Polycom 501 and asterisk 1.4.2

Asterisk Asterisk Support
1

Hello,

I’ve seen no report of this, the closest is bug report 9319 (bugs.digium.com/bug_view_advance … ug_id=9319) but my issue is different and remains UNFIXED by the fix for this one.

Using Polycom 501 phones on asterisk 1.4.1-1.4.2, and patching chan_sip.c to revision 59081 (svn.digium.com/view/asterisk?rev=59081&view=rev), but to no avail.

Making calls with the polycom 501 phones, or receiving calls, yields an approx 5 seconds delay with one way audio. The person on the polycom phone can hear, but the other side (the one not on the polycom) cannot hear anything. After about 5 seconds of waiting, communication is restored both ways.

Has anyone seen this or experienced this?

I’m thinking of checking out asterisk from svn branch 1.4… it’s all that remains. If this doesn’t work there’s nothing else to do, unless someone knows something.

The polycoms are on LAN with the PBX, no NAT or anything.

Thanks.

2

Going to get some sip debug stuff on what’s happening soon.

3

All right, I continued testing.

Been able to replicate the bug on one other machine (which makes 100% of the machines running asterisk 1.4.0, 1.4.1, 1.4.2 or latest from svn branch 1.4)

Here is the debug, it all appears pretty much normal, except when the call is answered there is a long (5-6 seconds) delay before the polycom can send sound.

Here is the debug on my local machine:

[code]PBX*CLI>
<------------>
– Executing [XXXXXXXXXXX@sip-internal:1] Set(“SIP/2300-0822fac0”, “SPYGROUP=10001”) in

new stack
– Executing [XXXXXXXXXXX@sip-internal:2] Dial(“SIP/2300-0822fac0”,

“Zap/g1/XXXXXXXXXXX”) in new stack
– Called g1/XXXXXXXXXXX
Audio is at 192.168.2.150 port 13582
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
PBX*CLI>
<— Transmitting (no NAT) to 192.168.2.42:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.42;branch=z9hG4bK66d40288655D07B5;received=192.168.2.42
From: “2300” sip:2300@192.168.2.150;tag=6ADE61D3-59F6A456
To: sip:XXXXXXXXXXX@192.168.2.150;user=phone;tag=as2f90b08c
Call-ID: 30f984af-e738f6d1-8e816404@192.168.2.42
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:XXXXXXXXXXX@192.168.2.150
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2627 2627 IN IP4 192.168.2.150
s=session
c=IN IP4 192.168.2.150
t=0 0
m=audio 13582 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Zap/1-1 is ringing
PBX*CLI>
<— Transmitting (no NAT) to 192.168.2.42:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.42;branch=z9hG4bK66d40288655D07B5;received=192.168.2.42
From: “2300” sip:2300@192.168.2.150;tag=6ADE61D3-59F6A456
To: sip:XXXXXXXXXXX@192.168.2.150;user=phone;tag=as2f90b08c
Call-ID: 30f984af-e738f6d1-8e816404@192.168.2.42
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:XXXXXXXXXXX@192.168.2.150
Content-Length: 0

<------------>
– Zap/1-1 answered SIP/2300-0822fac0
Audio is at 192.168.2.150 port 13582
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
PBX*CLI>
<— Reliably Transmitting (no NAT) to 192.168.2.42:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.42;branch=z9hG4bK66d40288655D07B5;received=192.168.2.42
From: “2300” sip:2300@192.168.2.150;tag=6ADE61D3-59F6A456
To: sip:XXXXXXXXXXX@192.168.2.150;user=phone;tag=as2f90b08c
Call-ID: 30f984af-e738f6d1-8e816404@192.168.2.42
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:XXXXXXXXXXX@192.168.2.150
ontent-Type: application/sdp
Content-Length: 264

v=0
o=root 2627 2628 IN IP4 192.168.2.150
s=session
c=IN IP4 192.168.2.150
t=0 0
m=audio 13582 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Apr 4 16:09:30] WARNING[2991]: cdr.c:482 ast_cdr_merge: CDR start disagreement for

SIP/2300-0822fac0
PBX*CLI>
<— SIP read from 192.168.2.42:5060 —>
ACK sip:XXXXXXXXXXX@192.168.2.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.42;branch=z9hG4bKd5876c194026A45E
From: “2300” sip:2300@192.168.2.150;tag=6ADE61D3-59F6A456
To: sip:XXXXXXXXXXX@192.168.2.150;user=phone;tag=as2f90b08c
CSeq: 2 ACK
Call-ID: 30f984af-e738f6d1-8e816404@192.168.2.42
Contact: sip:2300@192.168.2.42
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,

REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291
Proxy-Authorization: Digest username=“2300”, realm=“asterisk”, nonce=“693b6b02”,

uri=“sip:XXXXXXXXXXX@192.168.2.150:5060;user=phone”,

response=“2552bad5f4bdc1588dd4821e56d2259b”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —
[Apr 4 16:09:34] WARNING[2755]: app_voicemail.c:2299 inboxcount: Failed to obtain database

object for ‘’!
PBX*CLI>
<— SIP read from 192.168.2.42:5060 —>
BYE sip:XXXXXXXXXXX@192.168.2.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.42;branch=z9hG4bK6f40e9bD51AE10
From: “2300” sip:2300@192.168.2.150;tag=6ADE61D3-59F6A456
To: sip:XXXXXXXXXXX@192.168.2.150;user=phone;tag=as2f90b08c
CSeq: 3 BYE
Call-ID: 30f984af-e738f6d1-8e816404@192.168.2.42
Contact: sip:2300@192.168.2.42
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291
Proxy-Authorization: Digest username=“2300”, realm=“asterisk”, nonce=“693b6b02”,

uri=“sip:XXXXXXXXXXX@192.168.2.150:5060;user=phone”,

response=“36e0338e6856e6adc48201b241d9b925”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.2.42 : 5060 (no NAT)
PBX*CLI>
<— Transmitting (no NAT) to 192.168.2.42:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.42;branch=z9hG4bK6f40e9bD51AE10;received=192.168.2.42
From: “2300” sip:2300@192.168.2.150;tag=6ADE61D3-59F6A456
To: sip:XXXXXXXXXXX@192.168.2.150;user=phone;tag=as2f90b08c
Call-ID: 30f984af-e738f6d1-8e816404@192.168.2.42
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:XXXXXXXXXXX@192.168.2.150
Content-Length: 0

<------------>
– Hungup ‘Zap/1-1’
== Spawn extension (sip-internal, XXXXXXXXXXX, 2) exited non-zero on 'SIP/2300-0822fac0’
Really destroying SIP dialog ‘30f984af-e738f6d1-8e816404@192.168.2.42’ Method: BYE
[/code]

Anyone see anything. It looks normal but that annoying delay sure is there!

The polycoms are running SIP 2.0.1 firmware and bootrom 3.2.2 if I remember.