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Summary:ASTERISK-14779: [patch] caller id number is empty
Reporter:Elazar Broad (ebroad)Labels:
Date Opened:2009-09-06 11:59:09Date Closed:2009-09-08 09:29:24
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) blank_cidv2.patch
( 1) no-cid.log
( 2) parse_uri_fix.diff
Description:Calling number is empty(defaults to default_callerid). See

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

Relevant sip debug, full debug attached:

<--- SIP read from UDP:10.200.24.2:1398 --->
INVITE sip:201@10.200.50.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.24.2:1398;branch=z9hG4bK-d8754z-14607d58284fa44a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.200.24.2:1398;rinstance=888d1593b17d8919>
To: <sip:201@10.200.50.8:5060>
From: "3CX Phone"<sip:202@10.200.50.8:5060>;tag=8d3af519
Call-ID: YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXVoipPhone 3.1.8571.0
Content-Length: 278

v=0
o=3cxVCE 164534160 395114070 IN IP4 10.200.24.2
s=3cxVCE Audio Call
c=IN IP4 10.200.24.2
t=0 0
m=audio 40000 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] --- (13 headers 13 lines) ---
[Sep  6 12:49:27] DEBUG[2789] acl.c: Found IP address for this socket
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.200.50.8:5060
[Sep  6 12:49:27] VERBOSE[2789] netsock.c: [Sep  6 12:49:27]   == Using UDPTL TOS bits 184
[Sep  6 12:49:27] VERBOSE[2789] netsock.c: [Sep  6 12:49:27]   == Using UDPTL CoS mark 5
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting NAT on UDPTL to Off
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Allocating new SIP dialog for YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ. - INVITE (No RTP)
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Begin: parsing SIP "Supported: replaces"
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Found SIP option: -replaces-
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Matched SIP option: replaces
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Sending to 10.200.24.2 : 1398 (no NAT)
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Initializing initreq for method INVITE - callid YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Using INVITE request as basis request - YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found peer '202' for '10.200.50.8' from 10.200.24.2:1398
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting NAT on UDPTL to Off
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27]
<--- Reliably Transmitting (no NAT) to 10.200.24.2:1398 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.24.2:1398;branch=z9hG4bK-d8754z-14607d58284fa44a-1---d8754z-;rport;received=10.200.24.2
From: "3CX Phone"<sip:202@10.200.50.8:5060>;tag=8d3af519
To: <sip:201@10.200.50.8:5060>;tag=as0e83c859
Call-ID: YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="bhn-pbx01.broadhn.net", nonce="02ec816a"
Content-Length: 0


<------------>
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.200.24.2:1398
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Scheduling destruction of SIP dialog 'YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.' in 6848 ms (Method: INVITE)
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27]
<--- SIP read from UDP:10.200.24.2:1398 --->
ACK sip:201@10.200.50.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.24.2:1398;branch=z9hG4bK-d8754z-14607d58284fa44a-1---d8754z-;rport
Max-Forwards: 70
To: <sip:201@10.200.50.8:5060>;tag=as0e83c859
From: "3CX Phone"<sip:202@10.200.50.8:5060>;tag=8d3af519
Call-ID: YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
CSeq: 1 ACK
Content-Length: 0


<------------->
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] --- (8 headers 0 lines) ---
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Stopping retransmission on 'YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.' of Response 1: Match Found
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27]
<--- SIP read from UDP:10.200.24.2:1398 --->
INVITE sip:201@10.200.50.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.24.2:1398;branch=z9hG4bK-d8754z-bc6d745c7f575031-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.200.24.2:1398;rinstance=888d1593b17d8919>
To: <sip:201@10.200.50.8:5060>
From: "3CX Phone"<sip:202@10.200.50.8:5060>;tag=8d3af519
Call-ID: YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXVoipPhone 3.1.8571.0
Authorization: Digest username="202",realm="bhn-pbx01.broadhn.net",nonce="02ec816a",uri="sip:201@10.200.50.8:5060",response="5ea2cea8ea91193c21067a9caf267ce3",algorithm=MD5
Content-Length: 278

v=0
o=3cxVCE 164534160 395114070 IN IP4 10.200.24.2
s=3cxVCE Audio Call
c=IN IP4 10.200.24.2
t=0 0
m=audio 40000 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] --- (14 headers 13 lines) ---
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Sending to 10.200.24.2 : 1398 (no NAT)
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Initializing initreq for method INVITE - callid YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Using INVITE request as basis request - YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found peer '202' for '202' from 10.200.24.2:1398
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting NAT on UDPTL to Off
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xc54b48'
[Sep  6 12:49:27] DEBUG[2789] res_rtp_asterisk.c: Allocated port 13768 for RTP instance '0xc54b48'
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: RTP instance '0xc54b48' is setup and ready to go
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xc55ff8'
[Sep  6 12:49:27] DEBUG[2789] res_rtp_asterisk.c: Allocated port 19828 for RTP instance '0xc55ff8'
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: RTP instance '0xc55ff8' is setup and ready to go
[Sep  6 12:49:27] DEBUG[2789] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xc55ff8'
[Sep  6 12:49:27] DEBUG[2789] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xc54b48'
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting NAT on RTP to Off
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting NAT on VRTP to Off
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting NAT on UDPTL to Off
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found RTP audio format 0
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Setting payload 0 based on m type on 0x7b19afe32940
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found RTP audio format 8
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Setting payload 8 based on m type on 0x7b19afe32940
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found RTP audio format 3
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Setting payload 3 based on m type on 0x7b19afe32940
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found RTP audio format 101
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Setting payload 101 based on m type on 0x7b19afe32940
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Peer doesn't provide T.38 UDPTL
[Sep  6 12:49:27] DEBUG[2789] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc54b48'
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Peer audio RTP is at port 10.200.24.2:40000
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found audio description format PCMU for ID 0
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found audio description format PCMA for ID 8
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found audio description format GSM for ID 3
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Found audio description format telephone-event for ID 101
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Got unsupported a:fmtp:101 0-15 in SDP offer
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 4 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 2 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 1 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 32 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 32 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 128 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 8 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 4096 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 64 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 64 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 32 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 32 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 256 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 65536 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 262144 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 524288 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 1024 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 1048576 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 2097152 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 8192 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 1048576 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 4194304 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 67108864 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 134217728 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 512 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 2048 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 16 to 20
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Setting framing for 16384 to 20
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Incorporating payload 0 on 0x7b19afe32940
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Incorporating payload 3 on 0x7b19afe32940
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Incorporating payload 8 on 0x7b19afe32940
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Incorporating payload 101 on 0x7b19afe32940
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Capabilities: us - 0x3c187e (gsm|ulaw|alaw|g726|adpcm|slin|g726aal2|g722|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Copying payload 0 from 0x7b19afe32940 to 0xc54ba8
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Copying payload 3 from 0x7b19afe32940 to 0xc54ba8
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Copying payload 8 from 0x7b19afe32940 to 0xc54ba8
[Sep  6 12:49:27] DEBUG[2789] rtp_engine.c: Copying payload 101 from 0x7b19afe32940 to 0xc54ba8
[Sep  6 12:49:27] DEBUG[2789] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc54b48'
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Peer audio RTP is at port 10.200.24.2:40000
[Sep  6 12:49:27] DEBUG[2789] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xc55ff8'
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Peer video RTP is at port 10.200.24.2:17989
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw)
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Checking SIP call limits for device :202
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Updating call counter for incoming call
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Call from peer '202' is 1 out of 999
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] Looking for 201 in broadhn (domain 10.200.50.8)
[Sep  6 12:49:27] DEBUG[2782] chan_sip.c: Checking device state for peer 202
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: Changing state for SIP/202 - state 2 (In use)
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: device 'SIP/202' state '2'
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: *** Our native formats are 0x4 (ulaw)
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw)
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: *** Our capabilities are 0x3c187e (gsm|ulaw|alaw|g726|adpcm|slin|g726aal2|g722|h261|h263|h263p|h264)
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: This channel can handle video! HOLLYWOOD next!
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: build_route: Contact hop: <sip:202@10.200.24.2:1398;rinstance=888d1593b17d8919>
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] list_route: hop: <sip:202@10.200.24.2:1398;rinstance=888d1593b17d8919>
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Session timer started: 84 - YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: SIP/202-00c49be8: New call is still down.... Trying...
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27]
<--- Transmitting (no NAT) to 10.200.24.2:1398 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.200.24.2:1398;branch=z9hG4bK-d8754z-bc6d745c7f575031-1---d8754z-;rport;received=10.200.24.2
From: "3CX Phone"<sip:202@10.200.50.8:5060>;tag=8d3af519
To: <sip:201@10.200.50.8:5060>
Call-ID: YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.200.50.8>
Content-Length: 0


<------------>
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.200.24.2:1398
[Sep  6 12:49:27] DEBUG[2782] chan_sip.c: Checking device state for peer 202
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: Changing state for SIP/202 - state 2 (In use)
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: device 'SIP/202' state '2'
[Sep  6 12:49:27] DEBUG[2796] pbx.c: Launching 'Dial'
[Sep  6 12:49:27] VERBOSE[2796] pbx.c: [Sep  6 12:49:27]     -- Executing [201@broadhn:1] Dial("SIP/202-00c49be8", "SIP/201,27,RTKHrtkh") in new stack
[Sep  6 12:49:27] DEBUG[2782] chan_sip.c: Checking device state for peer 202
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: Changing state for SIP/202 - state 2 (In use)
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: device 'SIP/202' state '2'
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: SIP answering channel: SIP/202-00c49be8
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Audio is at 10.200.50.8 port 13768
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x4 (ulaw) to SDP
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: Found code 4 at payload 0 on 0xc54ba8
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x8 (alaw) to SDP
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: Found code 8 at payload 8 on 0xc54ba8
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x2 (gsm) to SDP
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: Found code 2 at payload 3 on 0xc54ba8
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding non-codec 0x1 (telephone-event) to SDP
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: Found code 1 at payload 101 on 0xc54ba8
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: -- Done with adding codecs to SDP
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw)
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27]
<--- Reliably Transmitting (no NAT) to 10.200.24.2:1398 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.24.2:1398;branch=z9hG4bK-d8754z-bc6d745c7f575031-1---d8754z-;rport;received=10.200.24.2
From: "3CX Phone"<sip:202@10.200.50.8:5060>;tag=8d3af519
To: <sip:201@10.200.50.8:5060>;tag=as6ecb5b9a
Call-ID: YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:201@10.200.50.8>
Content-Type: application/sdp
Content-Length: 302

v=0
o=asterisk 753731196 753731196 IN IP4 10.200.50.8
s=Asterisk PBX
c=IN IP4 10.200.50.8
t=0 0
m=audio 13768 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.200.24.2:1398
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Sep  6 12:49:27] VERBOSE[2796] netsock.c: [Sep  6 12:49:27]   == Using UDPTL TOS bits 184
[Sep  6 12:49:27] VERBOSE[2796] netsock.c: [Sep  6 12:49:27]   == Using UDPTL CoS mark 5
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Allocating new SIP dialog for 152734a73df713532ed24df649fc58ab@10.200.50.8 - INVITE (No RTP)
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xc55458'
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: Allocated port 17740 for RTP instance '0xc55458'
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: RTP instance '0xc55458' is setup and ready to go
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xc5d128'
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: Allocated port 17468 for RTP instance '0xc5d128'
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: RTP instance '0xc5d128' is setup and ready to go
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xc5d128'
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xc55458'
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Setting NAT on RTP to Off
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Setting NAT on VRTP to Off
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Setting NAT on UDPTL to Off
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Sep  6 12:49:27] DEBUG[2796] acl.c: Found IP address for this socket
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.200.50.8:5060
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: *** Our native formats are 0x4 (ulaw)
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: *** Joint capabilities are 0x4 (ulaw)
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: *** Our capabilities are 0x3c187e (gsm|ulaw|alaw|g726|adpcm|slin|g726aal2|g722|h261|h263|h263p|h264)
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw)
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: This channel can handle video! HOLLYWOOD next!
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: Seeded SDP of 'SIP/201-00c4b798' with that of 'SIP/202-00c49be8'
[Sep  6 12:49:27] DEBUG[2796] channel.c: Not copying variable DIALEDTIME.
[Sep  6 12:49:27] DEBUG[2796] channel.c: Not copying variable ANSWEREDTIME.
[Sep  6 12:49:27] DEBUG[2796] channel.c: Not copying variable DIALEDPEERNAME.
[Sep  6 12:49:27] DEBUG[2796] channel.c: Not copying variable DIALEDPEERNUMBER.
[Sep  6 12:49:27] DEBUG[2796] channel.c: Not copying variable DIALSTATUS.
[Sep  6 12:49:27] DEBUG[2796] channel.c: Not copying variable SIPCALLID.
[Sep  6 12:49:27] DEBUG[2796] channel.c: Not copying variable SIPDOMAIN.
[Sep  6 12:49:27] DEBUG[2796] channel.c: Not copying variable SIPURI.
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Outgoing Call for :201
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Updating call counter for outgoing call
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Call to peer '201' is 1 out of 999
[Sep  6 12:49:27] DEBUG[2782] chan_sip.c: Checking device state for peer 201
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: Changing state for SIP/201 - state 6 (Ringing)
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: device 'SIP/201' state '6'
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: This call needs video offers!
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: ** Our capability: 0x3c187e (gsm|ulaw|alaw|g726|adpcm|slin|g726aal2|g722|h261|h263|h263p|h264) Video flag: False Text flag: False
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Audio is at 10.200.50.8 port 17740
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Video is at 10.200.50.8 port 17468
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x4 (ulaw) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x8 (alaw) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x2 (gsm) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x40 (slin) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x1000 (g722) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x20 (adpcm) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x800 (g726) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding codec 0x10 (g726aal2) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding video codec 0x40000 (h261) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding video codec 0x80000 (h263) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding video codec 0x100000 (h263p) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding video codec 0x200000 (h264) to SDP
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Adding non-codec 0x1 (telephone-event) to SDP
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: -- Done with adding codecs to SDP
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Done building SDP. Settling with this capability: 0x3c187e (gsm|ulaw|alaw|g726|adpcm|slin|g726aal2|g722|h261|h263|h263p|h264)
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Initializing initreq for method INVITE - callid 1e8d78f85a04be081b7f4fc45183e36f@10.200.50.8
[Sep  6 12:49:27] VERBOSE[2796] chan_sip.c: [Sep  6 12:49:27] Reliably Transmitting (no NAT) to 10.200.24.2:3803:
INVITE sip:201@10.200.24.2:3803;rinstance=b483b1ebadd55590 SIP/2.0
Via: SIP/2.0/UDP 10.200.50.8:5060;branch=z9hG4bK0eac5901
Max-Forwards: 70
From: "3CX Phone" <sip:asterisk@10.200.50.8>;tag=as471e895f
To: <sip:201@10.200.24.2:3803;rinstance=b483b1ebadd55590>
Contact: <sip:asterisk@10.200.50.8>
Call-ID: 1e8d78f85a04be081b7f4fc45183e36f@10.200.50.8
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 06 Sep 2009 16:49:27 GMT
Session-Expires: 600
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 596

v=0
o=asterisk 395543531 395543531 IN IP4 10.200.50.8
s=Asterisk PBX
c=IN IP4 10.200.50.8
b=CT:8192
t=0 0
m=audio 17740 RTP/AVP 0 8 3 10 9 5 111 112 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:9 G722/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 17468 RTP/AVP 31 34 98 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
[Sep  6 12:49:27] DEBUG[2796] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.200.24.2:3803
[Sep  6 12:49:27] VERBOSE[2796] app_dial.c: [Sep  6 12:49:27]     -- Called 201
[Sep  6 12:49:27] DEBUG[2796] channel.c: Driver for channel 'SIP/202-00c49be8' does not support indication 3, emulating it
[Sep  6 12:49:27] DEBUG[2796] channel.c: Set channel SIP/202-00c49be8 to write format slin
[Sep  6 12:49:27] DEBUG[2796] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Sep  6 12:49:27] DEBUG[2796] channel.c: Set channel SIP/202-00c49be8 to write format ulaw
[Sep  6 12:49:27] DEBUG[2796] channel.c: Generator got voice, switching to phase locked mode
[Sep  6 12:49:27] DEBUG[2796] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: No remote address on RTP instance '0xc55458' so dropping frame
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: No remote address on RTP instance '0xc55458' so dropping frame
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27]
<--- SIP read from UDP:10.200.24.2:1398 --->
ACK sip:201@10.200.50.8 SIP/2.0
Via: SIP/2.0/UDP 10.200.24.2:1398;branch=z9hG4bK-d8754z-ed469e4c4836f058-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.200.24.2:1398;rinstance=888d1593b17d8919>
To: <sip:201@10.200.50.8:5060>;tag=as6ecb5b9a
From: "3CX Phone"<sip:202@10.200.50.8:5060>;tag=8d3af519
Call-ID: YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.
CSeq: 2 ACK
User-Agent: 3CXVoipPhone 3.1.8571.0
Authorization: Digest username="202",realm="bhn-pbx01.broadhn.net",nonce="02ec816a",uri="sip:201@10.200.50.8:5060",response="5ea2cea8ea91193c21067a9caf267ce3",algorithm=MD5
Content-Length: 0


<------------->
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] --- (11 headers 0 lines) ---
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: Stopping retransmission on 'YmI2NDk5MGY4ZTQ0MWYzOTEyYjRjZDE5ZmVhNDEyMWQ.' of Response 2: Match Found
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: No remote address on RTP instance '0xc55458' so dropping frame
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27]
<--- SIP read from UDP:10.200.24.2:3803 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.200.50.8:5060;branch=z9hG4bK0eac5901
Contact: <sip:201@10.200.24.2:3803;rinstance=b483b1ebadd55590>
To: <sip:201@10.200.24.2:3803;rinstance=b483b1ebadd55590>;tag=b4283c39
From: "3CX Phone"<sip:asterisk@10.200.50.8>;tag=as471e895f
Call-ID: 1e8d78f85a04be081b7f4fc45183e36f@10.200.50.8
CSeq: 102 INVITE
User-Agent: Pangolin v5.1, Build 08282009
Content-Length: 0


<------------->
[Sep  6 12:49:27] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:27] --- (9 headers 0 lines) ---
[Sep  6 12:49:27] DEBUG[2789] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1e8d78f85a04be081b7f4fc45183e36f@10.200.50.8' Request 102: Found
[Sep  6 12:49:27] DEBUG[2782] chan_sip.c: Checking device state for peer 201
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: Changing state for SIP/201 - state 6 (Ringing)
[Sep  6 12:49:27] DEBUG[2782] devicestate.c: device 'SIP/201' state '6'
[Sep  6 12:49:27] VERBOSE[2796] app_dial.c: [Sep  6 12:49:27]     -- SIP/201-00c4b798 is ringing
[Sep  6 12:49:27] DEBUG[2796] channel.c: Driver for channel 'SIP/202-00c49be8' does not support indication 3, emulating it
[Sep  6 12:49:27] DEBUG[2796] channel.c: Set channel SIP/202-00c49be8 to write format slin
[Sep  6 12:49:27] DEBUG[2796] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Sep  6 12:49:27] DEBUG[2796] channel.c: Generator got voice, switching to phase locked mode
[Sep  6 12:49:27] DEBUG[2796] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Sep  6 12:49:27] DEBUG[2796] rtp_engine.c: Found code 4 at payload 0 on 0xc54ba8
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[Sep  6 12:49:27] DEBUG[2796] res_rtp_asterisk.c: Created smoother: format: 4 ms: 20 len: 160
[Sep  6 12:49:38] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:38]
<--- SIP read from UDP:10.200.24.2:3803 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.50.8:5060;branch=z9hG4bK0eac5901
Contact: <sip:201@10.200.24.2:3803;rinstance=b483b1ebadd55590>
To: <sip:201@10.200.24.2:3803;rinstance=b483b1ebadd55590>;tag=b4283c39
From: "3CX Phone"<sip:asterisk@10.200.50.8>;tag=as471e895f
Call-ID: 1e8d78f85a04be081b7f4fc45183e36f@10.200.50.8
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Pangolin v5.1, Build 08282009
Content-Length: 300

v=0
o=- 6454183 6454183 IN IP4 10.200.24.2
s=http://www.portsip.com
c=IN IP4 10.200.24.2
t=0 0
m=audio 20802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 41062 RTP/AVP 99
a=fmtp:99 profile-level-id=42e015
a=rtpmap:99 H264/90000

<------------->
[Sep  6 12:49:38] VERBOSE[2789] chan_sip.c: [Sep  6 12:49:38] --- (12 headers 13 lines) ---
[Sep  6 12:49:38] DEBUG[2789] chan_sip.c: Acked pending invite 102
[Sep  6 12:49:38] DEBUG[2789] chan_sip.c: Stopping retransmission on '1e8d78f85a04be081b7f4fc45183e36f@10.200.50.8' of Request 102: Match Found
Comments:By: Elazar Broad (ebroad) 2009-09-06 12:01:57

Could be related to 0015830: channel variable ${CALLERID(num)} is empty
By: Elazar Broad (ebroad) 2009-09-06 17:38:51

The fix for this is pretty simple. The problem appeared at revision 215522( see http://lists.digium.com/pipermail/asterisk-commits/2009-September/036703.html ). Essentially the code that parses the scheme in parse_uri fails to strip off the ':' separating the scheme and the user@domain portion of the uri. This caused some later code that splits user:pass to set name to null, because the name was prepended with a ':'. The easy fix is to change 'url += l;' to 'url += (l + 1);' (which is what my patch does, pending approval), this automatically assumes that the scheme is followed by a ':', however, the cleaner way to do this would be to check for a trailing ':' when called(it is possible that sometime down the line, someone might call parse_uri with sip:,sips:) and append one if there is none. Additionally, this fixes '0015830: channel variable ${CALLERID(num)} is empty'.


By: Elazar Broad (ebroad) 2009-09-06 19:06:27

blank_cidv2.patch is the "cleaner" patch, it's a two liner which looks for the ':' and appends it if it doesn't exist.
By: Michiel van Baak (mvanbaak) 2009-09-07 00:48:28

assigned to dvossel because this is a regression introduced by the SIP uri parsing cleanup patch.
By: Digium Subversion (svnbot) 2009-09-08 09:27:36

Repository: asterisk
Revision: 216993

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r216993 | dvossel | 2009-09-08 09:27:35 -0500 (Tue, 08 Sep 2009) | 14 lines

caller id number empty

parse_uri was not being given the correct scheme's, as
a result, uri parsing did not parse the username correctly.
One of the side effects of this is an empty caller id.

(closes issue ASTERISK-14779)
Reported by: ebroad
Patches:
     blank_cidv2.patch uploaded by ebroad (license 878)
     parse_uri_fix.diff uploaded by dvossel (license 671)
Tested by: ebroad, dvossel


------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=216993
By: Digium Subversion (svnbot) 2009-09-08 09:28:22

Repository: asterisk
Revision: 216994

_U  branches/1.6.2/
U   branches/1.6.2/channels/chan_sip.c

------------------------------------------------------------------------
r216994 | dvossel | 2009-09-08 09:28:19 -0500 (Tue, 08 Sep 2009) | 20 lines

Merged revisions 216993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
 r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
 
 caller id number empty
 
 parse_uri was not being given the correct scheme's, as
 a result, uri parsing did not parse the username correctly.
 One of the side effects of this is an empty caller id.
 
 (closes issue ASTERISK-14779)
 Reported by: ebroad
 Patches:
       blank_cidv2.patch uploaded by ebroad (license 878)
       parse_uri_fix.diff uploaded by dvossel (license 671)
 Tested by: ebroad, dvossel
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=216994
By: Digium Subversion (svnbot) 2009-09-08 09:28:54

Repository: asterisk
Revision: 216995

_U  branches/1.6.1/
U   branches/1.6.1/channels/chan_sip.c

------------------------------------------------------------------------
r216995 | dvossel | 2009-09-08 09:28:54 -0500 (Tue, 08 Sep 2009) | 20 lines

Merged revisions 216993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
 r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
 
 caller id number empty
 
 parse_uri was not being given the correct scheme's, as
 a result, uri parsing did not parse the username correctly.
 One of the side effects of this is an empty caller id.
 
 (closes issue ASTERISK-14779)
 Reported by: ebroad
 Patches:
       blank_cidv2.patch uploaded by ebroad (license 878)
       parse_uri_fix.diff uploaded by dvossel (license 671)
 Tested by: ebroad, dvossel
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=216995
By: Digium Subversion (svnbot) 2009-09-08 09:29:24

Repository: asterisk
Revision: 216996

_U  branches/1.6.0/
U   branches/1.6.0/channels/chan_sip.c

------------------------------------------------------------------------
r216996 | dvossel | 2009-09-08 09:29:23 -0500 (Tue, 08 Sep 2009) | 20 lines

Merged revisions 216993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
 r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
 
 caller id number empty
 
 parse_uri was not being given the correct scheme's, as
 a result, uri parsing did not parse the username correctly.
 One of the side effects of this is an empty caller id.
 
 (closes issue ASTERISK-14779)
 Reported by: ebroad
 Patches:
       blank_cidv2.patch uploaded by ebroad (license 878)
       parse_uri_fix.diff uploaded by dvossel (license 671)
 Tested by: ebroad, dvossel
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=216996