[Home]

Summary:ASTERISK-04446: a=silenceSupp:off - - - -
Reporter:Alberto Fernandez (derkommissar)Labels:
Date Opened:2005-06-20 18:31:17Date Closed:2011-06-07 14:10:30
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:For some equipment if this is included the call would not complete and return a 500 Internal server error (IE: Huaweii SoftX3000). I was able to modify chan_sip.c and take it off with a comment, but this affected chan_sip globally and made it incompatible with other equipment (IE: Cisco AS5350). I would like to see this parametisable in sip.conf or somewhere of that sort.


****** ADDITIONAL INFORMATION ******

with a=silenceSupp:off - - - -

INVITE sip:59322580877@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4a670000
From: "2100014" <sip:2100014@XXX.XXX.XXX.XXX>;tag=as3cadd8a0
To: <sip:59322580877@XXX.XXX.XXX.XXX>
Contact: <sip:2100014@XXX.XXX.XXX.XXX>
Call-ID: 3768e14e6bff4c37737911de33f51e3a@XXX.XXX.XXX.XXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 20 Jun 2005 17:03:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 216
                                                                                                                                                           
v=0
o=root 1474 1474 IN IP4 XXX.XXX.XXX.XXX
s=session
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 12774 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
                                                                                                                                                           
---
   -- Called 59322580877@XXX.XXX.XXX.XXX
Wholesale*CLI>
<-- SIP read from XXX.XXX.XXX.XXX:5060:
SIP/2.0 100 Trying
From: "2100014" <sip:2100014@XXX.XXX.XXX.XXX>;tag=as3cadd8a0
To: <sip:59322580877@XXX.XXX.XXX.XXX>
CSeq: 102 INVITE
Call-ID: 3768e14e6bff4c37737911de33f51e3a@XXX.XXX.XXX.XXX
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4a670000
Content-Length: 0
                                                                                                                                                           
                                                                                                                                                           
--- (7 headers 0 lines)---
                                                                                                                                                           
<-- SIP read from XXX.XXX.XXX.XXX:5060:
SIP/2.0 500 Server Internal Error
From: "2100014" <sip:2100014@XXX.XXX.XXX.XXX>;tag=as3cadd8a0
To: <sip:59322580877@XXX.XXX.XXX.XXX>;tag=e702ffa8
CSeq: 102 INVITE
Call-ID: 3768e14e6bff4c37737911de33f51e3a@XXX.XXX.XXX.XXX
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4a670000
Content-Length: 0
                                                                                                                                                           
                                                                                                                                                           
--- (7 headers 0 lines)---
   -- Got SIP response 500 "Server Internal Error" back from XXX.XXX.XXX.XXX
Transmitting (no NAT) to XXX.XXX.XXX.XXX:5060:
ACK sip:59322580877@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4a670000
From: "2100014" <sip:2100014@XXX.XXX.XXX.XXX>;tag=as3cadd8a0
To: <sip:59322580877@XXX.XXX.XXX.XXX>;tag=e702ffa8
Contact: <sip:2100014@XXX.XXX.XXX.XXX>
Call-ID: 3768e14e6bff4c37737911de33f51e3a@XXX.XXX.XXX.XXX
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0






==-===-===-===-===-===-===-===-===-===-===-===-===-===-===-===-




without a=silenceSupp:off - - - -

INVITE sip:59322580877@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4a670000
From: "2100014" <sip:2100014@XXX.XXX.XXX.XXX>;tag=as3cadd8a0
To: <sip:59322580877@XXX.XXX.XXX.XXX>
Contact: <sip:2100014@XXX.XXX.XXX.XXX>
Call-ID: 3768e14e6bff4c37737911de33f51e3a@XXX.XXX.XXX.XXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 20 Jun 2005 17:03:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 216
                                                                                                                                                           
v=0
o=root 1474 1474 IN IP4 XXX.XXX.XXX.XXX
s=session
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 12774 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
                                                                                                                     
---
   -- Called 59322580877@XXX.XXX.XXX.XXX
Wholesale*CLI>
<-- SIP read from XXX.XXX.XXX.XXX:5060:
SIP/2.0 100 Trying
From: "2100014" <sip:2100014@XXX.XXX.XXX.XXX>;tag=as3cadd8a0
To: <sip:59322580877@XXX.XXX.XXX.XXX>
CSeq: 102 INVITE
Call-ID: 3768e14e6bff4c37737911de33f51e3a@XXX.XXX.XXX.XXX
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4a670000
Content-Length: 0
Comments:By: Kevin P. Fleming (kpfleming) 2005-06-20 19:08:17

Asterisk's behavior is fully compliant with RFC 3108, which specifies the silenceSupp attribute and the '-' parameter values. If the other device cannot accept this SDP attribute, it is non-conformant with the RFC.