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Summary:ASTERISK-13416: SIP on hold problems
Reporter:klaus3000 (klaus3000)Labels:
Date Opened:2009-01-21 07:20:38.000-0600Date Closed:2009-01-23 12:11:09.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Hi!

I have found 2 problems when the SIP phone puts a call on hold (a SNOM phone which uses a=sendonly).

1. Although only RTP should be activated it looks like RTCP is deactivated too, as I got these error message on the console:

 RTCP SR transmission error, rtcp halted

2. When the call is on hold and I unplug the phone, I would think that the session is terminated after "rtpholdtimeout". Instead the session is termianted after rtptimeout.

see trace below: rtptimeout=15, rtpholdtimeout=25

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


Call is established....press HOLD button on phone


[Jan 21 14:06:15] VERBOSE[1511] logger.c:
<--- SIP read from 83.136.33.3:2051 --->
INVITE sip:01505641636@11.11.111.184:5160 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.147:2051;branch=z9hG4bK-ucts0k0x83kb;rport
From: <sip:+437206200730102@app.nxdomain.at:5160>;tag=5jh0ed98jf
To: <sip:01505641636@app.nxdomain.at:5160;user=phone>;tag=as6063d6ac
Call-ID: 3c2673852e63-ptr9sda6v20u@snom320-00041324009C
CSeq: 3 INVITE
Max-Forwards: 70
Contact: <sip:+437206200730102@10.10.0.147:2051;line=ohati0c1>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/6.5.18
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 339

v=0
o=root 2052104143 2052104144 IN IP4 10.10.0.147
s=call
c=IN IP4 10.10.0.147
t=0 0
m=audio 54400 RTP/AVP 8 0 9 3 18 4 101
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:9 g722/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendonly
<------------->
[Jan 21 14:06:15] VERBOSE[1511] logger.c: --- (18 headers 16 lines) ---
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Sending to 83.136.33.3 : 2051 (NAT)
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found RTP audio format 8
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found RTP audio format 0
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found RTP audio format 9
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found RTP audio format 3
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found RTP audio format 18
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found RTP audio format 4
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found RTP audio format 101
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Peer audio RTP is at port 10.10.0.147:54400
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found audio description format pcma for ID 8
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found audio description format pcmu for ID 0
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found audio description format g722 for ID 9
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found audio description format gsm for ID 3
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found audio description format g729 for ID 18
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found audio description format g723 for ID 4
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Found audio description format telephone-event for ID 101
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x110f (g723|gsm|ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Peer audio RTP is at port 10.10.0.147:54400
[Jan 21 14:06:15] VERBOSE[1511] logger.c:
<--- Transmitting (NAT) to 83.136.33.3:2051 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.0.147:2051;branch=z9hG4bK-ucts0k0x83kb;received=83.136.33.3;rport=2051
From: <sip:+437206200730102@app.nxdomain.at:5160>;tag=5jh0ed98jf
To: <sip:01505641636@app.nxdomain.at:5160;user=phone>;tag=as6063d6ac
Call-ID: 3c2673852e63-ptr9sda6v20u@snom320-00041324009C
CSeq: 3 INVITE
User-Agent: InnoSIP-app
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01505641636@11.11.111.184:5160>
Content-Length: 0


<------------>
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Audio is at 11.11.111.184 port 14838
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Adding codec 0x4 (ulaw) to SDP
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Adding codec 0x8 (alaw) to SDP
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Adding codec 0x2 (gsm) to SDP
[Jan 21 14:06:15] VERBOSE[1511] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan 21 14:06:15] VERBOSE[1511] logger.c:
<--- Reliably Transmitting (NAT) to 83.136.33.3:2051 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.147:2051;branch=z9hG4bK-ucts0k0x83kb;received=83.136.33.3;rport=2051
From: <sip:+437206200730102@app.nxdomain.at:5160>;tag=5jh0ed98jf
To: <sip:01505641636@app.nxdomain.at:5160;user=phone>;tag=as6063d6ac
Call-ID: 3c2673852e63-ptr9sda6v20u@snom320-00041324009C
CSeq: 3 INVITE
User-Agent: InnoSIP-app
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01505641636@11.11.111.184:5160>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1500 1502 IN IP4 11.11.111.184
s=session
c=IN IP4 11.11.111.184
t=0 0
m=audio 14838 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=recvonly

<------------>
[Jan 21 14:06:15] VERBOSE[4993] logger.c:     -- Started music on hold, class 'default', on SIP/gw-asterisk-08b7edf8
[Jan 21 14:06:16] VERBOSE[1511] logger.c:
<--- SIP read from 83.136.33.3:2051 --->
ACK sip:01505641636@11.11.111.184:5160 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.147:2051;branch=z9hG4bK-7rp55xbhi6ey;rport
From: <sip:+437206200730102@app.nxdomain.at:5160>;tag=5jh0ed98jf
To: <sip:01505641636@app.nxdomain.at:5160;user=phone>;tag=as6063d6ac
Call-ID: 3c2673852e63-ptr9sda6v20u@snom320-00041324009C
CSeq: 3 ACK
Max-Forwards: 70
Contact: <sip:+437206200730102@10.10.0.147:2051;line=ohati0c1>;flow-id=1
Content-Length: 0

<------------->
[Jan 21 14:06:16] VERBOSE[1511] logger.c: --- (9 headers 0 lines) ---

[Jan 21 14:06:25] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:06:30] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:06:35] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:06:40] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:06:45] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:06:50] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:06:55] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:00] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:05] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:10] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:16] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:21] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:26] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:31] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:36] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:41] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:46] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:51] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:07:56] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:01] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:06] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:11] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:16] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:21] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:26] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:31] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:36] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:41] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:46] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:51] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:08:56] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:01] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:06] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:11] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:16] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:21] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:26] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:31] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:36] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:41] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:46] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:51] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:09:56] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:01] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:06] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:11] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:16] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:21] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:26] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:31] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:36] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:41] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:46] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:51] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:10:56] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:01] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:06] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:11] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:16] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:21] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:26] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:32] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:37] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:42] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:47] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:52] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:11:57] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:02] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:12] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:17] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:22] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:27] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:32] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:37] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:42] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:47] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:52] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:12:57] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:02] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:07] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:12] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:17] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:22] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:27] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted



here I unplug the cable, so Asterisk does not receive any RTP (SNOM sends a dummy RTP packet every 5 seconds) or RTCP packets (every 5 seconds) from the phone


[Jan 21 14:13:32] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:37] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:42] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:47] VERBOSE[1511] logger.c: RTCP SR transmission error, rtcp halted
[Jan 21 14:13:51] NOTICE[1511] chan_sip.c: Disconnecting call 'SIP/+437206200730102-b6f83a00' for lack of RTP activity in 16 seconds
[Jan 21 14:13:51] VERBOSE[4993] logger.c:     -- Stopped music on hold on SIP/gw-asterisk-08b7edf8
[Jan 21 14:13:51] DEBUG[4993] chan_sip.c: Call to peer 'gw-asterisk' removed from call limit 100
[Jan 21 14:13:51] VERBOSE[4993] logger.c: Scheduling destruction of SIP dialog '585d77c16208df4e61dac9436a0810dc@11.11.111.184' in 32000 ms (Method: INVITE)
[Jan 21 14:13:51] DEBUG[4993] chan_sip.c: Strict routing enforced for session 585d77c16208df4e61dac9436a0810dc@11.11.111.184
[Jan 21 14:13:51] VERBOSE[4993] logger.c: set_destination: Parsing <sip:+431505641636@11.11.111.183> for address/port to send to
[Jan 21 14:13:51] VERBOSE[4993] logger.c: set_destination: set destination to 11.11.111.183, port 5060
[Jan 21 14:13:51] VERBOSE[4993] logger.c: Reliably Transmitting (no NAT) to 11.11.111.183:5060:
BYE sip:+431505641636@11.11.111.183 SIP/2.0
Via: SIP/2.0/UDP 11.11.111.184:5160;branch=z9hG4bK60d82a06;rport
From: "+437206200730102" <sip:+437206200730102@11.11.111.184:5160>;tag=as4c52a014
To: <sip:+431505641636@11.11.111.183>;tag=as1612af0a
Call-ID: 585d77c16208df4e61dac9436a0810dc@11.11.111.184
CSeq: 103 BYE
User-Agent: InnoSIP-app
Max-Forwards: 70
Content-Length: 0








sip show settings
ofis1*app*CLI>

Global Settings:
----------------
 SIP Port:               5160
 Bindaddress:            11.11.111.184
 Videosupport:           No
 AutoCreatePeer:         No
 Allow unknown access:   Yes
 Allow subscriptions:    Yes
 Allow overlap dialing:  No
 Promsic. redir:         No
 SIP domain support:     No
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Our auth realm          asterisk
 Realm. auth:            No
 Always auth rejects:    Yes
 Call limit peers only:  No
 Direct RTP setup:       No
 User Agent:             InnoSIP-app
 MWI checking interval:  10 secs
 Reg. context:           (not set)
 Caller ID:              asterisk
 From: Domain:
 Record SIP history:     Off
 Call Events:            Off
 IP ToS SIP:             none
 IP ToS RTP audio:       none
 IP ToS RTP video:       none
 T38 fax pt UDPTL:       No
 RFC2833 Compensation:   No
 SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
 Codecs:                 0x40e (gsm|ulaw|alaw|ilbc)
 Codec Order:            ulaw:20,alaw:20,ilbc:30,gsm:20
 T1 minimum:             100
 Relax DTMF:             No
 Compact SIP headers:    No
 RTP Keepalive:          0 (Disabled)
 RTP Timeout:            15
 RTP Hold Timeout:       25
 MWI NOTIFY mime type:   application/simple-message-summary
 DNS SRV lookup:         Yes
 Pedantic SIP support:   Yes
 Reg. min duration       60 secs
 Reg. max duration:      3600 secs
 Reg. default duration:  120 secs
 Outbound reg. timeout:  20 secs
 Outbound reg. attempts: 0
 Notify ringing state:   Yes
 Notify hold state:      No
 SIP Transfer mode:      open
 Max Call Bitrate:       384 kbps
 Auto-Framing:           No

Default Settings:
-----------------
 Context:                fromSipAnonym
 Nat:                    RFC3581
 DTMF:                   rfc2833
 Qualify:                0
 Use ClientCode:         No
 Progress inband:        Never
 Language:               de
 MOH Interpret:          default
 MOH Suggest:
 Voice Mail Extension:   asterisk

Comments:By: Joshua C. Colp (jcolp) 2009-01-22 13:09:21.000-0600

The second issue about the RTCP log message has been solved already, I am investigating the first.
By: Digium Subversion (svnbot) 2009-01-23 12:03:42.000-0600

Repository: asterisk
Revision: 170504

U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r170504 | file | 2009-01-23 12:03:41 -0600 (Fri, 23 Jan 2009) | 4 lines

Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold.
(closes issue ASTERISK-13416)
Reported by: klaus3000

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

http://svn.digium.com/view/asterisk?view=rev&revision=170504
By: Digium Subversion (svnbot) 2009-01-23 12:09:19.000-0600

Repository: asterisk
Revision: 170505

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r170505 | file | 2009-01-23 12:09:18 -0600 (Fri, 23 Jan 2009) | 11 lines

Merged revisions 170504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines
 
 Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold.
 (closes issue ASTERISK-13416)
 Reported by: klaus3000
........

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

http://svn.digium.com/view/asterisk?view=rev&revision=170505
By: Digium Subversion (svnbot) 2009-01-23 12:10:15.000-0600

Repository: asterisk
Revision: 170506

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

------------------------------------------------------------------------
r170506 | file | 2009-01-23 12:10:15 -0600 (Fri, 23 Jan 2009) | 18 lines

Merged revisions 170505 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) | 11 lines
 
 Merged revisions 170504 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines
   
   Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold.
   (closes issue ASTERISK-13416)
   Reported by: klaus3000
 ........
................

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

http://svn.digium.com/view/asterisk?view=rev&revision=170506
By: Digium Subversion (svnbot) 2009-01-23 12:11:09.000-0600

Repository: asterisk
Revision: 170507

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

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r170507 | file | 2009-01-23 12:11:09 -0600 (Fri, 23 Jan 2009) | 18 lines

Merged revisions 170505 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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 r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) | 11 lines
 
 Merged revisions 170504 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
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   r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines
   
   Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold.
   (closes issue ASTERISK-13416)
   Reported by: klaus3000
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http://svn.digium.com/view/asterisk?view=rev&revision=170507