No subject

For me this issue is closed.

Best regards
Bernard

> Attended transfert with sendrpid=yes and directedmedia=yes  with aastra phone, return 500 error and not works
> -------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-18066
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-18066
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.4
>         Environment: Linux Ubuntu 10.04LTS. With Aastra phone
>            Reporter: Bernard Merindol
>         Attachments: bug-transfer-aastra.cap, chan_sip.c.patch, full.txt, sip.h.patch
>
>
> 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 at 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 at 192.168.169.60:5060;user=phone>;tag=as22755b3e^M
> To: "BME" <sip:1000 at 192.168.169.60:5060>;tag=a2e37c0386^M
> Contact: <sip:1001 at 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 at 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 at 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 at 192.168.169.60:5060;user=phone>;tag=as22755b3e^M
> To: "BME" <sip:1000 at 192.168.169.60:5060>;tag=a2e37c0386^M
> Contact: <sip:1001 at 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 at 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 at 192.168.169.60:5060;user=phone>;tag=as22755b3e
> To: "BME" <sip:1000 at 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.

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