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Summary:ASTERISK-02541: Audio from PBX not heard on Polycom IP500/SIP phone with current release or -HEAD
Reporter:ltropiano (ltropiano)Labels:
Date Opened:2004-10-06 00:14:41Date Closed:2008-06-07 10:45:35
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) debugplcm.txt
( 1) zapdiff.txt
Description:I had CVS-HEAD 8/15 and it was working.  Upgraded to 1.0.1 and things stopped also tried Current -HEAD.  Same thing.

Polycom IP500/SIP Phones can talk to each other, through the PBX (ie. no reinvite).  Audio both directions is fine.

But when I call VM or the IVR (although I see it try (from the console) I don't hear any audio on the phone.  tethereal shows RTP packets from the phone to the PBX just fine.  I see the SIP handshaking and then *one* RTP packet from the PBX to the phone and then it stops.  No NAT, no firewall.  They are on the same "switch/LAN".  I have a debug file that
I captured that might shed some light...

http://lenny.tropiano.org/asterisk/debugplcm.txt


****** STEPS TO REPRODUCE ******

sip.conf:

[201]
type=friend
insecure=yes
username=201
secret=ext201
host=dynamic
callerid="201 Lenny Tropiano" <201>
callgroup=1
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=ulaw
context=internal-extensions

[202]
type=friend
insecure=yes
username=202
secret=ext202
host=dynamic
callerid="202 Polycom 2" <202>
callgroup=1
mailbox=202@internal-extensions
dtmfmode=rfc2833
reinvite=no
canreinvite=no
qualify=yes
context=internal-extensions

Comments:By: Mark Spencer (markster) 2004-10-06 00:25:17

This is a zaptel configuration issue (i.e. not taking interrupts, not having the spans configured, etc).
By: ltropiano (ltropiano) 2004-10-06 00:39:22

Sorry to be a pain.  But I've never had this problem before...  

The system will be connected to a T1/Supertrunk E&M Winkstart (1-24 channels).  Here's my pertenant info in

/etc/zaptel.conf:

span=1,0,0,d4,ami
e&m=1-24
loadzone = us
defaultzone=us

/etc/asterisk/zapata.conf:

switchtype=5ess
signalling=em_w
usecallerid=no
hidecallerid=no
callwaiting=no
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=random
echocancel=yes
echocancelwhenbridged=no
context=Inbound-Calls
channel => 1-24

It's *not* hooked to the T1 while I'm building it, so it's in RED alarm (to be expected), but ...

[chan_zap.so] => (Zapata Telephony w/PRI)
 == Parsing '/etc/asterisk/zapata.conf': Found
   -- Registered channel 1, E & M Wink signalling
   -- Registered channel 2, E & M Wink signalling
   -- Registered channel 3, E & M Wink signalling
   -- Registered channel 4, E & M Wink signalling
   -- Registered channel 5, E & M Wink signalling
   -- Registered channel 6, E & M Wink signalling
   -- Registered channel 7, E & M Wink signalling
   -- Registered channel 8, E & M Wink signalling
   -- Registered channel 9, E & M Wink signalling
   -- Registered channel 10, E & M Wink signalling
   -- Registered channel 11, E & M Wink signalling
   -- Registered channel 12, E & M Wink signalling
   -- Registered channel 13, E & M Wink signalling
   -- Registered channel 14, E & M Wink signalling
   -- Registered channel 15, E & M Wink signalling
   -- Registered channel 16, E & M Wink signalling
   -- Registered channel 17, E & M Wink signalling
   -- Registered channel 18, E & M Wink signalling
   -- Registered channel 19, E & M Wink signalling
   -- Registered channel 20, E & M Wink signalling
   -- Registered channel 21, E & M Wink signalling
   -- Registered channel 22, E & M Wink signalling
   -- Registered channel 23, E & M Wink signalling
   -- Registered channel 24, E & M Wink signalling
   -- Automatically generated pseudo channel


If you can suggest/recommend where to look, I'd be grateful.  This btw, is a Quad card.  

[root@pbx asterisk]# ztcfg -vvv

Zaptel Configuration
======================

SPAN 1:  D4/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: E & M (Default) (Slaves: 01)
Channel 02: E & M (Default) (Slaves: 02)
Channel 03: E & M (Default) (Slaves: 03)
Channel 04: E & M (Default) (Slaves: 04)
Channel 05: E & M (Default) (Slaves: 05)
Channel 06: E & M (Default) (Slaves: 06)
Channel 07: E & M (Default) (Slaves: 07)
Channel 08: E & M (Default) (Slaves: 08)
Channel 09: E & M (Default) (Slaves: 09)
Channel 10: E & M (Default) (Slaves: 10)
Channel 11: E & M (Default) (Slaves: 11)
Channel 12: E & M (Default) (Slaves: 12)
Channel 13: E & M (Default) (Slaves: 13)
Channel 14: E & M (Default) (Slaves: 14)
Channel 15: E & M (Default) (Slaves: 15)
Channel 16: E & M (Default) (Slaves: 16)
Channel 17: E & M (Default) (Slaves: 17)
Channel 18: E & M (Default) (Slaves: 18)
Channel 19: E & M (Default) (Slaves: 19)
Channel 20: E & M (Default) (Slaves: 20)
Channel 21: E & M (Default) (Slaves: 21)
Channel 22: E & M (Default) (Slaves: 22)
Channel 23: E & M (Default) (Slaves: 23)
Channel 24: E & M (Default) (Slaves: 24)

24 channels configured.
By: Mark Spencer (markster) 2004-10-06 00:40:36

Find me on IRC or call Digium technical support tomorrow.  You may not be taking interrupts.
By: ltropiano (ltropiano) 2004-10-06 12:08:23

Per Mark (after debugging this on my system)... a "one liner" in zaptel he says...
By: Mark Spencer (markster) 2004-10-06 12:13:56

It was a zap problem, but as it turns out, a zap bug.  Also should go to 1.0 :)
By: Russell Bryant (russell) 2004-10-06 18:10:30

added to the 1.0 branch
By: Digium Subversion (svnbot) 2008-06-07 10:45:35

Repository: dahdi
Revision: 465

U   branches/v1-0/zaptel.c

------------------------------------------------------------------------
r465 | russell | 2008-06-07 10:45:34 -0500 (Sat, 07 Jun 2008) | 2 lines

minor zaptel edit (bug ASTERISK-2541)

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

http://svn.digium.com/view/dahdi?view=rev&revision=465