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Summary:ASTERISK-18990: After upgrade from 1.6 to 1.8 one side audio in SPA941
Reporter:Badalian Vyacheslav (slavon)Labels:
Date Opened:2011-12-08 04:52:35.000-0600Date Closed:2012-01-23 13:03:02.000-0600
Priority:CriticalRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.7.1 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) ASTERISK-18990_v2.patch
( 1) ASTERISK-18990.patch
( 2) no.voice
( 3) sip-contact-route.diff
( 4) voice
Description:After upgrade from 1.6 to 1.8 SPA941 have:
If call from SPA - Normal
If call to SPA - One side audio

Attach 2 tcpdump!
Comments:By: Badalian Vyacheslav (slavon) 2011-12-08 04:52:55.218-0600

No voice TCPDUMP
By: Badalian Vyacheslav (slavon) 2011-12-08 04:53:10.428-0600

Have voice TCPDUMP
By: Badalian Vyacheslav (slavon) 2011-12-08 04:55:43.751-0600

[common](!)
type=friend
host=dynamic
context=office
subscribecontext=office
disallow=all
allow=alaw
allow=ulaw
allow=h264
allow=h263p
allow=h263
allow=h261
transport=udp,tcp
canreinvite=no


[111_office](common)
username=111_office
secret=XXXX
callerid=Alexandr Sidorov <111>


[112_office](common)
username=112_office
secret=XXXXX
nat=yes
callerid=Admins <112>


Connected to Asterisk 1.8.7.1 currently running on office (pid = 2958)
Verbosity is at least 3
office*CLI> sip show settings


Global Settings:
----------------
 UDP Bindaddress:        0.0.0.0:5060
 TCP SIP Bindaddress:    0.0.0.0:5060
 TLS SIP Bindaddress:    Disabled
 Videosupport:           Yes
 Textsupport:            No
 Ignore SDP sess. ver.:  No
 AutoCreate Peer:        No
 Match Auth Username:    No
 Allow unknown access:   No
 Allow subscriptions:    Yes
 Allow overlap dialing:  Yes
 Allow promisc. redir:   No
 Enable call counters:   Yes
 SIP domain support:     No
 Realm. auth:            No
 Our auth realm          asterisk
 Use domains as realms:  No
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Always auth rejects:    Yes
 Direct RTP setup:       No
 User Agent:             Asterisk PBX 1.8.7.1
 SDP Session Name:       Asterisk PBX 1.8.7.1
 SDP Owner Name:         root
 Reg. context:           (not set)
 Regexten on Qualify:    No
 Legacy userfield parse: No
 Caller ID:              asterisk
 From: Domain:
 Record SIP history:     Off
 Call Events:            Off
 Auth. Failure Events:   Off
 T.38 support:           No
 T.38 EC mode:           Unknown
 T.38 MaxDtgrm:          -1
 SIP realtime:           Disabled
 Qualify Freq :          5000 ms
 Q.850 Reason header:    No
 Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
 IP ToS SIP:             CS0
 IP ToS RTP audio:       CS0
 IP ToS RTP video:       CS0
 IP ToS RTP text:        CS0
 802.1p CoS SIP:         4
 802.1p CoS RTP audio:   5
 802.1p CoS RTP video:   6
 802.1p CoS RTP text:    5
 Jitterbuffer enabled:   No

Network Settings:
---------------------------
 SIP address remapping:  Disabled, no localnet list
 Externhost:             <none>
 Externaddr:             (null)
 Externrefresh:          10

Global Signalling Settings:
---------------------------
 Codecs:                 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
 Codec Order:            none
 Relax DTMF:             No
 RFC2833 Compensation:   No
 Symmetric RTP:          No
 Compact SIP headers:    No
 RTP Keepalive:          0 (Disabled)
 RTP Timeout:            0 (Disabled)
 RTP Hold Timeout:       0 (Disabled)
 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
   Include CID:          Yes
 Notify hold state:      Yes
 SIP Transfer mode:      open
 Max Call Bitrate:       384 kbps
 Auto-Framing:           No
 Outb. proxy:            <not set>
 Session Timers:         Accept
 Session Refresher:      uas
 Session Expires:        1800 secs
 Session Min-SE:         90 secs
 Timer T1:               500
 Timer T1 minimum:       100
 Timer B:                32000
 No premature media:     Yes
 Max forwards:           70

Default Settings:
-----------------
 Allowed transports:     UDP
 Outbound transport:     UDP
 Context:                IVR
 Force rport:            No
 DTMF:                   rfc2833
 Qualify:                2000
 Use ClientCode:         No
 Progress inband:        Never
 Language:               ru
 MOH Interpret:          default
 MOH Suggest:
 Voice Mail Extension:   asterisk

----
office*CLI> exit


By: Leif Madsen (lmadsen) 2011-12-14 14:12:23.095-0600

I believe this to be a configuration or networking issue. From what I'm seeing in the dumps, the one with voice is setting up appropriately, but the no.voice one almost looks like the other end isn't sending the information back to Asterisk that it requires.

Please provide a console trace and SIP debug from Asterisk per the issue guidelines.
By: Badalian Vyacheslav (slavon) 2011-12-15 08:57:15.570-0600

Ok. I try get it tomorrow or at morning.
Strange that if i rollback to 1.6 version (config does not change) - all is work.
By: Badalian Vyacheslav (slavon) 2011-12-27 07:56:21.089-0600

<--- SIP read from UDP:192.168.100.95:5060 --->
SIP/2.0 200 OK
To: <sip:112_office@192.168.100.95:5060>;tag=5215788f5c882b51i0
From: "Roman Solovev" <sip:104@192.168.100.1>;tag=as65cf4701
Call-ID: 62e5365a7010575e33cc540251bddab4@192.168.100.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK0a89d2fe
Contact: "Admins <112>" <sip:112_office@192.168.100.95:5060>
Server: Linksys/SPA941-5.1.8
Remote-Party-ID: "Admins <112>" <sip:112_office@192.168.100.1>;screen=yes;party=called
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 12309 12309 IN IP4 192.168.100.95
s=-
c=IN IP4 192.168.100.95
t=0 0
m=audio 16458 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
set_destination: Parsing <112> for address/port to send to
set_destination: set destination to 0.0.0.112:5060
Transmitting (NAT) to 192.168.100.95:5060:
ACK 112 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK1be6f5cd;rport
Max-Forwards: 70
From: "Roman Solovev" <sip:104@192.168.100.1>;tag=as65cf4701
To: <sip:112_office@192.168.100.95:5060>;tag=5215788f5c882b51i0
Contact: <sip:104@192.168.100.1:5060>
Call-ID: 62e5365a7010575e33cc540251bddab4@192.168.100.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0


---


As you see if in display name we have "Admins <112>" asterisk say:

set_destination: Parsing <112> for address/port to send to
set_destination: set destination to 0.0.0.112:5060

In 1.6 this was work!
By: Badalian Vyacheslav (slavon) 2011-12-27 07:56:36.246-0600

comment added
By: Matt Jordan (mjordan) 2011-12-30 16:46:08.151-0600

What happens if you change your caller-id fields to use "(" and ")" instead of "<" and ">"?
By: Badalian Vyacheslav (slavon) 2011-12-31 04:18:30.684-0600

Dont't now...
I delete <..> from DisplayName and now all work. I have two side sound.
i can test (...) only after Jan 10
By: Mark Michelson (mmichelson) 2012-01-16 15:39:38.577-0600

ASTERISK-18990.patch is a quick fix that may do the trick. Similar lines may need to be added in other places where name-addrs are being passed to parse_uri() instead of just URIs.
By: Jason Parker (jparker) 2012-01-16 17:16:00.047-0600

The Record-Route header is going to need some more work, but this should fix the issue we're seeing with the Contact header.
By: Mark Michelson (mmichelson) 2012-01-16 17:21:22.002-0600

Jason's fix will get the contact header. I'm attaching ASTERISK-18990_v2.patch, which will also fix the problem in Record-Route headers. If you apply just ASTERISK-18990_v2.patch, this should completely fix the issue. Please let us know if it does.
By: Mark Michelson (mmichelson) 2012-01-17 11:03:02.662-0600

I've gone ahead and committed ASTERISK-18990_v2.patch to branches 1.8 and 10, and to Asterisk trunk. I'm leaving this issue open for now though in case problems are reported.
By: Mark Michelson (mmichelson) 2012-01-23 13:03:02.996-0600

Closing issue because I'm certain the committed patch will fix the issue.