[Home]

Summary:ASTERISK-18066: Attended transfert with sendrpid=yes and directedmedia=yes with aastra phone, return 500 error and not works
Reporter:Bernard Merindol (bernard merindol)Labels:
Date Opened:2011-06-25 10:55:22Date Closed:2015-03-15 00:01:35
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.4 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Linux Ubuntu 10.04LTS. With Aastra phoneAttachments:( 0) bug-transfer-aastra.cap
( 1) chan_sip.c.patch
( 2) full.txt
( 3) sip.h.patch
Description:When use Attended transfer with sendrpid=yes, diretmedia=yes on 3 AAstra SIP phone (67XX). Asterisk send 2 re-invite with out wait the answer for the first re-invite. AAstra phone Answer 500 error on second re-invite.

The first re-invite is to finish the refer and bridge directly two audio (directmedia=yes), second re-invite is to change de P_Asserted-Identity with the name of first phone.

See the log:
The first re-invite
[Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.102:5060:
INVITE sip:1000@192.168.169.102:5060;transport=udp SIP/2.0^M
Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK7b03c887;rport^M
Max-Forwards: 70^M
From: <sip:1001@192.168.169.60:5060;user=phone>;tag=as22755b3e^M
To: "BME" <sip:1000@192.168.169.60:5060>;tag=a2e37c0386^M
Contact: <sip:1001@192.168.169.60:5060>^M
Call-ID: 2c063ea24067f43a^M
CSeq: 104 INVITE^M
User-Agent: FPBX-2.9.0(1.8.4.2)^M
Require: timer^M
Session-Expires: 900;refresher=uas^M
Min-SE: 90^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
X-asterisk-Info: SIP re-invite (External RTP bridge)^M
P-Asserted-Identity: "Cedric Autier" <sip:1001@192.168.169.60:5060>^M
Content-Type: application/sdp^M
Content-Length: 239^M
^M
v=0^M
o=root 191191818 191191821 IN IP4 192.168.169.100^M
s=Asterisk PBX 1.8.4.2^M
c=IN IP4 192.168.169.100^M
t=0 0^M
m=audio 8000 RTP/AVP 8 101^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=sendrecv^M


The second re-invite:
[Jun 23 16:57:37] VERBOSE[2041] chan_sip.c: [Jun 23 16:57:37] Reliably Transmitting (NAT) to 192.168.169.102:5060:
INVITE sip:1000@192.168.169.102:5060;transport=udp SIP/2.0^M
Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK3f57ba26;rport^M
Max-Forwards: 70^M
From: <sip:1001@192.168.169.60:5060;user=phone>;tag=as22755b3e^M
To: "BME" <sip:1000@192.168.169.60:5060>;tag=a2e37c0386^M
Contact: <sip:1001@192.168.169.60:5060>^M
Call-ID: 2c063ea24067f43a^M
CSeq: 105 INVITE^M
User-Agent: FPBX-2.9.0(1.8.4.2)^M
Require: timer^M
Session-Expires: 900;refresher=uas^M
Min-SE: 90^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
P-Asserted-Identity: "TCE" <sip:1002@192.168.169.60:5060>^M
Content-Type: application/sdp^M
Content-Length: 239^M
^M
v=0^M
o=root 191191818 191191822 IN IP4 192.168.169.100^M
s=Asterisk PBX 1.8.4.2^M
c=IN IP4 192.168.169.100^M
t=0 0^M
m=audio 8000 RTP/AVP 8 101^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=sendrecv^M


The answer from Aastra:
Jun 23 16:57:37] VERBOSE[1540] chan_sip.c: [Jun 23 16:57:37]
<--- SIP read from UDP:192.168.169.102:5060 --->
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.169.60:5060;branch=z9hG4bK3f57ba26;rport=5060;received=192.168.169.60
From: <sip:1001@192.168.169.60:5060;user=phone>;tag=as22755b3e
To: "BME" <sip:1000@192.168.169.60:5060>;tag=a2e37c0386
Call-ID: 2c063ea24067f43a
CSeq: 105 INVITE
Retry-After: 3
Server: Aastra 6739i/3.2.2.41
Content-Length: 0


After the call is cancelled by Asterisk.

If not use sendrpid=yes the transfert works fine.


Comments:By: Bernard Merindol (bernard merindol) 2011-06-25 10:59:25.424-0500

Pcap file with SIP frame.

See Frame 72 and 74.
By: Bernard Merindol (bernard merindol) 2011-06-25 11:12:51.368-0500

The astrerisk log.

By: Bernard Merindol (bernard merindol) 2011-08-11 11:06:39.087-0500

Same problem on 1.8.5

No news ?

Best regards

By: Bernard Merindol (bernard merindol) 2012-04-27 01:59:27.503-0500

Hi All,

I hav epatched Asterisk in version 1.8.11 to resolve this issue. My patch works fine with sendrpid=pai, directmedia=yes with Aastra, Thomson en snom
I have added new state in channels/sip/include/sip.h (INV_REINVITE  = 8) and test this state in chan_sip.c

I have downloades this two patch (chan_sip.c.patch and sip.h.patch)

Best regards
Bernard
By: Bernard Merindol (bernard merindol) 2012-07-26 15:18:00.633-0500

From Asterisk version 1.8.13 all works wit out patch.

For me this issue is closed.

Best regards
Bernard