: 102
CANCEL..User-Agent: SIP-By-Deltapath..Content-Length: 0....
#####

Asterisk (B) to Asterisk (A)
 SIP/2.0 100 Trying..Via: SIP/2.0/UDP
210.94.74.5:5070;branch=z9hG4bK79b69dfd..From: "Kenny Lam"
<sip:4006 at 210.94.74.5:
 5070>;tag=as469cf124..To:
<sip:*12345 at 210.94.74.4>;tag=as4738bffe..Call-ID:
761165010967f26137fb2b33579499ca at 202.83.19
 8.5..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER..Contact: <sip:*12345@
 210.94.74.4>..Content-Length: 0....
###
Asterisk (B) to Asterisk (A)
 SIP/2.0 200 OK..Via: SIP/2.0/UDP
210.94.74.5:5070;branch=z9hG4bK79b69dfd..From: "Kenny Lam"
<sip:4006 at 210.94.74.5:5070
 >;tag=as469cf124..To:
<sip:*12345 at 210.94.74.4>;tag=as4738bffe..Call-ID:
761165010967f26137fb2b33579499ca at 210.94.74.5.
 .CSeq: 102 CANCEL..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER..Contact: <sip:*12345 at 202.
 83.198.4>..Content-Length: 0....

A channel is established between Asterisk A and B and remain there forever.

Asterisk (A) extension.conf
exten => _*.,1,Dial(SIP/${EXTEN}@asteriskb)
exten => _*.,2,Hangup

Asterisk (B) extension.conf
exten => _*.,1,Dial(Zap/g1d/${EXTEN:1})
exten => _*.,2,Hangup 

Kenny Lam
SIP Application Engineer
Deltapath Commerce & Technology Limited 
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