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Summary:ASTERISK-15220: chan_sip transforms %23 to # (UTF-8 issue) in Contact field
Reporter:Kristijan Vrban (vrban)Labels:
Date Opened:2009-11-26 08:41:40.000-0600Date Closed:2011-06-07 14:08:16
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:As you can see in the sip trace, chan_sip transforms the To number from %23xxx to #xxx in the Contact field. I dont like that, ok. But what does the RFC say to this? Is this the correct behavior?

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

<--- SIP read from 127.0.0.1:4456 --->
INVITE sip:%23xxx@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:4456;branch=z9hG4bK-20683-121-0
From: sipp <sip:sipp@127.0.1.1:4456>;tag=20683SIPpTag00121
To: sut <sip:%23xxx@127.0.0.1:5060>
Call-ID: 121-20683@127.0.1.1
CSeq: 1 INVITE
Contact: sip:sipp@127.0.1.1:4456
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length:   127

v=0
o=user1 53655765 2353687637 IN IP4 127.0.1.1
s=-
c=IN IP4 127.0.1.1
t=0 0
m=audio 5788 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<------------->
--- (11 headers 7 lines) ---
Sending to 127.0.1.1 : 4456 (no NAT)
Using INVITE request as basis request - 121-20683@127.0.1.1
Found peer 'sipp'
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 127.0.1.1:5788
Looking for %23xxx in K0000 (domain 127.0.0.1)
list_route: hop: <sip:sipp@127.0.1.1:4456>
testpbx01*CLI>
<--- Transmitting (no NAT) to 127.0.1.1:4456 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.1.1:4456;branch=z9hG4bK-20683-121-0;received=127.0.0.1
From: sipp <sip:sipp@127.0.1.1:4456>;tag=20683SIPpTag00121
To: sut <sip:%23xxx@127.0.0.1:5060>
Call-ID: 121-20683@127.0.1.1
CSeq: 1 INVITE
User-Agent: nfon nvoice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:#xxx@127.0.0.1>
Content-Length: 0
Comments:By: Leif Madsen (lmadsen) 2009-12-01 12:42:41.000-0600

What does your sip.conf look like? Specifically, are you using pedantic mode? Not sure if that would change anything here.

I'm also not entire sure if this is a feature request or not, as I'm not sure if the expected behaviour is to support UTF-8 in chan_sip or not. This might be a good question for the asterisk-dev mailing list.

Thanks!
By: Leif Madsen (lmadsen) 2009-12-21 10:15:25.000-0600

Closed due to lack of response from reporter.