Re: [asterisk-users] POlycom phone not ringing behind firewall (401 permission denied)

[Ira]

Title: Re: [asterisk-users] POlycom phone not ringing behind firewall (401 permission denied)


Hello Jerry,

Tuesday, June 30, 2020, 5:23:15 AM, you wrote:





I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. 




One of my friends just had a cloud phone system set up and the vendor supplied Yealink phones which didn't work correctly with the bosses cordless headset so they replaced his phone with a Polycom one. When doing this, they insisted I set the DNS in the router to 8.8.8.8, 8.8.4.4 as they claimed it would not work with their internet providers DNS!  Seems odd, and I never tried it with the old DNS settings, but maybe it will help.

-- Ira


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[asterisk-users] POlycom phone not ringing behind firewall (401 permission denied)

[Jerry Geis]

Hi All,

I have polycom phones setup in an office connected to a cloud asterisk
server.
The polycom phones can call out just fine - audio just fine.
However a call coming into the cloud asterisk answers fine - get the
autoattendant, enter the extension and the polycom does not ring. The CLI
shows that the correct SIP extension is being Dialed  (SIP/524)

Looks like I'm getting a 401 permission denied.

What might I be missing here ?

my 524 extensions has:
[524]
type=friend
defaultname=524
defaultuser=524
secret=
dtmfmode=RFC2833
host=dynamic
context=sip-exten
qualify=yes
rtptimeout=60
rtpholdtimeout=60
rtpkeepalive=60
callerid="524 524" <524>
qualify=no
canreinvite=yes
timezone=1
nat=force_rport,comedia
disallow=all
allow=g722
allow=ulaw
allow=alaw
allow=gsm

Thanks,

Jerry

  == Using SIP RTP CoS mark 5
Audio is at 15876
Adding codec ulaw to SDP
Adding codec g722 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 1X
INVITE sip:524@X
Via: SIP/2.0/UDP 3X:5060;branch=z9hG4bK2795cec0;rport
Max-Forwards: 70
From: "WIRELESS CALLER" ;tag=as45ffbb22
To: 
Contact: 
Call-ID: 4f83ccef5bfaebf55271bc674e26165d@X:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.33.0
Date: Tue, 30 Jun 2020 12:17:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 422927332 422927332 IN IP4 X
s=Asterisk PBX 13.33.0
c=IN IP4 X
t=0 0
m=audio 15876 RTP/AVP 0 9 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
-- Called SIP/524
Retransmitting #6 (no NAT) to X
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP X:56790;branch=z9hG4bK334206641;received=X
From: ;tag=134014
To: ;tag=as181c1453
Call-ID: 192924635-1732461672-2061354149
CSeq: 1 INVITE
Server: Asterisk PBX 13.33.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2625e522"
Content-Length: 0
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[asterisk-users] Polycom and multicast

[Jerry Geis]

Does polycom support "normal" multicast from asterisk as the source?

I'm getting the impression that it only supports its OWN phone to phone
multicast or something.
Thanks,

Jerry
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Re: [asterisk-users] Polycom and SIP message

[C.Maj]

On 2019-12-19 06:10, Antony Stone wrote:
> On Thursday 19 December 2019 at 14:04:36, Jerry Geis wrote:
> 
>> I presume it would just be sending a SIP message - no need to get anything
>> back. Just want to pop a message on the phone.

I think there are some Polycoms that support RFC 3428 SIP MESSAGE for
instant messaging eg. Kirks.

> Yes, but *what* message do you need to send?

Also how long should it display for, do you want a sound to play, should
it interrupt users currently on a call, present an interface to
acknowledge the message, etc.

> How does a Polycom do this?

Disregarding RFC 3428 method, for SPIP, VVX, and probably a few other
Polycom models, you can push XML to the phones via HTTP using something
like CURL...

Here's some Asterisk dial plan example (but put this all on one line):

exten => s,n,TrySystem(
curl
--max-time 1
-d
"hi.php"
-u "push_username:push_password"
--digest
-H "Content-Type: application/x-com-polycom-spipx"
http://192.168.1.100/push
);

You'll need to update the IP to match your phone, run a webserver to
host hi.php, and set a few things up in your Polycom configs, such as
push_username and push_password, but hopefully that helps pop you in the
right direction.

Regards,

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Re: [asterisk-users] Polycom and SIP message

[Antony Stone]

On Thursday 19 December 2019 at 14:04:36, Jerry Geis wrote:

> I presume it would just be sending a SIP message - no need to get anything
> back. Just want to pop a message on the phone.

Yes, but *what* message do you need to send?

How does a Polycom do this?

Without knowing that, I don't think we can work out how to ask Asterisk to do 
it.


Antony.

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   Please reply to the list;
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Re: [asterisk-users] Polycom and SIP message

[Jerry Geis]

I presume it would just be sending a SIP message - no need to get anything
back. Just want to pop a message on the phone.

Thanks,

Jerry
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Re: [asterisk-users] Polycom and SIP message

[Antony Stone]

On Wednesday 18 December 2019 at 22:32:24, Jerry Geis wrote:

> Hi all,
> 
> I want to send a text message to a polycom phone.
> I know how to create a call file - but that will "call" the phone and
> nothing happens till the phone is answered.
> 
> How do I create a call file that will "send" a text message over SIP to the
> polycom phone?
> So the phone does not have to answer - just shows the message.

What is the SIP command you need to send?

Is it a simple "send and you're done" or is there any sort of acknowledgement 
you need to get back again?


Antony.

-- 
Because it messes up the order in which people normally read text.
> Why is top-posting such a bad thing?
> > Top-posting.
> > > What is the most annoying way of replying to e-mail?

   Please reply to the list;
 please *don't* CC me.

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[asterisk-users] Polycom and SIP message

[Jerry Geis]

Hi all,

I want to send a text message to a polycom phone.
I know how to create a call file - but that will "call" the phone and
nothing happens till the phone is answered.

How do I create a call file that will "send" a text message over SIP to the
polycom phone?
So the phone does not have to answer - just shows the message.

Thanks,

Jerry
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Re: [asterisk-users] Polycom BLF Question

[Carlos Chavez]

    This is done via the custom extension state or hints. Basically you 
create a custom hint for 444 and monitor that on your phone like any 
other extension.  You then enable or disable the hint in the same 
dialplan for 444 and 555.


https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/usingCustomDeviceStates.html


On 9/8/2019 4:32 PM, Alexander Perkins wrote:
Hi All. I have an interesting scenario. We use the Polycom VXX phones 
and have an auto-attendant on our Asterisk system. The receptionist 
can turn the auto-attendant off and on as she would like (she dials 
444 to enable and 555 to disable). However, I’d like to have one of 
the BLFs on her Polycom light up if the auto-attendant is enabled and 
off if it is disabled.


Any suggestions on how I can have the one of the Polycom BLFs stay on 
if the auto-attendant is enabled?


Any help is appreciated

Thanks,
Alex


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[asterisk-users] Polycom BLF Question

[Alexander Perkins]

Hi All. I have an interesting scenario. We use the Polycom VXX phones and
have an auto-attendant on our Asterisk system. The receptionist can turn
the auto-attendant off and on as she would like (she dials 444 to enable
and 555 to disable). However, I’d like to have one of the BLFs on her
Polycom light up if the auto-attendant is enabled and off if it is
disabled.

Any suggestions on how I can have the one of the Polycom BLFs stay on if
the auto-attendant is enabled?

Any help is appreciated

Thanks,
Alex
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Re: [asterisk-users] Polycom UC 4.x Unreachable

[Gary Reuter]

Solved it!
Turns out UCS Polycoms are quite picky about blank callerids, to the
extant they ignore those packets completely.
My global "callerid=" in sip.conf was intentionally blank.  In ten
years, in never caused a problem.
By setting to 0, the Polycoms that didn't respond to SIP OPTIONS (nor
the NOTIFY for waiting messages) now work fine.

If anyone is curious, the problem is easily reproduced in the dialplan
by setting the callerid there to blank, then the UCS polycom will
ignore that INVITE as well.  Set the callerid to anything else and
it'll ring.

On 23 August 2017 at 19:29, John Covici  wrote:
> I always set it to no, but set the registration time to 60 seconds,
> and that has always worked for me.
>
> On Wed, 23 Aug 2017 17:23:38 -0400,
> Gary Reuter wrote:
>>
>> Hello,
>> We've had dozens of Polycom 3.x firmware phones deployed and working
>> great for years.
>> Now I've finally been charged with the long-overdue task of figuring
>> out why newer Polycom devices with 4.x firmware register fine but do
>> not respond to SIP OPTIONS request and therefore always become
>> UNREACHABLE if the sip qualify setting is set to yes.
>>
>> To my dismay, searches for solutions from others who have encountered
>> this problem have given zero results.
>>
>>
>> Thanks!
>>
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> How do
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Re: [asterisk-users] Polycom UC 4.x Unreachable

[John Covici]

I always set it to no, but set the registration time to 60 seconds,
and that has always worked for me.

On Wed, 23 Aug 2017 17:23:38 -0400,
Gary Reuter wrote:
> 
> Hello,
> We've had dozens of Polycom 3.x firmware phones deployed and working
> great for years.
> Now I've finally been charged with the long-overdue task of figuring
> out why newer Polycom devices with 4.x firmware register fine but do
> not respond to SIP OPTIONS request and therefore always become
> UNREACHABLE if the sip qualify setting is set to yes.
> 
> To my dismay, searches for solutions from others who have encountered
> this problem have given zero results.
> 
> 
> Thanks!
> 
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> 
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[asterisk-users] Polycom UC 4.x Unreachable

[Gary Reuter]

Hello,
We've had dozens of Polycom 3.x firmware phones deployed and working
great for years.
Now I've finally been charged with the long-overdue task of figuring
out why newer Polycom devices with 4.x firmware register fine but do
not respond to SIP OPTIONS request and therefore always become
UNREACHABLE if the sip qualify setting is set to yes.

To my dismay, searches for solutions from others who have encountered
this problem have given zero results.


Thanks!

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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Motty Cruz]

Yves, 

I have a SoundStation IP 6000:

My sip.conf

[1006]

type=friend

context=sip-phone

call-limit=1

trustrpid=no

callerid="Conference Room3" <1006>

disallow=all

allow=ulaw

allow=alaw

username=1006

secret=secret1

dtmfmode=rfc2833

host=dynamic

mailbox=1000

nat=yes

canreinvite=no

 

Asterisk server IP XX.XX.42.16 (Public)

Client IP: 192.168.1.56

 

I would look for typos in your configuration!

 

-Motty

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, December 22, 2016 6:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

 

Do you have any LLDP or CDP enabled anywhere ?

 

2016-12-21 19:50 GMT+01:00 Victor Villarreal :

Hi Yves,

Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of the 
phone. Maybe with the snom this not happen because your switch don't see the 
MAC of the Snom as a "supperted IP Phone".

 

2016-12-21 13:59 GMT-03:00 Yves :

sorry... typo
the problematic phone has the 192.168.0.13
the asterisk has 192.168.1.211

when i connect a snom phone on the cable that was in the soundstation 6000 
before and configure the
phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...

it would be helpful if someone, that has a running soundstation ip 6000 could 
send the configuration... :-/

regards,
yves




Am 21.12.2016 um 15:13 schrieb Mauricio Tavares:

On Wed, Dec 21, 2016 at 7:50 AM, Yves  wrote:

Hi Mark,

yes, you are right... these are different VLANs
I configured the other phone to use the same IP (192.168.1.13)... and it
worked flawlessly... on the SAME Networkcable in the same plug...
so it must have something to do with the polycom phone config... remember...
when I use tcp the phone tries to register, but does not even try with
udp...

thank you,
yves

   I am a bit confused: is your problematic phone's IP 192.168.0.13
(what the error log is reporting below) or 192.168.1.13?

Am 21.12.2016 um 13:34 schrieb Mark Wiater:

Yves,

Didn't you say that

AsteriskServer: 192.168.1.211
SIP-user: 165

?

On 12/21/2016 4:24 AM, Yves wrote:

. It is sure for 100% that there is no firewall or something else mangeling
in between... another Hardphone works as expected using the same
Netzworkcable on the same Networkplug with UDP on Port 5060...


This other hardphone, what IP does it have?


50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
255.255.255.0

The line above suggests to me that your phone and your asterisk server are
on a different network, there has to be something that routes between those
two networks. Often what routes, can firewall.

000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
Temporarily not available



Mark




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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Olivier]

Do you have any LLDP or CDP enabled anywhere ?

2016-12-21 19:50 GMT+01:00 Victor Villarreal :

> Hi Yves,
>
> Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC
> of the phone. Maybe with the snom this not happen because your switch don't
> see the MAC of the Snom as a "supperted IP Phone".
>
> 2016-12-21 13:59 GMT-03:00 Yves :
>
>> sorry... typo
>> the problematic phone has the 192.168.0.13
>> the asterisk has 192.168.1.211
>>
>> when i connect a snom phone on the cable that was in the soundstation
>> 6000 before and configure the
>> phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...
>>
>> it would be helpful if someone, that has a running soundstation ip 6000
>> could send the configuration... :-/
>>
>> regards,
>> yves
>>
>>
>>
>> Am 21.12.2016 um 15:13 schrieb Mauricio Tavares:
>>
>>> On Wed, Dec 21, 2016 at 7:50 AM, Yves  wrote:
>>>
 Hi Mark,

 yes, you are right... these are different VLANs
 I configured the other phone to use the same IP (192.168.1.13)... and it
 worked flawlessly... on the SAME Networkcable in the same plug...
 so it must have something to do with the polycom phone config...
 remember...
 when I use tcp the phone tries to register, but does not even try with
 udp...

 thank you,
 yves

I am a bit confused: is your problematic phone's IP 192.168.0.13
>>> (what the error log is reporting below) or 192.168.1.13?
>>>
>>> Am 21.12.2016 um 13:34 schrieb Mark Wiater:

 Yves,

 Didn't you say that

 AsteriskServer: 192.168.1.211
 SIP-user: 165

 ?

 On 12/21/2016 4:24 AM, Yves wrote:

 . It is sure for 100% that there is no firewall or something else
 mangeling
 in between... another Hardphone works as expected using the same
 Netzworkcable on the same Networkplug with UDP on Port 5060...


 This other hardphone, what IP does it have?


 50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
 255.255.255.0

 The line above suggests to me that your phone and your asterisk server
 are
 on a different network, there has to be something that routes between
 those
 two networks. Often what routes, can firewall.

 000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
 Temporarily not available



 Mark




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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Victor Villarreal]

Hi Yves,

Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of
the phone. Maybe with the snom this not happen because your switch don't
see the MAC of the Snom as a "supperted IP Phone".

2016-12-21 13:59 GMT-03:00 Yves :

> sorry... typo
> the problematic phone has the 192.168.0.13
> the asterisk has 192.168.1.211
>
> when i connect a snom phone on the cable that was in the soundstation 6000
> before and configure the
> phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...
>
> it would be helpful if someone, that has a running soundstation ip 6000
> could send the configuration... :-/
>
> regards,
> yves
>
>
>
> Am 21.12.2016 um 15:13 schrieb Mauricio Tavares:
>
>> On Wed, Dec 21, 2016 at 7:50 AM, Yves  wrote:
>>
>>> Hi Mark,
>>>
>>> yes, you are right... these are different VLANs
>>> I configured the other phone to use the same IP (192.168.1.13)... and it
>>> worked flawlessly... on the SAME Networkcable in the same plug...
>>> so it must have something to do with the polycom phone config...
>>> remember...
>>> when I use tcp the phone tries to register, but does not even try with
>>> udp...
>>>
>>> thank you,
>>> yves
>>>
>>>I am a bit confused: is your problematic phone's IP 192.168.0.13
>> (what the error log is reporting below) or 192.168.1.13?
>>
>> Am 21.12.2016 um 13:34 schrieb Mark Wiater:
>>>
>>> Yves,
>>>
>>> Didn't you say that
>>>
>>> AsteriskServer: 192.168.1.211
>>> SIP-user: 165
>>>
>>> ?
>>>
>>> On 12/21/2016 4:24 AM, Yves wrote:
>>>
>>> . It is sure for 100% that there is no firewall or something else
>>> mangeling
>>> in between... another Hardphone works as expected using the same
>>> Netzworkcable on the same Networkplug with UDP on Port 5060...
>>>
>>>
>>> This other hardphone, what IP does it have?
>>>
>>>
>>> 50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
>>> 255.255.255.0
>>>
>>> The line above suggests to me that your phone and your asterisk server
>>> are
>>> on a different network, there has to be something that routes between
>>> those
>>> two networks. Often what routes, can firewall.
>>>
>>> 000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
>>> Temporarily not available
>>>
>>>
>>>
>>> Mark
>>>
>>>
>>>
>>>
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>>
>
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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Yves]

sorry... typo
the problematic phone has the 192.168.0.13
the asterisk has 192.168.1.211

when i connect a snom phone on the cable that was in the soundstation 
6000 before and configure the

phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP...

it would be helpful if someone, that has a running soundstation ip 6000 
could send the configuration... :-/


regards,
yves


Am 21.12.2016 um 15:13 schrieb Mauricio Tavares:

On Wed, Dec 21, 2016 at 7:50 AM, Yves  wrote:

Hi Mark,

yes, you are right... these are different VLANs
I configured the other phone to use the same IP (192.168.1.13)... and it
worked flawlessly... on the SAME Networkcable in the same plug...
so it must have something to do with the polycom phone config... remember...
when I use tcp the phone tries to register, but does not even try with
udp...

thank you,
yves


   I am a bit confused: is your problematic phone's IP 192.168.0.13
(what the error log is reporting below) or 192.168.1.13?


Am 21.12.2016 um 13:34 schrieb Mark Wiater:

Yves,

Didn't you say that

AsteriskServer: 192.168.1.211
SIP-user: 165

?

On 12/21/2016 4:24 AM, Yves wrote:

. It is sure for 100% that there is no firewall or something else mangeling
in between... another Hardphone works as expected using the same
Netzworkcable on the same Networkplug with UDP on Port 5060...


This other hardphone, what IP does it have?


50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
255.255.255.0

The line above suggests to me that your phone and your asterisk server are
on a different network, there has to be something that routes between those
two networks. Often what routes, can firewall.

000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
Temporarily not available



Mark




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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Mauricio Tavares]

On Wed, Dec 21, 2016 at 7:50 AM, Yves  wrote:
> Hi Mark,
>
> yes, you are right... these are different VLANs
> I configured the other phone to use the same IP (192.168.1.13)... and it
> worked flawlessly... on the SAME Networkcable in the same plug...
> so it must have something to do with the polycom phone config... remember...
> when I use tcp the phone tries to register, but does not even try with
> udp...
>
> thank you,
> yves
>
  I am a bit confused: is your problematic phone's IP 192.168.0.13
(what the error log is reporting below) or 192.168.1.13?

>
> Am 21.12.2016 um 13:34 schrieb Mark Wiater:
>
> Yves,
>
> Didn't you say that
>
> AsteriskServer: 192.168.1.211
> SIP-user: 165
>
> ?
>
> On 12/21/2016 4:24 AM, Yves wrote:
>
> . It is sure for 100% that there is no firewall or something else mangeling
> in between... another Hardphone works as expected using the same
> Netzworkcable on the same Networkplug with UDP on Port 5060...
>
>
> This other hardphone, what IP does it have?
>
>
> 50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask
> 255.255.255.0
>
> The line above suggests to me that your phone and your asterisk server are
> on a different network, there has to be something that routes between those
> two networks. Often what routes, can firewall.
>
> 000122.941|sip  |4|03|Registration failed User: 165, Error Code:480
> Temporarily not available
>
>
>
> Mark
>
>
>
>
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>
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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Yves]

Hi Mark,

yes, you are right... these are different VLANs
I configured the other phone to use the same IP (192.168.1.13)... and it 
worked flawlessly... on the SAME Networkcable in the same plug...
so it must have something to do with the polycom phone config... 
remember... when I use tcp the phone tries to register, but does not 
even try with udp...


thank you,
yves


Am 21.12.2016 um 13:34 schrieb Mark Wiater:

Yves,

Didn't you say that


AsteriskServer: 192.168.1.211
SIP-user: 165

?

On 12/21/2016 4:24 AM, Yves wrote:
. It is sure for 100% that there is no firewall or something else 
mangeling
in between... another Hardphone works as expected using the same 
Netzworkcable on the same Networkplug with UDP on Port 5060...


This other hardphone, what IP does it have?



50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet 
mask 255.255.255.0


The line above suggests to me that your phone and your asterisk server 
are on a different network, there has to be something that routes 
between those two networks. Often what routes, can firewall.


000122.941|sip |4|03|Registration failed User: 165, Error Code:480 
Temporarily not available





Mark




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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Mark Wiater]

Yves,

Didn't you say that


AsteriskServer: 192.168.1.211
SIP-user: 165

?

On 12/21/2016 4:24 AM, Yves wrote:
. It is sure for 100% that there is no firewall or something else 
mangeling
in between... another Hardphone works as expected using the same 
Netzworkcable on the same Networkplug with UDP on Port 5060...


This other hardphone, what IP does it have?



50.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet 
mask 255.255.255.0


The line above suggests to me that your phone and your asterisk server 
are on a different network, there has to be something that routes 
between those two networks. Often what routes, can firewall.


000122.941|sip |4|03|Registration failed User: 165, Error Code:480 
Temporarily not available





Mark
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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Yves]

Hi,

I do not have a switch to mirror the traffic... I am only remotely 
connected to the office, where all is set up.
I have full control over asterisk and the phone and I tcpdumped the 
traffic coming from the phone.
The weird thing is... if I configure the SIP-Server Setting to use TCP 
on Port 80, I see REGISTER requests.
If I configure to use UDP only on Port 5060, I do not see nothing at 
all... not a single Request coming
from the phone... and, yes... It is sure for 100% that there is no 
firewall or something else mangeling
in between... another Hardphone works as expected using the same 
Netzworkcable on the same Networkplug

with UDP on Port 5060...
Meanwhile I tried all available firmware-Versions, with and without 
provisioning. I am wondering about
downloads, the phone is trying to receive from downloads.polycom.com 
that constantly fail (yes, these

files do not exists there, the phone can communicate with the internet...)
On the other hand, I don´t think that this has something to do with the 
problem, as the phone tries to

REGISTER when I use TCP / 80

Olivier, would you mind and mail me your config-files and some 
screenshots from the phone-webconfig?

Which software-versions are you using?

thank you,
yves

if someone wants to take a look at the phone-logs:

boot-log
02.335|so   |*|01|-- Initial log entry --
02.335|so   |*|01|+++ Note that Updater log times are in GMT +++
02.335|boot |*|01|Initial log entry. Current logging level 3
02.335|copy |*|01|Initial log entry. Current logging level 3
02.335|utilm|*|01|Initial log entry. Current logging level 4
02.335|hw   |*|01|Initial log entry. Current logging level 4
02.335|ethf |*|01|Initial log entry. Current logging level 4
02.335|dns  |*|01|Initial log entry. Current logging level 3
02.335|curl |*|01|Initial log entry. Current logging level 3
02.335|sec  |*|01|Initial log entry. Current logging level 4
02.641|wdog |*|01|Initial log entry. Current logging level 4
02.641|lldp |*|01|Initial log entry. Current logging level 3
02.641|cdp  |*|01|Initial log entry. Current logging level 3
02.641|key  |*|01|Initial log entry. Current logging level 4
02.642|so   |3|01|Platform: Model=SoundStation IP 6000, 
Assembly=3111-15600-001 Rev=W Region=

02.642|so   |3|01|Platform: Board=3111-15600-001 B 0
02.642|so   |3|01|Platform: MAC=0004f2070cd3
02.643|so   |3|01|Platform: BootBlock=3.0.4.0001 (15600-001) 
11-Jul-12 08:53

02.644|so   |*|01|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56
02.644|so   |3|01|Application, main: Label=Updater, Version=Azurite 
5.0.5.2324 09-Dec-13 15:31

02.644|so   |3|01|Application, main: P/N=-Y-YYY
02.644|log  |*|01|Install file upload callback for 'Updater'

02.644|app1 |*|01|Initial log entry. Current logging level 3
02.645|cfg  |*|01|Initial log entry. Current logging level 2
02.651|app1 |3|01|Application, load: Type=SIP, Version=4.0.4.2906 
18-Apr-13 01:11

02.652|boot |*|01|Using TFFS for flash load
02.652|boot |*|01|Code length: 0x0097A585
02.652|boot |*|01|Code checksum:   0x4B86ABFB
03.631|so   |3|01|Link status is Net up Speed 100 full Duplex.
17.497|app1 |4|01|Loaded application sip.ld from local system 
successfully.


App-log
001139.870|app1 |*|03|Manual Reboot
001139.870|so   |5|03|soAudioChannel compiledOffsetsApply error: 
unrecognized verAudio 11 for headset

001140.026|so   |*|03|SoNcasC::procMsg: Client service shutdown complete
001144.025|wdog |*|03|Watchdog Expired: tSup
04.975|log  |*|03|-- Initial log entry --
04.975|so   |*|03|Platform: Model=SoundStation IP 6000, 
Assembly=3111-15600-001 Rev=W Region=

04.975|so   |*|03|Platform: Interfaceeth0 MAC=0004f2070cd3
04.977|so   |*|03|Platform: BootBlock=3.0.4.0001 (15600-001) 
11-Jul-12 08:53

04.977|so   |*|03|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56
04.977|so   |*|03|Platform: Updater=5.0.5.2324 09-Dec-13 15:31
04.977|so   |*|03|Application, main: Label=SIP, Version=Mink 
4.0.4.2906 18-Apr-13 01:11

04.977|so   |*|03|Application, main: P/N=3150-11530-404
04.977|rdisk|*|03|RAM disk created, size: 8,388,608 bytes
04.978|ocsp |*|03|O.C.S.P. Enabled = 0
04.978|tls  |*|03|Initial log entry. Current logging level 4
04.998|pmt  |*|03|Initial log entry. Current logging level 4
04.998|wdog |*|03|Initial log entry. Current logging level 4
04.998|ethf |*|03|Initial log entry. Current logging level 4
04.998|hw   |*|03|Initial log entry. Current logging level 4
04.998|ares |*|03|Initial log entry. Current logging level 4
04.998|dns  |*|03|Initial log entry. Current logging level 4
04.998|cfg  |*|03|Initial log entry. Current logging level 4
04.998|dot1x|*|03|Initial log entry. Current logging level 4
05.000|cfg  |*|03|RT|Network eth0 link went up
05.000|cfg  |*|03|RT|cfgRtNetInterfaceUpdate

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Mark Wiater]

On 12/19/2016 10:26 AM, Yves wrote:
There are no SIP Packets arriving at my asterisk at all... and it has 
nothing to do with a firewall or similar...



I can ping the phone from the asterisk,


If both of these items are true, then I'd look at the phone 
configurations. Does the provisioning file contain an address for the 
phone to contact?


Mine has voIpProt.server.1.address, but I think you can also use a 
reg.x.address in the provisioning files too.


Mark

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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Motty Cruz]

can you provide the configuration on sip.conf file? Do you have the following 
settings under the account number or ext number? 

 

host=dynamic

nat=yes

 

for instance my configuration sip.conf file is as follow: 

[1005]

type=friend

context=sip-phone

call-limit=1

trustrpid=no

callerid="iuser 1005"

disallow=all

allow=ulaw

allow=alaw

username=1005

auth=md5

secret=819c8ebd2d1525235235325235

dtmfmode=rfc2833

host=dynamic

nat=yes

canreinvite=no

 

Thanks, 
Motty

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, December 19, 2016 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

 

 

 

2016-12-19 16:26 GMT+01:00 Yves :

Hi,

I am pulling my hair for days now...

I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register with 
my Asterisk.

There are no SIP Packets arriving at my asterisk at all... and it has nothing 
to do with a firewall or similar...

Simple Question:
Does anybody have a running SoundStation IP 6000 registerd with asterisk?

 

yes, I've got several of them running.
 

If so... would you please be so kind to tell me whats wrong with my setup?

AsteriskServer: 192.168.1.211
SIP-user: 165

(the SIP-Settings on asterisk-side are OK, tested with a normal Softphone... 
registering and placing calls is no problem...)

The phone-log only says: "Registration failed User: 165, Error Code:480 
Temporarily not available"

I tried with newest firmware, resetting to factory 100 times, using a 
provisionig file (which the SoundStation correctly downloads)
but it is always the same... the SoundStation does not contact the asterisk for 
registering...


1. Do you have any switch able to mirror traffic sent and received by Polycom 
phone ?

Capturing such traffic would help to understand what's happening.

2. Some phones support zero touch config with which they download their config 
files from the Internet.

Are you sure this doesn't happen ?

3. Is SNTP/NTP correctly configured on the phone ?



 

 


Phoneversion:


Telefoninformationen 


Telefonmodell 

SoundStation IP 6000 


Teilenummer 

3111-15600-001 Rev:W 


MAC-Adresse 

00:04:F2:07:0C:D3 


IP-Adresse 

192.168.0.13 


UC-Softwareversion 

4.0.11.0583 


BootROM-Softwareversion 

5.0.5.2324 


I can ping the phone from the asterisk, the phone can reach the asterisk server 
(as it downloads the tftp files, if used with
a provisioning profile), so the route and everything is correct... I even 
connected another Hardphone on the same cable
that stuck in the Polycom... no problem... the other phone can register and 
works, so there is really no cable or firewall
related problem here... it must be a setting!

thank you





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Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

[Olivier]

2016-12-19 16:26 GMT+01:00 Yves :

> Hi,
>
> I am pulling my hair for days now...
>
> I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register
> with my Asterisk.
>
> There are no SIP Packets arriving at my asterisk at all... and it has
> nothing to do with a firewall or similar...
>
> Simple Question:
> Does anybody have a running SoundStation IP 6000 registerd with asterisk?
>

yes, I've got several of them running.


> If so... would you please be so kind to tell me whats wrong with my setup?
>
> AsteriskServer: 192.168.1.211
> SIP-user: 165
>
> (the SIP-Settings on asterisk-side are OK, tested with a normal
> Softphone... registering and placing calls is no problem...)
>
> The phone-log only says: "Registration failed User: 165, Error Code:480
> Temporarily not available"
>
> I tried with newest firmware, resetting to factory 100 times, using a
> provisionig file (which the SoundStation correctly downloads)
> but it is always the same... the SoundStation does not contact the
> asterisk for registering...
>

1. Do you have any switch able to mirror traffic sent and received by
Polycom phone ?
Capturing such traffic would help to understand what's happening.

2. Some phones support zero touch config with which they download their
config files from the Internet.
Are you sure this doesn't happen ?

3. Is SNTP/NTP correctly configured on the phone ?





>
> Phoneversion:
> Telefoninformationen
> Telefonmodell SoundStation IP 6000
> Teilenummer 3111-15600-001 Rev:W
> MAC-Adresse 00:04:F2:07:0C:D3
> IP-Adresse 192.168.0.13
> UC-Softwareversion 4.0.11.0583
> BootROM-Softwareversion 5.0.5.2324
> I can ping the phone from the asterisk, the phone can reach the asterisk
> server (as it downloads the tftp files, if used with
> a provisioning profile), so the route and everything is correct... I even
> connected another Hardphone on the same cable
> that stuck in the Polycom... no problem... the other phone can register
> and works, so there is really no cable or firewall
> related problem here... it must be a setting!
>
> thank you
>
>
>
>
> --
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>
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[asterisk-users] Polycom SoundStation IP 6000 does not register

[Yves]

Hi,

I am pulling my hair for days now...

I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register 
with my Asterisk.


There are no SIP Packets arriving at my asterisk at all... and it has 
nothing to do with a firewall or similar...


Simple Question:
Does anybody have a running SoundStation IP 6000 registerd with asterisk?
If so... would you please be so kind to tell me whats wrong with my setup?

AsteriskServer: 192.168.1.211
SIP-user: 165

(the SIP-Settings on asterisk-side are OK, tested with a normal 
Softphone... registering and placing calls is no problem...)


The phone-log only says: "Registration failed User: 165, Error Code:480 
Temporarily not available"


I tried with newest firmware, resetting to factory 100 times, using a 
provisionig file (which the SoundStation correctly downloads)
but it is always the same... the SoundStation does not contact the 
asterisk for registering...


Phoneversion:
Telefoninformationen
Telefonmodell   SoundStation IP 6000
Teilenummer 3111-15600-001 Rev:W
MAC-Adresse 00:04:F2:07:0C:D3
IP-Adresse  192.168.0.13
UC-Softwareversion  4.0.11.0583
BootROM-Softwareversion 5.0.5.2324


I can ping the phone from the asterisk, the phone can reach the asterisk 
server (as it downloads the tftp files, if used with
a provisioning profile), so the route and everything is correct... I 
even connected another Hardphone on the same cable
that stuck in the Polycom... no problem... the other phone can register 
and works, so there is really no cable or firewall

related problem here... it must be a setting!

thank you



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Re: [asterisk-users] Polycom phone registering

[Michael Ulitskiy]

It sounds like you have problems with your firewall. Your 401 replies don't 
reach the phones.

On Thursday, October 08, 2015 02:50:24 PM Jerry Geis wrote:
> Do polycom phones not LIKE using something other than port 5060 ???
> 
> I have five of them behind a firewall and my asterisk server is remote.
> Other devices are registering just fine, just not my polycom phones.
> 
> Today I notices that ONE registered, but it grabbed port 5060.
> 
> 1004/1004 12.215.64.135D  Yes
>  Yes55068
> 1006/1006 12.215.64.135D  Yes
>  Yes55066
> 401/401   (Unspecified)D  Yes
>  Yes0
> 510/510   (Unspecified)D  Yes
>  Yes0
> 511/511   12.215.64.135D  Yes
>  Yes5060
> 524/524   (Unspecified)D  Yes
>  Yes0
> 535/535   (Unspecified)D  Yes
>  Yes0
> 537/537   (Unspecified)D  Yes
>  Yes0
> 
> The 1XXX are non polycom phones and are working just fine. The other devices
> are polycom phones and only one is registering.
> 
> Each has this exact definition but unique to extension of course:
> 
> [524]
> type=friend
> defaultname=524
> defaultuser=524
> secret=
> dtmfmode=RFC2833
> host=dynamic
> description=Polycom 0004f2323292
> 
> context=smvoice-sip
> rtptimeout=60
> rtpholdtimeout=60
> rtpkeepalive=60
> callerid="Jerry"
> qualify=no
> canreinvite=yes
> timezone=1
> nat=force_rport,comedia
> disallow=all
> allow=g722
> allow=ulaw
> allow=alaw
> 
> I am getting "401 unathorized" as the response going to the polycom phones
> - then nothing back from the phone until the next registration attempt.
> 
> How do I either make the polycom ONLY use port 5060 or tell teh polycom its
> OK to use another port?
> 
> I tried takign off the force_rport and just leave comedia but that did not
> make a
> difference either.
> 
> thanks,
> 
> Jerry-- 
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[asterisk-users] Polycom phone registering

[Jerry Geis]

Do polycom phones not LIKE using something other than port 5060 ???

I have five of them behind a firewall and my asterisk server is remote.
Other devices are registering just fine, just not my polycom phones.

Today I notices that ONE registered, but it grabbed port 5060.

1004/1004 12.215.64.135D  Yes
 Yes55068
1006/1006 12.215.64.135D  Yes
 Yes55066
401/401   (Unspecified)D  Yes
 Yes0
510/510   (Unspecified)D  Yes
 Yes0
511/511   12.215.64.135D  Yes
 Yes5060
524/524   (Unspecified)D  Yes
 Yes0
535/535   (Unspecified)D  Yes
 Yes0
537/537   (Unspecified)D  Yes
 Yes0

The 1XXX are non polycom phones and are working just fine. The other devices
are polycom phones and only one is registering.

Each has this exact definition but unique to extension of course:

[524]
type=friend
defaultname=524
defaultuser=524
secret=
dtmfmode=RFC2833
host=dynamic
description=Polycom 0004f2323292

context=smvoice-sip
rtptimeout=60
rtpholdtimeout=60
rtpkeepalive=60
callerid="Jerry"
qualify=no
canreinvite=yes
timezone=1
nat=force_rport,comedia
disallow=all
allow=g722
allow=ulaw
allow=alaw

I am getting "401 unathorized" as the response going to the polycom phones
- then nothing back from the phone until the next registration attempt.

How do I either make the polycom ONLY use port 5060 or tell teh polycom its
OK to use another port?

I tried takign off the force_rport and just leave comedia but that did not
make a
difference either.

thanks,

Jerry
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Re: [asterisk-users] polycom phone behind firewall with asterisk 11.19

[D'Arcy J.M. Cain]

On Thu, 27 Aug 2015 16:17:38 -0400
Jerry Geis  wrote:
> I have a polycom phone behind a firewall.
> The phone registers - but I only hear half channel audio.

What version of Asterisk?
Which half can you hear?

After a recent update I had a problem with one way audio.  Maybe you
are having the same problem.  See the recent thread on one way audio
started by me with my solution.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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[asterisk-users] polycom phone behind firewall with asterisk 11.19

[Jerry Geis]

I have a polycom phone behind a firewall.
The phone registers - but I only hear half channel audio.

I have tried nat=yes, nat=force_rport,comedia and
nat=autio_force_rport,auto_comedia (reloading asterisk every time).

made no difference.

How might I get full audio path?

Thanks,

Jerry
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Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 01/23/2015 10:38:27 AM:

> Would you be willing to send the configuration from asterisk for this?

It is pretty bog standard, but sure:

[111]
callerid="Conf Rm" <111>
secret=x
type=friend
host=dynamic
call-limit=5
context=xx
qualify=yes
directmedia=yes
subscribemwi=no
sendrpid=pai
trustrpid=yes
allowsubscribe=yes
description=Conf. Rm. - IP6000 Speakerphone-- 
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Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration

[Jordan Cook - Gyron Networks]

> We run a variety of 5000, 6000, and 7000 series Soundstations running
> Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these
> registration issues.

Would you be willing to send the configuration from asterisk for this?


This message may be private and confidential. If you have received this message 
in error, please notify us and remove it from your system.

Gyron may monitor email traffic data and the content of email for the purposes 
of security and staff training.

Gyron Internet Ltd is a limited company registered in England and Wales. 
Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel 
Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark.

Gyron is a Deloitte Technology Fast 50 ranked company.
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Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration [Spam score:11%]

[Patrick Beaumont]

I encountered a bug in some Polycom models where it would refuse to register to 
a domain that started with a number (e.g 3something.voip.com). Could that be 
applicable here?

Regards,
Patrick.

From: Jordan Cook - Gyron Networks 
mailto:jordan.c...@gyron.net>>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Date: Friday, 23 January 2015 16:24
To: "asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>" 
mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users] Polycom SoundStation 6000 Dropping Registration [Spam 
score:11%]

Hello,

I'm having a problem with a few Polycom SoundStation 6000s. Everything works 
fine, but they drop registration to asterisk after about maybe 30 minutes – the 
phone does not re-try to register and if you try to dial out on the phone it 
says “URI Dialing is Disabled”

Has anyone else had this issue? I'm running asterisk 11.7.0.


This message may be private and confidential. If you have received this message 
in error, please notify us and remove it from your system.

Gyron may monitor email traffic data and the content of email for the purposes 
of security and staff training.

Gyron Internet Ltd is a limited company registered in England and Wales. 
Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel 
Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark.

Gyron is a Deloitte Technology Fast 50 ranked company.
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Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 01/23/2015 10:24:24 AM:
> Hello,
> 
> I'm having a problem with a few Polycom SoundStation 6000s. 
> Everything works fine, but they drop registration to asterisk after 
> about maybe 30 minutes – the phone does not re-try to register and 
> if you try to dial out on the phone it says “URI Dialing is Disabled”
> 
> Has anyone else had this issue? I'm running asterisk 11.7.0.

We run a variety of 5000, 6000, and 7000 series Soundstations running 
Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these 
registration issues.
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[asterisk-users] Polycom SoundStation 6000 Dropping Registration

[Jordan Cook - Gyron Networks]

Hello,

I'm having a problem with a few Polycom SoundStation 6000s. Everything works 
fine, but they drop registration to asterisk after about maybe 30 minutes - the 
phone does not re-try to register and if you try to dial out on the phone it 
says "URI Dialing is Disabled"

Has anyone else had this issue? I'm running asterisk 11.7.0.


This message may be private and confidential. If you have received this message 
in error, please notify us and remove it from your system.

Gyron may monitor email traffic data and the content of email for the purposes 
of security and staff training.

Gyron Internet Ltd is a limited company registered in England and Wales. 
Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel 
Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark.

Gyron is a Deloitte Technology Fast 50 ranked company.
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Re: [asterisk-users] Polycom instant messages

[Michael Englehorn]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Yeah, I found the information on how to enable it here:
http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/uc_ag_4_0_5.pdf

It's on page 117 (or page 205 according to Adobe).

It gives you an extra option in Messaging that lets you send and receive
messages.

On Mon, Jan 12, 2015 at 10:52:24AM -0500, Tech Support wrote:
> I didn't know that instant messaging was a feature with the Polycom's. Do
> you have any documentation, how-to's, etc. that you can point me to? That
> would be just way too cool.
> Thanks;
> John 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
> Englehorn
> Sent: Monday, January 12, 2015 12:19 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Polycom instant messages
> 
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Is it possible to use the instant messaging feature of Polycom phones in
> Asterisk? At the moment I'm seeing this in the SIP messaging when I try to
> send one from a Polycom 450.
> 
> <--- SIP read from UDP::5060 ---> INVITE
> sip:0100@:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP ;branch=z9hG4bK484dcd1fDD872ECE
> From: "Michael" >;tag=D2DAE96E-D8618427
> To: ;user=phone>
> CSeq: 2 INVITE
> Call-ID: d2c5011e-1d718f7-9203b936@
> Contact: >
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
> PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514
> Accept-Language: en
> Supported: 100rel,replaces
> Allow-Events: conference,talk,hold
> Authorization: Digest username="3109", realm="asterisk", nonce="",
> uri="sip:0100@:5060;user=phone",
> response="", algorithm=MD5
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 143
> 
> v=0
> o=- 1421039199 1421039199 IN IP4  s=Polycom IP Phone
> c=IN IP4 
> t=0 0
> m=message 5060 sip sip:3109@
> <->
> SIP/2.0 488 Not acceptable here
> Via: SIP/2.0/UDP  IP>;branch=z9hG4bK484dcd1fDD872ECE;received= IP>;rport=5060
> From: "Michael" >;tag=D2DAE96E-D8618427
> To: ;user=phone>;tag=as3d0d8c04
> Call-ID: d2c5011e-1d718f7-9203b936@
> CSeq: 2 INVITE
> Server: FPBX-2.11.0(11.9.0)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
> 
> Thank you,
> - --
> Michael J. Englehorn
> H: 952-884-6776 (Use first)
> E-Mail: mich...@englehorn.com
> 
> - 
> Those parts of the system that you can hit with a hammer (not advised) are
> called hardware; those program instructions that you can only curse at are
> called software.
> -- Levitating Trains and Kamikaze Genes: Technological
>  Literacy for the 1990's
>  
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> =HTwj
> -END PGP SIGNATURE-
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
> Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Polycom instant messages

[Tech Support]

I didn't know that instant messaging was a feature with the Polycom's. Do
you have any documentation, how-to's, etc. that you can point me to? That
would be just way too cool.
Thanks;
John 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Englehorn
Sent: Monday, January 12, 2015 12:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom instant messages

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Is it possible to use the instant messaging feature of Polycom phones in
Asterisk? At the moment I'm seeing this in the SIP messaging when I try to
send one from a Polycom 450.

<--- SIP read from UDP::5060 ---> INVITE
sip:0100@:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP ;branch=z9hG4bK484dcd1fDD872ECE
From: "Michael" >;tag=D2DAE96E-D8618427
To: ;user=phone>
CSeq: 2 INVITE
Call-ID: d2c5011e-1d718f7-9203b936@
Contact: >
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Authorization: Digest username="3109", realm="asterisk", nonce="",
uri="sip:0100@:5060;user=phone",
response="", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 143

v=0
o=- 1421039199 1421039199 IN IP4  s=Polycom IP Phone
c=IN IP4 
t=0 0
m=message 5060 sip sip:3109@
<->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP ;branch=z9hG4bK484dcd1fDD872ECE;received=;rport=5060
From: "Michael" >;tag=D2DAE96E-D8618427
To: ;user=phone>;tag=as3d0d8c04
Call-ID: d2c5011e-1d718f7-9203b936@
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Thank you,
- --
Michael J. Englehorn
H: 952-884-6776 (Use first)
E-Mail: mich...@englehorn.com

- 
Those parts of the system that you can hit with a hammer (not advised) are
called hardware; those program instructions that you can only curse at are
called software.
-- Levitating Trains and Kamikaze Genes: Technological
   Literacy for the 1990's
   
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.14 (GNU/Linux)

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Re: [asterisk-users] Polycom instant messages

[Matthew Jordan]

On Sun, Jan 11, 2015 at 11:19 PM, Michael Englehorn 
wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Is it possible to use the instant messaging feature of Polycom phones in
> Asterisk? At the moment I'm seeing this in the SIP messaging when I try
> to send one from a Polycom 450.
>
> <--- SIP read from UDP::5060 --->
> INVITE sip:0100@:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP ;branch=z9hG4bK484dcd1fDD872ECE
> From: "Michael" >;tag=D2DAE96E-D8618427
> To: ;user=phone>
> CSeq: 2 INVITE
> Call-ID: d2c5011e-1d718f7-9203b936@
> Contact: >
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514
> Accept-Language: en
> Supported: 100rel,replaces
> Allow-Events: conference,talk,hold
> Authorization: Digest username="3109", realm="asterisk",
> nonce="", uri="sip:0100@:5060;user=phone",
> response="", algorithm=MD5
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 143
>
> v=0
> o=- 1421039199 1421039199 IN IP4 
> s=Polycom IP Phone
> c=IN IP4 
> t=0 0
> m=message 5060 sip sip:3109@
> <->
> SIP/2.0 488 Not acceptable here
> Via: SIP/2.0/UDP  IP>;branch=z9hG4bK484dcd1fDD872ECE;received= IP>;rport=5060
> From: "Michael" >;tag=D2DAE96E-D8618427
> To: ;user=phone>;tag=as3d0d8c04
> Call-ID: d2c5011e-1d718f7-9203b936@
> CSeq: 2 INVITE
> Server: FPBX-2.11.0(11.9.0)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
>
>
Asterisk does not understand or support an SDP media type of 'message'.

Both chan_pjsip and chan_sip can support SIP MESSAGE requests, received
both in dialog and out of dialog. In addition, chan_sip will handle media
types of 'text' for real-time text received in the RTP stream.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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[asterisk-users] Polycom instant messages

[Michael Englehorn]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Is it possible to use the instant messaging feature of Polycom phones in
Asterisk? At the moment I'm seeing this in the SIP messaging when I try
to send one from a Polycom 450.

<--- SIP read from UDP::5060 --->
INVITE sip:0100@:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP ;branch=z9hG4bK484dcd1fDD872ECE
From: "Michael" >;tag=D2DAE96E-D8618427
To: ;user=phone>
CSeq: 2 INVITE
Call-ID: d2c5011e-1d718f7-9203b936@
Contact: >
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Authorization: Digest username="3109", realm="asterisk",
nonce="", uri="sip:0100@:5060;user=phone",
response="", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 143

v=0
o=- 1421039199 1421039199 IN IP4 
s=Polycom IP Phone
c=IN IP4 
t=0 0
m=message 5060 sip sip:3109@
<->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP ;branch=z9hG4bK484dcd1fDD872ECE;received=;rport=5060
From: "Michael" >;tag=D2DAE96E-D8618427
To: ;user=phone>;tag=as3d0d8c04
Call-ID: d2c5011e-1d718f7-9203b936@
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Thank you,
- -- 
Michael J. Englehorn
H: 952-884-6776 (Use first)
E-Mail: mich...@englehorn.com

- 
Those parts of the system that you can hit with a hammer (not advised)
are called hardware; those program instructions that you can only curse
at are called software.
-- Levitating Trains and Kamikaze Genes: Technological
   Literacy for the 1990's
   
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Re: [asterisk-users] Polycom DND + Intercom/Paging Override?

[Nathan Anderson]

On Thursday, September 18, 2014 10:31 AM, John Kiniston wrote:

> There is one product that I know of that is Compatible with Polycom
> paging. The Algo 8180 Audio Alerter. [snip]
> 
> You can call it via SIP from asterisk and it can multicast in the special
> Polycom format to your phones. 

Wow, I had no idea!  I have looked at SIP-based PAs in the past, including this 
one, but this completely escaped my attention.  I just browsed through the 
manual, and sure enough, this is an advertised feature.

Kinda weird that you have to buy an all-in-one loudspeaker to acquire a device 
that can act as a SIP-to-Polycom-multicast bridge...it would be nice if they 
sold a cheaper version that omitted the speaker.  (Or, even better yet, if 
Asterisk just supported this natively so that you didn't have to buy some 
hardware box.)  But still, it's nice to know that this exists and is an option.

Thanks for the heads-up!

-- 
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com

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Re: [asterisk-users] Polycom DND + Intercom/Paging Override?

[Tim Nelson]

- Original Message - 
> Tim,

> I THINK but I'm not sure that you can do this with the Polycom
> multicast page function. Have you attempted this yet?

> Thanks
> david

Given the odd nature of multicast paging with Polycom, I was hoping to avoid 
such a setup. My recollection is having this work previously with an older 
version of Asterisk (1.4.x?), and the same handsets. Time to check archived 
backups...

Thank you for the suggestion though, I may have to go that route.

--Tim

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Re: [asterisk-users] Polycom DND + Intercom/Paging Override?

[John Kiniston]

On Wed, Sep 17, 2014 at 10:06 PM, Nathan Anderson  wrote:

> BUT Polycom handsets cannot be configured to just listen to RTP being
> multicasted to a particular multicast IP like many other IP phones
> can...the signalling for Polycom multicast paging and PTT functionality is
> completely proprietary and not SIP-based, and in fact the audio itself is
> not RTP.  It is a proprietary audio packet format that has a header
> prefixed to it containing signalling information, on every audio
> packet/frame.  Therefore nothing else can initiate a multicast page except
> another Polycom phone on the same layer 2 broadcast domain...you cannot
> programmatically have Asterisk/FreePBX do this.
>

There is one product that I know of that is Compatible with Polycom paging.
The Algo 8180 Audio Alerter.

http://www.algosolutions.com/products/Audible-and-Visual-Alerting/8180-sip-audio-alerter.html

You can call it via SIP from asterisk and it can multicast in the special
Polycom format to your phones.



-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
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Re: [asterisk-users] Polycom DND + Intercom/Paging Override?

[Nathan Anderson]

Yes, I am pretty sure that if a Polycom unit is set DND and you initiate a 
multicast page from another Polycom handset on a page or PTT channel that the 
DND handset is subscribed to (like the emergency channel), then you will hear 
audio on that handset.

BUT Polycom handsets cannot be configured to just listen to RTP being 
multicasted to a particular multicast IP like many other IP phones can...the 
signalling for Polycom multicast paging and PTT functionality is completely 
proprietary and not SIP-based, and in fact the audio itself is not RTP.  It is 
a proprietary audio packet format that has a header prefixed to it containing 
signalling information, on every audio packet/frame.  Therefore nothing else 
can initiate a multicast page except another Polycom phone on the same layer 2 
broadcast domain...you cannot programmatically have Asterisk/FreePBX do this.

Polycom has released an engineering advisory documenting the format, in case 
anyone in Asterisk land is interested in writing a channel driver that can 
interoperate with this.  I for one think it would be very handy to be able to 
have Asterisk initiate group paging and push-to-talk on Polycom handsets.

The document is here: 
http://support.polycom.com/global/documents/support/technical/products/voice/Audio_Packet_Format.pdf

--
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com

On Wednesday, September 17, 2014 6:03 PM, David Wessell <> wrote:

> Tim,
> 
> I THINK but I'm not sure that you can do this with the Polycom multicast
> page function. Have you attempted this yet? 
> 
> Thanks
> david
> 
> On Tue, Sep 16, 2014 at 10:07 PM, Tim Nelson 
> wrote: 
> 
> 
>   Greetings-
> 
>   As many of your are Polycom "experienced", I was hoping some kind soul
> could provide direction on a specific issue. 
> 
>   On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding
> an instance where, using intercom/paging functionality of FreePBX, I need
> to override an end user's 'Do Not Disturb' selection on the handset. By
> default, DND simply rejects all inbound SIP INVITEs. However, a
> page/intercom needs to be allowed through.
> 
>   Any suggestions? I've read reports this is doable using Polycom config
> options for call priorities, but I've had no such luck yet. 
> 
>   Thanks!
> 
>   --Tim
> 
> 
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Re: [asterisk-users] Polycom DND + Intercom/Paging Override?

[David Wessell]

Tim,

I THINK but I'm not sure that you can do this with the Polycom multicast
page function. Have you attempted this yet?

Thanks
david

On Tue, Sep 16, 2014 at 10:07 PM, Tim Nelson  wrote:

> Greetings-
>
> As many of your are Polycom "experienced", I was hoping some kind soul
> could provide direction on a specific issue.
>
> On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an
> instance where, using intercom/paging functionality of FreePBX, I need to
> override an end user's 'Do Not Disturb' selection on the handset. By
> default, DND simply rejects all inbound SIP INVITEs. However, a
> page/intercom needs to be allowed through.
>
> Any suggestions? I've read reports this is doable using Polycom config
> options for call priorities, but I've had no such luck yet.
>
> Thanks!
>
> --Tim
>
> --
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-- 

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PO BOX 1994 Hendersonville, NC 28793
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[asterisk-users] Polycom DND + Intercom/Paging Override?

[Tim Nelson]

Greetings- 

As many of your are Polycom "experienced", I was hoping some kind soul could 
provide direction on a specific issue. 

On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an 
instance where, using intercom/paging functionality of FreePBX, I need to 
override an end user's 'Do Not Disturb' selection on the handset. By default, 
DND simply rejects all inbound SIP INVITEs. However, a page/intercom needs to 
be allowed through. 

Any suggestions? I've read reports this is doable using Polycom config options 
for call priorities, but I've had no such luck yet. 

Thanks! 

--Tim 
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[asterisk-users] Polycom 4.1.6 BLF/URL Dialing

[David Wessell]

I've just deployed several VVX 600's with the Color Expansion Module.
And I'm having a minor issue with them.

Intermittently when a call comes into a ring group the user is
presented with the call pickup option associated with a BLF entry. Not
the normal answer/reject option.

I've explicitly disabled the call appearance feature that should allow
this pop up:

attendant.behaviors.display.spontaneousCallAppearances.normal="0"
attendant.behaviors.display.spontanouseCallAppearances.automata="0"

To further confuse the issue when the call pickup option is presented
it's presented in the manner of a URL call sip:ext@pbx. Not the normal
PSTN caller ID: NPANXX. However URL dialing is disabled.

 

Has anyone run across this so far with Polycom's new 4.1.6 release?

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[asterisk-users] Polycom SSIP 7000 : how to boot with DHCP (instead of BOOTP) ?

[Olivier]

Hi,

I'm configuring a brand new polycom SSIP 7000.

To my surprise, when this telephone boots up, my DHCP server receives a
request that Wireshark classifies a BootP request from which I can't find
any Vendor identification.

The trouble is my DHCP server uses option vendor-class-identifier to serve
different settings to different devices.

Did I miss something ?

Regards
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[asterisk-users] Polycom does not register from outside to asterisk

[bilal ghayyad]

Hello;

I have asterisk Asterisk 1.8.23.0-vici and Polycom 331 and I am able to 
register from local area network and not able to register from outside the 
office. Also from outside the office, I am able to register via PhonerLite 
softphone and not able to register via Zoiper softphone.

So from outside the office, I am not able to register from Zoiper softphone and 
not able to register from Polycom 331. 

I set the externip to the router real IP address.
Also, I set nat=yes.

What could be the problem? Why I am able to register from outside the office 
using Phoner Lite softphone and not able using zoiper softphone or Polycom 331?

Is there anything special settings that can resolve this problem?

Regards
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Re: [asterisk-users] Polycom voicemail menu and alarm as beep with light

[Doug Lytle]

>> 1) The red light and the beep: How I can let the Phone only have the red 
>> light without the beep sound that keep hearing it periodically and it is 
>> bothering? Because I tried from the Polycom web based >>settings but nothing 
>> related to this .. Maybe, it is settings need to be from the setting file 
>> that we have to place it in the TFTP Server? Any advise? 

http://www.voip-connections.com/howto/how-to-stop-annoying-beep-sounds-on-polycom-phones-voice-mail-notification/
 


>>2) Regarding to accessing the voicemail from the phone (this option that is 
>>existed at the Polycom IP Phone as long the voice mail is existed), when 
>>clicking on this button, it is not accessing the voicemail main >>menu .. How 
>>I can do this? 

It depends on the model of Polycom phone. 

On the IP500 series it's defined in the sip.cfg by: 

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[asterisk-users] Polycom voicemail menu and alarm as beep with light

[bilal ghayyad]

Hello;

I am using vicidial which is using asterisk 1.8, mean while when the extension 
has voicemail, I always see the red light on the Polycom and hear the beep 
sound (toot toot) in period time. Also, I can see at the LCD an option to 
select it for accessing the voicemail  but I am facing the following problems:

1) The red light and the beep: How I can let the Phone only have the red light 
without the beep sound that keep hearing it periodically and it is bothering? 
Because I tried from the Polycom web based settings but nothing related to this 
.. Maybe, it is settings need to be from the setting file that we have to place 
it in the TFTP Server? Any advise?

2) Regarding to accessing the voicemail from the phone (this option that is 
existed at the Polycom IP Phone as long the voice mail is existed), when 
clicking on this button, it is not accessing the voicemail main menu .. How I 
can do this?

Regards
Bilal--
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Re: [asterisk-users] Polycom and forwarding.

[Carlos Alvarez]

On Wed, May 15, 2013 at 12:10 PM, Ken D'Ambrosio  wrote:

> Hey, all.  I've got an office set up with Asterisk, and forwarding's got a
> bit of a glitch:
> When they forward, they listen for the remote phone to ring, then hang up.
>  If the remote phone doesn't connect, it goes to the original phone's VM.
>  Is this Polycom's "fault," or Asterisk's?  I've been reading up on
> blind/supervised forwards, and, honestly, have myself more confused than
> when I started.  If someone could give me a solid idea of how forwarding
> works, and a sample of how to send it to a remote extension, and have it
> *not* come back to the original extension, that'd be great.
>

You said "forwarding" but described a process that sounds like call
transfer.  I'm going to assume you mean the latter?

We just had a report of this from a customer on their own server.  I
haven't had time to investigate it.  We have confirmed it with Grandstream
and Cisco SPA phones, so it's not just Polycom.

As far as the atxferdropcall someone suggested, I did try that and then the
call is just dropped off into limbo.  The caller is left on hold, and the
nothing happens on the called extension or transfer-to extension.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Polycom and forwarding.

[Richard Mudgett]

> Hey, all.  I've got an office set up with Asterisk, and forwarding's
> got
> a bit of a glitch:
> When they forward, they listen for the remote phone to ring, then
> hang
> up.  If the remote phone doesn't connect, it goes to the original
> phone's VM.  Is this Polycom's "fault," or Asterisk's?  I've been
> reading up on blind/supervised forwards, and, honestly, have myself
> more
> confused than when I started.  If someone could give me a solid idea
> of
> how forwarding works, and a sample of how to send it to a remote
> extension, and have it *not* come back to the original extension,
> that'd
> be great.

What you are describing is an attended call transfer not
call forwarding.  Call forwarding is a different feature.
>From the behavior you describe, you are using DTMF to initiate the
attended transfer.  There is an option in features.conf called
atxferdropcall that you need to set to yes to have the call not come
back to the transferrer.

Richard

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[asterisk-users] Polycom and forwarding.

[Ken D'Ambrosio]

Hey, all.  I've got an office set up with Asterisk, and forwarding's got 
a bit of a glitch:
When they forward, they listen for the remote phone to ring, then hang 
up.  If the remote phone doesn't connect, it goes to the original 
phone's VM.  Is this Polycom's "fault," or Asterisk's?  I've been 
reading up on blind/supervised forwards, and, honestly, have myself more 
confused than when I started.  If someone could give me a solid idea of 
how forwarding works, and a sample of how to send it to a remote 
extension, and have it *not* come back to the original extension, that'd 
be great.


Thanks,

-Ken

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Re: [asterisk-users] Polycom Soundpoint IP 330 provisioning

[Daniel - Asterisk]

Thank you both,

Finally I found proper bootrom at:
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/previous_voip_software.html
As my very version is not available I choose: "SoundPoint IP / SoundStation
IP BootROM 3.2.3 Rev B" which has been downloaded from my FTP with files at
spip_ssip_vvx_3_2_3_release_sig_combined.zip

My phones are running again!
I'm reading the guide to set up my future installations.

Thanks again!


Elder


On Fri, Apr 12, 2013 at 1:58 PM, Dave Fullerton <
dfullertaster...@shorelinecontainer.com> wrote:

> Daniel,
>   The bootom is not part of the SIP application that you downloaded. You
> need to download the appropriate bootrom from the link Kevin supplied.
> Before you do any more though, you really need to download the SoundpointIP
> Admin guide here:
>
> http://support.polycom.com/**global/documents/support/**
> setup_maintenance/products/**voice/spip_ssip_vvx_Admin_**
> Guide_SIP_3_2_2_eng.pdf<http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf>
>
> Read chapters 2 and 3 at a minimum. There is a lot to setting up a
> provisioning system for polycom phones and it helps to have the proper
> background before getting started.
>
> -Dave
>
>
> On 04/12/2013 01:50 PM, Daniel - Asterisk wrote:
>
>> Hello Kevin,
>> Could you please tell me where I can found the 'application' my phones
>> are looking for?
>> I've already downloaded spip_ssip_vvx_3_2_3_release_**sig combined and
>> split zips, which lack a bootrom.ld file
>> Thank you!
>> Elder
>>
>> On Fri, Apr 12, 2013 at 12:44 PM, Kevin Larsen
>> 
>> <mailto:kevin.larsen@**pioneerballoon.com>>
>> wrote:
>>
>> _http://support.polycom.com/**PolycomService/support/us/**
>> support/voice/soundpoint_ip/**soundpoint_ip330_320.html_<http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html_>
>>
>>
>> Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
>>
>>
>>
>> From: Daniel - Asterisk > <mailto:earohua...@gmail.com>>
>>
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> 
>> <mailto:asterisk-users@lists.**digium.com
>> >>,
>>
>> Date: 04/12/2013 12:42 PM
>> Subject: [asterisk-users] Polycom Soundpoint IP 330 provisioning
>> Sent by: 
>> asterisk-users-bounces@lists.**digium.com
>> 
>> <mailto:asterisk-users-**boun...@lists.digium.com
>> >
>> --**--**
>> 
>>
>>
>>
>>
>> Hello all,
>>
>> I need the bootrom.ld file to set up some Polycoms I have
>>
>> Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
>>
>>
>> I've publiched on my FTP files downloaded from
>> _http://support.polycom.com/**PolycomService/support/us/**
>> support/voice/soundpoint_ip/**soundpoint_ip330_320.html_<http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html_>
>>
>>
>> (3.2.3 combined and split zips) but my phones are still showing the
>> message: "error, application is not present"
>>
>> I apologize it is not a pure Asterisk question but I'm sure some of
>> you can help me.
>>
>> Thanks in advance!
>>
>> Elder Arohuanca
>> Lima - Peru--
>> __**__**
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
>> <http://www.api-digital.com/>-**-
>>
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>> http://www.asterisk.org/hello
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>> 
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Re: [asterisk-users] Polycom Soundpoint IP 330 provisioning

[Dave Fullerton]

Daniel,
  The bootom is not part of the SIP application that you downloaded. 
You need to download the appropriate bootrom from the link Kevin 
supplied. Before you do any more though, you really need to download the 
SoundpointIP Admin guide here:


http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf

Read chapters 2 and 3 at a minimum. There is a lot to setting up a 
provisioning system for polycom phones and it helps to have the proper 
background before getting started.


-Dave

On 04/12/2013 01:50 PM, Daniel - Asterisk wrote:

Hello Kevin,
Could you please tell me where I can found the 'application' my phones
are looking for?
I've already downloaded spip_ssip_vvx_3_2_3_release_sig combined and
split zips, which lack a bootrom.ld file
Thank you!
Elder

On Fri, Apr 12, 2013 at 12:44 PM, Kevin Larsen
mailto:kevin.lar...@pioneerballoon.com>> wrote:


_http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html_

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From: Daniel - Asterisk mailto:earohua...@gmail.com>>
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>,
Date: 04/12/2013 12:42 PM
Subject: [asterisk-users] Polycom Soundpoint IP 330 provisioning
Sent by: asterisk-users-boun...@lists.digium.com
<mailto:asterisk-users-boun...@lists.digium.com>




Hello all,

I need the bootrom.ld file to set up some Polycoms I have

Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A


I've publiched on my FTP files downloaded from

_http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html_

(3.2.3 combined and split zips) but my phones are still showing the
message: "error, application is not present"

I apologize it is not a pure Asterisk question but I'm sure some of
you can help me.

Thanks in advance!

Elder Arohuanca
Lima - Peru--
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DO NOT SEND WITH THIS ACCOUNT


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Re: [asterisk-users] Polycom Soundpoint IP 330 provisioning

[Daniel - Asterisk]

This can be useful too: Application, main: Label=BOOT, Version=3.2.3.0021
29-Mar-07 16:05

It is the log the phones are sending to my FTP

Thanks!

Elder


On Fri, Apr 12, 2013 at 12:50 PM, Daniel - Asterisk wrote:

> Hello Kevin,
>
> Could you please tell me where I can found the 'application' my phones are
> looking for?
>
> I've already downloaded spip_ssip_vvx_3_2_3_release_sig combined and split
> zips, which lack a bootrom.ld file
>
> Thank you!
>
> Elder
>
> On Fri, Apr 12, 2013 at 12:44 PM, Kevin Larsen <
> kevin.lar...@pioneerballoon.com> wrote:
>
>> *
>> http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
>> *<http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html>
>>
>> Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
>>
>>
>>
>> From:Daniel - Asterisk 
>> To:Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>,
>> Date:04/12/2013 12:42 PM
>> Subject:[asterisk-users] Polycom Soundpoint IP 330 provisioning
>> Sent by:asterisk-users-boun...@lists.digium.com
>> --
>>
>>
>>
>> Hello all,
>>
>> I need the bootrom.ld file to set up some Polycoms I have
>>
>> Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
>>
>>
>> I've publiched on my FTP files downloaded from *
>> http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
>> *<http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html>
>> (3.2.3 combined and split zips) but my phones are still showing the
>> message: "error, application is not present"
>>
>> I apologize it is not a pure Asterisk question but I'm sure some of you
>> can help me.
>>
>> Thanks in advance!
>>
>> Elder Arohuanca
>> Lima - Peru--
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Polycom Soundpoint IP 330 provisioning

[Daniel - Asterisk]

Hello Kevin,

Could you please tell me where I can found the 'application' my phones are
looking for?

I've already downloaded spip_ssip_vvx_3_2_3_release_sig combined and split
zips, which lack a bootrom.ld file

Thank you!

Elder

On Fri, Apr 12, 2013 at 12:44 PM, Kevin Larsen <
kevin.lar...@pioneerballoon.com> wrote:

> *
> http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
> *<http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html>
>
> Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
>
>
>
> From:Daniel - Asterisk 
> To:Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>,
> Date:    04/12/2013 12:42 PM
> Subject:[asterisk-users] Polycom Soundpoint IP 330 provisioning
> Sent by:asterisk-users-boun...@lists.digium.com
> --
>
>
>
> Hello all,
>
> I need the bootrom.ld file to set up some Polycoms I have
>
> Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
>
>
> I've publiched on my FTP files downloaded from *
> http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
> *<http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html>
> (3.2.3 combined and split zips) but my phones are still showing the
> message: "error, application is not present"
>
> I apologize it is not a pure Asterisk question but I'm sure some of you
> can help me.
>
> Thanks in advance!
>
> Elder Arohuanca
> Lima - Peru--
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Polycom Soundpoint IP 330 provisioning

[Kevin Larsen]

http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
 


Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From:   Daniel - Asterisk 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
, 
Date:   04/12/2013 12:42 PM
Subject:[asterisk-users] Polycom Soundpoint IP 330 provisioning
Sent by:asterisk-users-boun...@lists.digium.com



Hello all,
 
I need the bootrom.ld file to set up some Polycoms I have
 
Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
 
 
I've publiched on my FTP files downloaded from 
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
(3.2.3 combined and split zips) but my phones are still showing the 
message: "error, application is not present"
 
I apologize it is not a pure Asterisk question but I'm sure some of you 
can help me.
 
Thanks in advance!
 
Elder Arohuanca
Lima - Peru--
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[asterisk-users] Polycom Soundpoint IP 330 provisioning

[Daniel - Asterisk]

Hello all,

I need the bootrom.ld file to set up some Polycoms I have

Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A


I've publiched on my FTP files downloaded from
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
(3.2.3 combined and split zips) but my phones are still showing the
message: "error, application is not present"

I apologize it is not a pure Asterisk question but I'm sure some of you can
help me.

Thanks in advance!

Elder Arohuanca
Lima - Peru
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Re: [asterisk-users] Polycom SPIP config

[Bryan Anderson]

Ok, thanks for the info.

-Bryan Anderson


On Thu, Mar 7, 2013 at 6:07 PM, Chad Wallace wrote:

> On Thu, 7 Mar 2013 17:12:47 -0800
> Bryan Anderson  wrote:
>
> > Has any one ever worked with placing idle display images onto the
> > Polycom SPIP331 phones?  I have got it working but when the image is
> > displayed the clock is moved to the top of the screen.  That is
> > great  but it scrolls between the clock and the registered
> > extension(s) .  Has anyone figured out a way to stop the scrolling
> > and just display the time?  If so could you provide me the
> > configuration parameter?
>
> Sorry to say... we have the same problem with the 321s.  Never
> managed to figure it out.  I asked Polycom about it, and they said we'd
> have to get our vendor to order it as a feature request, or something
> like that.
>
>
> --
>
> C. Chad Wallace, B.Sc.
> The Lodging Company
> http://www.lodgingcompany.com/
> OpenPGP Public Key ID: 0x262208A0
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Polycom SPIP config

[Chad Wallace]

On Thu, 7 Mar 2013 17:12:47 -0800
Bryan Anderson  wrote:

> Has any one ever worked with placing idle display images onto the
> Polycom SPIP331 phones?  I have got it working but when the image is
> displayed the clock is moved to the top of the screen.  That is
> great  but it scrolls between the clock and the registered
> extension(s) .  Has anyone figured out a way to stop the scrolling
> and just display the time?  If so could you provide me the
> configuration parameter?

Sorry to say... we have the same problem with the 321s.  Never
managed to figure it out.  I asked Polycom about it, and they said we'd
have to get our vendor to order it as a feature request, or something
like that.


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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[asterisk-users] Polycom SPIP config

[Bryan Anderson]

Has any one ever worked with placing idle display images onto the Polycom
SPIP331 phones?  I have got it working but when the image is displayed the
clock is moved to the top of the screen.  That is great  but it scrolls
between the clock and the registered extension(s) .  Has anyone figured out
a way to stop the scrolling and just display the time?  If so could you
provide me the configuration parameter?

thanks,
Bryan Anderson
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Re: [asterisk-users] Polycom IP6000 upgrading and looping

[Kevin Larsen]

Justin,

I haven't seen it on that model, but I did have a case awhile back where 
it happened to me with a different conference phone. Pretty much the same 
symptoms you had. Even more fun it was remote so I couldn't get my hands 
on it.

I tracked mine down to being an incorrect firmware for that phone. I 
downloaded the recommended firmware for my phone and placed it in a 
subdirectory of my ftp server. Then I deleted the link to the old firmware 
for that model only and relinked it to the new firmware.

There are two pieces of firmware you might need to look at. There is the 
bootrom.ld file for your particular phone and then the sip_xxx.ld file for 
your particular phone. Either one of them could be the problem. In my 
case, the phone was updating the -app.log and -boot.log files, so I did have some clues as  to where it was 
running off the rails. 

I was able to modify the .cfg file to point only the 
conference phone to the different firmware so that I could still keep the 
other phones on the known working firmware. 

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208--
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[asterisk-users] Polycom IP6000 upgrading and looping

[Justin Sherrill]

I have a Polycom IP6000 conference phone, along with a lot of Polycom IP550 
units.  I've been updating all the 650s to Polycom's from 3.2.3  to the 4.0.3 
software release, by hodling 468*and having them pull the update.  

It's been fine with the 650s, but the IP6000 (held 68* for that one) keeps 
going in a loop - downloads updater, saves it, formats the filesystem, 
downloads the new bootROM, and then repeats.  There's no error on screen and no 
successful upload of logs to show an error.

Has anyone updated these models before and seen this?

Justin Sherrill - American Rock Salt
P: 585-991-6825 F: 585-991-6925 C: 585-298-6826


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Re: [asterisk-users] Polycom phones and ring no answer/302 Moved Temporarily

[Justin Sherrill]

I think it's 'divert.noanswer', found in site.cfg, or at least that's where I 
have it.  It's set to enabled and it still doesn't work.  Out of curiosity, do 
you have  reg.1.fwd.noanswer.status set anywhere?

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Wednesday, December 12, 2012 5:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom phones and ring no answer/302 Moved 
Temporarily

I have Polycom IP550. The "Forward" "No Answer" is working fine when enabled. I 
was looking at the sip.cfg but don't know exactly what to look for, can you 
give me a hint to where would i find that option?

Thanks,
On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill 
mailto:justin.sherr...@americanrocksalt.com>>
 wrote:
I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8.

Setting forwarding for "Always"  works as expected; the phone issues a 302 
Moved Temporarily, and Asterisk shifts the call to the new location.

Setting forwarding to "No Answer" means a 302 never gets issued.  It just rings 
and eventually goes to voicemail.  Watching with Wireshark, I never see a 302 
SIP message issued.  I can't find anything in the phone settings that look like 
it would disable this.

Anyone else with a Polycom set that sees this, or does not see this and has 
"forward no answer" working?

Justin Sherrill - American Rock Salt
P: 585-991-6825 F: 585-991-6925 C: 
585-298-6826


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Re: [asterisk-users] Polycom phones and ring no answer/302 Moved Temporarily

[motty cruz]

I have Polycom IP550. The "Forward" "No Answer" is working fine when
enabled. I was looking at the sip.cfg but don't know exactly what to look
for, can you give me a hint to where would i find that option?

Thanks,

On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill <
justin.sherr...@americanrocksalt.com> wrote:

> I have several Polycom IP550 phones running UC 4.0.3, connected to
> Asterisk 1.8.
>
> Setting forwarding for "Always"  works as expected; the phone issues a 302
> Moved Temporarily, and Asterisk shifts the call to the new location.
>
> Setting forwarding to "No Answer" means a 302 never gets issued.  It just
> rings and eventually goes to voicemail.  Watching with Wireshark, I never
> see a 302 SIP message issued.  I can't find anything in the phone settings
> that look like it would disable this.
>
> Anyone else with a Polycom set that sees this, or does not see this and
> has "forward no answer" working?
>
> Justin Sherrill - American Rock Salt
> P: 585-991-6825 F: 585-991-6925 C: 585-298-6826
>
>
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[asterisk-users] Polycom phones and ring no answer/302 Moved Temporarily

[Justin Sherrill]

I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8.

Setting forwarding for "Always"  works as expected; the phone issues a 302 
Moved Temporarily, and Asterisk shifts the call to the new location.

Setting forwarding to "No Answer" means a 302 never gets issued.  It just rings 
and eventually goes to voicemail.  Watching with Wireshark, I never see a 302 
SIP message issued.  I can't find anything in the phone settings that look like 
it would disable this.

Anyone else with a Polycom set that sees this, or does not see this and has 
"forward no answer" working?

Justin Sherrill - American Rock Salt
P: 585-991-6825 F: 585-991-6925 C: 585-298-6826


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Re: [asterisk-users] Polycom Phone Configuration Overrides Not Saved

[Paul Belanger]

On 12-09-06 10:46 AM, Chris Nighswonger wrote:

I have some Polycom 351 on Asterisk 10. On the same box as * I have a tftp
server running to handle configs, etc. The Polycom phones have no problem
grabbing config foo from the tftp server as well as writing log files back
to the server. However, when I use the web-if on a phone to set a custom
ring-tone, the web interface saves the change locally, but throws an error
stating that it cannot write the override config to the tftp server.

A look at a tcpdump shows that the phone indeed attempts to push the file
to the server:

10:33:10.197294 IP 192.168.0.23.50672 > pbx1.campus.foundations.edu.tftp:
  38 RRQ "0004f2a5b892.cfg" octet blksize 4096
10:33:10.473171 IP 192.168.0.23.29766 > pbx1.campus.foundations.edu.tftp:
  43 RRQ "2345-12365-001.sip.ld" octet blksize 4096
10:33:10.711510 IP 192.168.0.23.33183 > pbx1.campus.foundations.edu.tftp:
  42 RRQ "0004f2a5b892_reg.cfg" octet blksize 4096
10:33:10.758896 IP 192.168.0.23.28895 > pbx1.campus.foundations.edu.tftp:
  29 RRQ "sip.cfg" octet blksize 4096
10:33:11.620103 IP 192.168.0.23.21917 > pbx1.campus.foundations.edu.tftp:
  54 RRQ "overrides/0004f2a5b892-phone.cfg" octet blksize 4096
10:33:11.825803 IP 192.168.0.23.27460 > pbx1.campus.foundations.edu.tftp:
  52 RRQ "overrides/0004f2a5b892-web.cfg" octet blksize 4096
10:33:11.850646 IP 192.168.0.23.18554 > pbx1.campus.foundations.edu.tftp:
  55 RRQ "licenses/-license.cfg" octet blksize 4096
10:33:11.873077 IP 192.168.0.23.34766 > pbx1.campus.foundations.edu.tftp:
  55 RRQ "licenses/0004f2a5b892-license.cfg" octet blksize 4096
10:33:26.294928 IP 192.168.0.23.25504 > pbx1.campus.foundations.edu.tftp:
  37 WRQ "cq_de_ku4dd.wav" octet blksize 4096
10:33:36.238357 IP 192.168.0.23.63322 > pbx1.campus.foundations.edu.tftp:
  52 WRQ "overrides/0004f2a5b892-web.cfg" octet blksize 4096
10:33:36.539747 IP 192.168.0.23.37433 > pbx1.campus.foundations.edu.tftp:
  69 RRQ "languages/Website_dictionary_language_en-us.xml" octe


A look at /var/log/messages shows:

Sep  6 10:33:11 pbx1 in.tftpd[18368]: tftpd: read(ack): Connection refused

Now why is it that the phone is refused only when writing the override file?

Note that the only logging difference between a successful and unsuccessful
write is the above line from the message log. The tcpdump looks the same.


Permissions issues?

If you switched to FTP or HTTP does it work?

--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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[asterisk-users] Polycom Phone Configuration Overrides Not Saved

[Chris Nighswonger]

I have some Polycom 351 on Asterisk 10. On the same box as * I have a tftp
server running to handle configs, etc. The Polycom phones have no problem
grabbing config foo from the tftp server as well as writing log files back
to the server. However, when I use the web-if on a phone to set a custom
ring-tone, the web interface saves the change locally, but throws an error
stating that it cannot write the override config to the tftp server.

A look at a tcpdump shows that the phone indeed attempts to push the file
to the server:

10:33:10.197294 IP 192.168.0.23.50672 > pbx1.campus.foundations.edu.tftp:
 38 RRQ "0004f2a5b892.cfg" octet blksize 4096
10:33:10.473171 IP 192.168.0.23.29766 > pbx1.campus.foundations.edu.tftp:
 43 RRQ "2345-12365-001.sip.ld" octet blksize 4096
10:33:10.711510 IP 192.168.0.23.33183 > pbx1.campus.foundations.edu.tftp:
 42 RRQ "0004f2a5b892_reg.cfg" octet blksize 4096
10:33:10.758896 IP 192.168.0.23.28895 > pbx1.campus.foundations.edu.tftp:
 29 RRQ "sip.cfg" octet blksize 4096
10:33:11.620103 IP 192.168.0.23.21917 > pbx1.campus.foundations.edu.tftp:
 54 RRQ "overrides/0004f2a5b892-phone.cfg" octet blksize 4096
10:33:11.825803 IP 192.168.0.23.27460 > pbx1.campus.foundations.edu.tftp:
 52 RRQ "overrides/0004f2a5b892-web.cfg" octet blksize 4096
10:33:11.850646 IP 192.168.0.23.18554 > pbx1.campus.foundations.edu.tftp:
 55 RRQ "licenses/-license.cfg" octet blksize 4096
10:33:11.873077 IP 192.168.0.23.34766 > pbx1.campus.foundations.edu.tftp:
 55 RRQ "licenses/0004f2a5b892-license.cfg" octet blksize 4096
10:33:26.294928 IP 192.168.0.23.25504 > pbx1.campus.foundations.edu.tftp:
 37 WRQ "cq_de_ku4dd.wav" octet blksize 4096
10:33:36.238357 IP 192.168.0.23.63322 > pbx1.campus.foundations.edu.tftp:
 52 WRQ "overrides/0004f2a5b892-web.cfg" octet blksize 4096
10:33:36.539747 IP 192.168.0.23.37433 > pbx1.campus.foundations.edu.tftp:
 69 RRQ "languages/Website_dictionary_language_en-us.xml" octe


A look at /var/log/messages shows:

Sep  6 10:33:11 pbx1 in.tftpd[18368]: tftpd: read(ack): Connection refused

Now why is it that the phone is refused only when writing the override file?

Note that the only logging difference between a successful and unsuccessful
write is the above line from the message log. The tcpdump looks the same.

Kind Regards,
Chris
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Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

[Tim Nelson]

- Original Message -
> On 07/26/2012 03:32 PM, Danny Nicholas wrote:
> > Question 1 - I think asterisk only supports a limited set of
> > statuses
> 
> Asterisk does not *receive* presence updates from Polycom phones (or
> really, non-Digium phones) at all. Instead, the presence (status)
> updates you are seeing appear on your phones are the statuses that
> Asterisk itself generates based on the phones' activity.
> 

Ah, I was suspecting that to be the case. Thanks for the info!

--Tim

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Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

[Dave Fullerton]

On 07/26/2012 04:28 PM, Tim Nelson wrote:

Greetings-

I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0 
system. Everything is running smoothly with few problems. However, I have an 
issue that maybe someone could shed light on...

Many of the phones have 'buddy watch' enabled for the other phones, basically 
Polycom's version of BLF. This works fine when watched extensions are on the 
phone, ringing, etc, as the LED lights/flashes appropriately for the status. 
However, the phones also offer various presence states such as 'Out to Lunch' 
or 'Away from Desk' etc. When a phone is set to one of these presence states, 
the other phones watching never show that status. Does that make sense? Is 
there any reason why those states would not propagate between the phones 
(through Asterisk?) ?

And, on a side note, if anyone knows how to remove the 'thistle' background 
from a Polycom phone I'd be especially delighted. It was set by a user on a 
device, and there is no option to remove it, or replace it with the blank 
background which is the default. :/


If you just want to reset the background on that phone then you want:
Menu, 3, 1, 1, 4, 2, 2, Select (Seems like you should get god mode for 
that too, but alas, no).


If you want to prohibit anyone from setting that particular background 
you could always remove the jpg from the provisioning server.


As for the buddy status, I don't think it works (or ever will work) with 
asterisk. I always turn that button off when I set up my site sip.conf 
to avoid any questions.


-Dave



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Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

[Kevin P. Fleming]

On 07/26/2012 03:32 PM, Danny Nicholas wrote:

Question 1 - I think asterisk only supports a limited set of statuses


Asterisk does not *receive* presence updates from Polycom phones (or 
really, non-Digium phones) at all. Instead, the presence (status) 
updates you are seeing appear on your phones are the statuses that 
Asterisk itself generates based on the phones' activity.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

[Danny Nicholas]

Question 1 - I think asterisk only supports a limited set of statuses
Question 2 - you could reset the phone and re-provision it or possibly just
tweak the config file and update it.  I have 501's so the 550 is just a WAG.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, July 26, 2012 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

Greetings-

I've got a handful of Polycom IP 550 handsets connected to an Asterisk
1.8.12.0 system. Everything is running smoothly with few problems. However,
I have an issue that maybe someone could shed light on...

Many of the phones have 'buddy watch' enabled for the other phones,
basically Polycom's version of BLF. This works fine when watched extensions
are on the phone, ringing, etc, as the LED lights/flashes appropriately for
the status. However, the phones also offer various presence states such as
'Out to Lunch' or 'Away from Desk' etc. When a phone is set to one of these
presence states, the other phones watching never show that status. Does that
make sense? Is there any reason why those states would not propagate between
the phones (through Asterisk?) ?

And, on a side note, if anyone knows how to remove the 'thistle' background
from a Polycom phone I'd be especially delighted. It was set by a user on a
device, and there is no option to remove it, or replace it with the blank
background which is the default. :/

--Tim

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[asterisk-users] Polycom Presence with Asterisk 1.8.12.0

[Tim Nelson]

Greetings-

I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0 
system. Everything is running smoothly with few problems. However, I have an 
issue that maybe someone could shed light on...

Many of the phones have 'buddy watch' enabled for the other phones, basically 
Polycom's version of BLF. This works fine when watched extensions are on the 
phone, ringing, etc, as the LED lights/flashes appropriately for the status. 
However, the phones also offer various presence states such as 'Out to Lunch' 
or 'Away from Desk' etc. When a phone is set to one of these presence states, 
the other phones watching never show that status. Does that make sense? Is 
there any reason why those states would not propagate between the phones 
(through Asterisk?) ?

And, on a side note, if anyone knows how to remove the 'thistle' background 
from a Polycom phone I'd be especially delighted. It was set by a user on a 
device, and there is no option to remove it, or replace it with the blank 
background which is the default. :/

--Tim

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Re: [asterisk-users] Polycom, Dial Specific Number on Handset Pickup

[Chad Wallace]

On Thu, 14 Jun 2012 00:46:25 +
"Klaverstyn, David C"  wrote:

> I have a Polycom Handset on a front door and I'd like the phone to
> dial a number as soon as the handset is lifted without having to
> press and buttons or enter any numbers.  I know how to do this on a
> Linksys but I can't find out how to do it on a Polycom.
> 
> I would be greatly appreciate is some is able to tell me how this is
> accomplished.




-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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Re: [asterisk-users] Polycom, Dial Specific Number on Handset Pickup

[Justin Sherrill]

http://lists.digium.com/pipermail/asterisk-users/2012-February/270427.html

That worked for me with the polycom 3.x firmware; I haven't tried it with 4.0 
firmware yet.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David 
C
Sent: Wednesday, June 13, 2012 8:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
(asterisk-users@lists.digium.com)
Subject: [asterisk-users] Polycom, Dial Specific Number on Handset Pickup

Hi All,

I have a Polycom Handset on a front door and I'd like the phone to dial a 
number as soon as the handset is lifted without having to press and buttons or 
enter any numbers.  I know how to do this on a Linksys but I can't find out how 
to do it on a Polycom.

I would be greatly appreciate is some is able to tell me how this is 
accomplished.

Regards
David.
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Re: [asterisk-users] Polycom Caller ID

[Warren Selby]

On Tue, Jun 12, 2012 at 4:15 PM, Jon Caum  wrote:

> Hello,
>
> I have an issue I remember seeing a while ago and forgot to investigate
> further. Now it is turning into an issue and will need to be resolved. A
> customer has Polycom 335 phones (and a couple Soundstation 6000s), and when
> an extension is calling out, the screen on the 335 shows the company's
> internal CID number instead of the person they are dialing. This also
> applies to receiving calls - the internal CID is displayed as opposed to
> who was calling.
>
> I remember seeing something about connectedline issues with Polycom
> phones, but I can't find the bug I had seen 6 months ago. Does anybody know
> about this issue and what can be done to resolve?
>
>
> Thanks!
>


How is your username defined in the sip.conf entry?  I had this issue once
before when I used "fromuser=" instead of "defaultuser=" for each phone.
Almost the exact same issue you're reporting...

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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[asterisk-users] Polycom, Dial Specific Number on Handset Pickup

[Klaverstyn, David C]

Hi All,

I have a Polycom Handset on a front door and I'd like the phone to dial a 
number as soon as the handset is lifted without having to press and buttons or 
enter any numbers.  I know how to do this on a Linksys but I can't find out how 
to do it on a Polycom.

I would be greatly appreciate is some is able to tell me how this is 
accomplished.

Regards
David.--
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[asterisk-users] Polycom Caller ID

[Jon Caum]

Hello,

I have an issue I remember seeing a while ago and forgot to investigate 
further. Now it is turning into an issue and will need to be resolved. A 
customer has Polycom 335 phones (and a couple Soundstation 6000s), and when an 
extension is calling out, the screen on the 335 shows the company's internal 
CID number instead of the person they are dialing. This also applies to 
receiving calls - the internal CID is displayed as opposed to who was calling.

I remember seeing something about connectedline issues with Polycom phones, but 
I can't find the bug I had seen 6 months ago. Does anybody know about this 
issue and what can be done to resolve?


Thanks!




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Re: [asterisk-users] Polycom CX3000 IP with Asterisk?

[Danny Nicholas]

According to the specifications, it should connect with little difficulty.
http://www.voipsupply.com/polycom-cx3000 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Wednesday, March 14, 2012 12:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Polycom CX3000 IP with Asterisk?

 

I have a customer that has a CX3000 IP that was designed for MS Lync.
Anyone know if these can run as standard SIP so we can use it with Asterisk?

Thanks

Bryant

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Re: [asterisk-users] Polycom CX3000 IP with Asterisk?

[Bryant Zimmerman]

I have a customer that has a CX3000 IP that was designed for MS Lync.
Anyone know if these can run as standard SIP so we can use it with 
Asterisk?

Thanks

Bryant
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

Thanks for the clarification. You do learn something new every day.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gord Urquhart
Sent: Wednesday, February 15, 2012 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

 

It appears you need the "info=" if the string you are using is enclosed in
angle brackets.
   Alert-Info: fooworks
   Alert-Info: does not work
   Alert-Info:info= works 




On Wed, Feb 15, 2012 at 2:09 PM, Mike  wrote:

With Polycom firmware 4.0.1b?

I have 1.8, one of the latest can`t remember which is installed on that
server. Maybe the fact that my alert info has two words, and isn`t parsed
correctly by Polycom...?



Mike




> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Dave Fullerton

> Sent: Wednesday, February 15, 2012 10:20 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
>
> Which version of asterisk are you using? I just have this in 1.4 and it
> works fine:
>
> SIPAddHeader(Alert-Info: intercom);
>
> -Dave
>
> On 02/14/2012 08:10 PM, Mike wrote:
> > In case anybody was following this thread, or someone Googles it in
> > the future, here is the solution:
> >
> > This worked fine with Polycom firmware 3.3x:
> > exten =>  s,n,SIPAddHeader(Alert-Info:)
> >
> > For firmware 4.0+, apparently I needed to add info=, i.e.:
> > exten =>  s,n,SIPAddHeader(Alert-Info: info=)
> >
> > Simple, yet quite obscure (for me at least).
> >
> >
> > Mike
> >
> >> -Original Message-
> >> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> >> boun...@lists.digium.com] On Behalf Of Mike
> >> Sent: Monday, February 13, 2012 10:17 AM
> >> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> >> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> >>
> >> Thanks Dave, it at least gives me hope that my efforts aren`t wasted.
> >>
> >> Mike
> >>
> >>> -Original Message-
> >>> From: asterisk-users-boun...@lists.digium.com
> >>> [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave
> >>> Fullerton
> >>> Sent: Monday, February 13, 2012 9:39 AM
> >>> To: asterisk-users@lists.digium.com
> >>> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> >>>
> >>> On 02/10/2012 05:30 PM, Mike wrote:
> >>>> Hi,
> >>>>
> >>>> I just moved many Polycom phones from firmware v3 to 4.0.1b.
> >>>> Anto-Answer simply stopped functioning. I can downgrade and make it
> >>>> work, upgrading kills it again. There obviously is a difference in
> >>>> how the newer firmware is treating this auto answer sip header.
> >>>>
> >>>> Can anybody tell me if they have Polycom firmware 4.x.x working
> >>>> with auto-answer/paging? Just so I know it's worth my time to
> >>>> investigate, as opposed to knowing it`s a Polycom firmware bug? If
> >>>> so, did you have to make any changes to the SIP header sent to make
> >>>> Polycom phones auto
> >>> answer?
> >>>>
> >>>
> >>> I would second the others suggestions about rewriting the configs.
> >>> Polycom made extensive changes between 3.2 and 3.3, and I think they
> >> made
> >>> a fair number of changes between 3.3 and 4.0.  I have two phones
> >>> that
> >> I've
> >>> upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I
> >>> believe I have auto answer working as you describe. Here's the
> >>> pertinent snippet from my config:
> >>>
> >>> 
> >>> 
> >>>   
> >>>  >>> voIpProt.SIP.alertInfo.1.class="ringAutoAnswer"
> >>> voIpProt.SIP.alertInfo.1.value="intercom"
> >>> voIpProt.SIP.alertInfo.2.class="ringAnswerMute"
> >>> voIpProt.SIP.alertInfo.2.value="page"
> >>> voIpProt.SIP.alertInfo.3.class="autoAnswer"
> >>> voIpProt.SIP.alertInfo.3.value="silentanswer">
> >>> 
> >>>   
> >>> 
> >&

Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Gord Urquhart]

It appears you need the "info=" if the string you are using is enclosed in
angle brackets.
   Alert-Info: fooworks
   Alert-Info: does not work
   Alert-Info:info= works



On Wed, Feb 15, 2012 at 2:09 PM, Mike  wrote:

> With Polycom firmware 4.0.1b?
>
> I have 1.8, one of the latest can`t remember which is installed on that
> server. Maybe the fact that my alert info has two words, and isn`t parsed
> correctly by Polycom...?
>
>
> Mike
>
>
>
>
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Dave Fullerton
> > Sent: Wednesday, February 15, 2012 10:20 AM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> >
> > Which version of asterisk are you using? I just have this in 1.4 and it
> > works fine:
> >
> > SIPAddHeader(Alert-Info: intercom);
> >
> > -Dave
> >
> > On 02/14/2012 08:10 PM, Mike wrote:
> > > In case anybody was following this thread, or someone Googles it in
> > > the future, here is the solution:
> > >
> > > This worked fine with Polycom firmware 3.3x:
> > > exten =>  s,n,SIPAddHeader(Alert-Info:)
> > >
> > > For firmware 4.0+, apparently I needed to add info=, i.e.:
> > > exten =>  s,n,SIPAddHeader(Alert-Info: info=)
> > >
> > > Simple, yet quite obscure (for me at least).
> > >
> > >
> > > Mike
> > >
> > >> -Original Message-
> > >> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > >> boun...@lists.digium.com] On Behalf Of Mike
> > >> Sent: Monday, February 13, 2012 10:17 AM
> > >> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > >> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> > >>
> > >> Thanks Dave, it at least gives me hope that my efforts aren`t wasted.
> > >>
> > >> Mike
> > >>
> > >>> -Original Message-
> > >>> From: asterisk-users-boun...@lists.digium.com
> > >>> [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave
> > >>> Fullerton
> > >>> Sent: Monday, February 13, 2012 9:39 AM
> > >>> To: asterisk-users@lists.digium.com
> > >>> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> > >>>
> > >>> On 02/10/2012 05:30 PM, Mike wrote:
> > >>>> Hi,
> > >>>>
> > >>>> I just moved many Polycom phones from firmware v3 to 4.0.1b.
> > >>>> Anto-Answer simply stopped functioning. I can downgrade and make it
> > >>>> work, upgrading kills it again. There obviously is a difference in
> > >>>> how the newer firmware is treating this auto answer sip header.
> > >>>>
> > >>>> Can anybody tell me if they have Polycom firmware 4.x.x working
> > >>>> with auto-answer/paging? Just so I know it's worth my time to
> > >>>> investigate, as opposed to knowing it`s a Polycom firmware bug? If
> > >>>> so, did you have to make any changes to the SIP header sent to make
> > >>>> Polycom phones auto
> > >>> answer?
> > >>>>
> > >>>
> > >>> I would second the others suggestions about rewriting the configs.
> > >>> Polycom made extensive changes between 3.2 and 3.3, and I think they
> > >> made
> > >>> a fair number of changes between 3.3 and 4.0.  I have two phones
> > >>> that
> > >> I've
> > >>> upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I
> > >>> believe I have auto answer working as you describe. Here's the
> > >>> pertinent snippet from my config:
> > >>>
> > >>> 
> > >>> 
> > >>>   
> > >>>  > >>> voIpProt.SIP.alertInfo.1.class="ringAutoAnswer"
> > >>> voIpProt.SIP.alertInfo.1.value="intercom"
> > >>> voIpProt.SIP.alertInfo.2.class="ringAnswerMute"
> > >>> voIpProt.SIP.alertInfo.2.value="page"
> > >>> voIpProt.SIP.alertInfo.3.class="autoAnswer"
> > >>> voIpProt.SIP.alertInfo.3.value="silentanswer">
> > >>> 
> > &

Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

With Polycom firmware 4.0.1b?

I have 1.8, one of the latest can`t remember which is installed on that
server. Maybe the fact that my alert info has two words, and isn`t parsed
correctly by Polycom...?


Mike




> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Dave Fullerton
> Sent: Wednesday, February 15, 2012 10:20 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> 
> Which version of asterisk are you using? I just have this in 1.4 and it
> works fine:
> 
> SIPAddHeader(Alert-Info: intercom);
> 
> -Dave
> 
> On 02/14/2012 08:10 PM, Mike wrote:
> > In case anybody was following this thread, or someone Googles it in
> > the future, here is the solution:
> >
> > This worked fine with Polycom firmware 3.3x:
> > exten =>  s,n,SIPAddHeader(Alert-Info:)
> >
> > For firmware 4.0+, apparently I needed to add info=, i.e.:
> > exten =>  s,n,SIPAddHeader(Alert-Info: info=)
> >
> > Simple, yet quite obscure (for me at least).
> >
> >
> > Mike
> >
> >> -Original Message-
> >> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> >> boun...@lists.digium.com] On Behalf Of Mike
> >> Sent: Monday, February 13, 2012 10:17 AM
> >> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> >> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> >>
> >> Thanks Dave, it at least gives me hope that my efforts aren`t wasted.
> >>
> >> Mike
> >>
> >>> -Original Message-
> >>> From: asterisk-users-boun...@lists.digium.com
> >>> [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave
> >>> Fullerton
> >>> Sent: Monday, February 13, 2012 9:39 AM
> >>> To: asterisk-users@lists.digium.com
> >>> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> >>>
> >>> On 02/10/2012 05:30 PM, Mike wrote:
> >>>> Hi,
> >>>>
> >>>> I just moved many Polycom phones from firmware v3 to 4.0.1b.
> >>>> Anto-Answer simply stopped functioning. I can downgrade and make it
> >>>> work, upgrading kills it again. There obviously is a difference in
> >>>> how the newer firmware is treating this auto answer sip header.
> >>>>
> >>>> Can anybody tell me if they have Polycom firmware 4.x.x working
> >>>> with auto-answer/paging? Just so I know it's worth my time to
> >>>> investigate, as opposed to knowing it`s a Polycom firmware bug? If
> >>>> so, did you have to make any changes to the SIP header sent to make
> >>>> Polycom phones auto
> >>> answer?
> >>>>
> >>>
> >>> I would second the others suggestions about rewriting the configs.
> >>> Polycom made extensive changes between 3.2 and 3.3, and I think they
> >> made
> >>> a fair number of changes between 3.3 and 4.0.  I have two phones
> >>> that
> >> I've
> >>> upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I
> >>> believe I have auto answer working as you describe. Here's the
> >>> pertinent snippet from my config:
> >>>
> >>> 
> >>> 
> >>>   
> >>>  >>> voIpProt.SIP.alertInfo.1.class="ringAutoAnswer"
> >>> voIpProt.SIP.alertInfo.1.value="intercom"
> >>> voIpProt.SIP.alertInfo.2.class="ringAnswerMute"
> >>> voIpProt.SIP.alertInfo.2.value="page"
> >>> voIpProt.SIP.alertInfo.3.class="autoAnswer"
> >>> voIpProt.SIP.alertInfo.3.value="silentanswer">
> >>> 
> >>>   
> >>> 
> >>> 
> >>>
> >>> I have also added an  section to adjust the ringer and
> >>> timeouts
> >> for
> >>> these ring tones.
> >>>
> >>> -Dave
> >>>
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
> to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Dave Fullerton]

Which version of asterisk are you using? I just have this in 1.4 and it 
works fine:


SIPAddHeader(Alert-Info: intercom);

-Dave

On 02/14/2012 08:10 PM, Mike wrote:

In case anybody was following this thread, or someone Googles it in the
future, here is the solution:

This worked fine with Polycom firmware 3.3x:
exten =>  s,n,SIPAddHeader(Alert-Info:)

For firmware 4.0+, apparently I needed to add info=, i.e.:
exten =>  s,n,SIPAddHeader(Alert-Info: info=)

Simple, yet quite obscure (for me at least).


Mike


-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mike
Sent: Monday, February 13, 2012 10:17 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

Thanks Dave, it at least gives me hope that my efforts aren`t wasted.

Mike


-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Monday, February 13, 2012 9:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

On 02/10/2012 05:30 PM, Mike wrote:

Hi,

I just moved many Polycom phones from firmware v3 to 4.0.1b.
Anto-Answer simply stopped functioning. I can downgrade and make it
work, upgrading kills it again. There obviously is a difference in how
the newer firmware is treating this auto answer sip header.

Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it's worth my time to investigate,
as opposed to knowing it`s a Polycom firmware bug? If so, did you have
to make any changes to the SIP header sent to make Polycom phones auto

answer?




I would second the others suggestions about rewriting the configs.
Polycom made extensive changes between 3.2 and 3.3, and I think they

made

a fair number of changes between 3.3 and 4.0.  I have two phones that

I've

upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I
believe I have auto answer working as you describe. Here's the pertinent
snippet from my config:



  


  



I have also added an  section to adjust the ringer and timeouts

for

these ring tones.

-Dave



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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

In case anybody was following this thread, or someone Googles it in the
future, here is the solution:

This worked fine with Polycom firmware 3.3x: 
exten => s,n,SIPAddHeader(Alert-Info: )

For firmware 4.0+, apparently I needed to add info=, i.e.:
exten => s,n,SIPAddHeader(Alert-Info: info=)

Simple, yet quite obscure (for me at least).


Mike

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Mike
> Sent: Monday, February 13, 2012 10:17 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> 
> Thanks Dave, it at least gives me hope that my efforts aren`t wasted.
> 
> Mike
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> > boun...@lists.digium.com] On Behalf Of Dave Fullerton
> > Sent: Monday, February 13, 2012 9:39 AM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> >
> > On 02/10/2012 05:30 PM, Mike wrote:
> > > Hi,
> > >
> > > I just moved many Polycom phones from firmware v3 to 4.0.1b.
> > > Anto-Answer simply stopped functioning. I can downgrade and make it
> > > work, upgrading kills it again. There obviously is a difference in how
> > > the newer firmware is treating this auto answer sip header.
> > >
> > > Can anybody tell me if they have Polycom firmware 4.x.x working with
> > > auto-answer/paging? Just so I know it's worth my time to investigate,
> > > as opposed to knowing it`s a Polycom firmware bug? If so, did you have
> > > to make any changes to the SIP header sent to make Polycom phones auto
> > answer?
> > >
> >
> > I would second the others suggestions about rewriting the configs.
> > Polycom made extensive changes between 3.2 and 3.3, and I think they
> made
> > a fair number of changes between 3.3 and 4.0.  I have two phones that
> I've
> > upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I
> > believe I have auto answer working as you describe. Here's the pertinent
> > snippet from my config:
> >
> > 
> >
> >  
> > > voIpProt.SIP.alertInfo.1.class="ringAutoAnswer"
> > voIpProt.SIP.alertInfo.1.value="intercom"
> > voIpProt.SIP.alertInfo.2.class="ringAnswerMute"
> > voIpProt.SIP.alertInfo.2.value="page"
> > voIpProt.SIP.alertInfo.3.class="autoAnswer"
> > voIpProt.SIP.alertInfo.3.value="silentanswer">
> >
> >  
> >
> > 
> >
> > I have also added an  section to adjust the ringer and timeouts
> for
> > these ring tones.
> >
> > -Dave
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New
> > to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] Polycom IP331 Configuration

[Mark Johnson]

Thanks David. I will check it out.


-Original message-
From: "Klaverstyn, David C" 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Mon, Feb 13, 2012 04:34:30 GMT+00:00
Subject: Re: [asterisk-users] Polycom IP331 Configuration

This may help you --> 
http://www.klaverstyn.com.au/david/wiki/index.php?title=Provision_Polycom

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Johnson
Sent: Monday, 13 February 2012 5:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom IP331 Configuration

I hope this doesn't already exist, but I couldn't find anything to help.  I am 
installing a brand new Asterisk server, and want to use the Polycom IP331 
phones.  Does anyone have any steps on how to configure these?  I have 
softphones working just fine, but for some reason I can't find a clear step by 
step on provisioning the Polycoms.  Any help is greatly appreciated!

Mark J.
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

Thanks Dave, it at least gives me hope that my efforts aren`t wasted.

Mike

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Dave Fullerton
> Sent: Monday, February 13, 2012 9:39 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> 
> On 02/10/2012 05:30 PM, Mike wrote:
> > Hi,
> >
> > I just moved many Polycom phones from firmware v3 to 4.0.1b.
> > Anto-Answer simply stopped functioning. I can downgrade and make it
> > work, upgrading kills it again. There obviously is a difference in how
> > the newer firmware is treating this auto answer sip header.
> >
> > Can anybody tell me if they have Polycom firmware 4.x.x working with
> > auto-answer/paging? Just so I know it's worth my time to investigate,
> > as opposed to knowing it`s a Polycom firmware bug? If so, did you have
> > to make any changes to the SIP header sent to make Polycom phones auto
> answer?
> >
> 
> I would second the others suggestions about rewriting the configs.
> Polycom made extensive changes between 3.2 and 3.3, and I think they made
> a fair number of changes between 3.3 and 4.0.  I have two phones that I've
> upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I
> believe I have auto answer working as you describe. Here's the pertinent
> snippet from my config:
> 
> 
>
>  
> voIpProt.SIP.alertInfo.1.class="ringAutoAnswer"
> voIpProt.SIP.alertInfo.1.value="intercom"
> voIpProt.SIP.alertInfo.2.class="ringAnswerMute"
> voIpProt.SIP.alertInfo.2.value="page"
> voIpProt.SIP.alertInfo.3.class="autoAnswer"
> voIpProt.SIP.alertInfo.3.value="silentanswer">
>
>  
>
> 
> 
> I have also added an  section to adjust the ringer and timeouts for
> these ring tones.
> 
> -Dave
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Dave Fullerton]

On 02/10/2012 05:30 PM, Mike wrote:

Hi,

I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer
firmware is treating this auto answer sip header.

Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it’s worth my time to investigate, as
opposed to knowing it`s a Polycom firmware bug? If so, did you have to
make any changes to the SIP header sent to make Polycom phones auto answer?



I would second the others suggestions about rewriting the configs. 
Polycom made extensive changes between 3.2 and 3.3, and I think they 
made a fair number of changes between 3.3 and 4.0.  I have two phones 
that I've upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, 
and I believe I have auto answer working as you describe. Here's the 
pertinent snippet from my config:



  

  voIpProt.SIP.alertInfo.1.class="ringAutoAnswer" 
voIpProt.SIP.alertInfo.1.value="intercom" 
voIpProt.SIP.alertInfo.2.class="ringAnswerMute" 
voIpProt.SIP.alertInfo.2.value="page"
voIpProt.SIP.alertInfo.3.class="autoAnswer" 
voIpProt.SIP.alertInfo.3.value="silentanswer">

  

  


I have also added an  section to adjust the ringer and timeouts 
for these ring tones.


-Dave

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Re: [asterisk-users] Polycom IP331 Configuration

[Klaverstyn, David C]

This may help you --> 
http://www.klaverstyn.com.au/david/wiki/index.php?title=Provision_Polycom

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Johnson
Sent: Monday, 13 February 2012 5:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom IP331 Configuration

I hope this doesn't already exist, but I couldn't find anything to help.  I am 
installing a brand new Asterisk server, and want to use the Polycom IP331 
phones.  Does anyone have any steps on how to configure these?  I have 
softphones working just fine, but for some reason I can't find a clear step by 
step on provisioning the Polycoms.  Any help is greatly appreciated!

Mark J.
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Eric Wieling]

The Missed Calls list is broken on 4.0.x  (see the release notes).  The notes 
say the issue only happens for a couple of Polycom models, but that is not my 
experience.

We use 3.3.x

On 3.3.x I needed to add the following to make Intercom work with FreePBX.


  

  


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Sunday, February 12, 2012 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

Thanks.  Make sense, I'll try rewriting a basic phone config just for that and 
see what happens.

 

>From the lack of response I assume nobody has 4.x.x and paging working?

 

Mike

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Sunday, February 12, 2012 12:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

 

Are you using the same cfg files?
If yes I would try rewriting them from scratch using the blank cfg files that 
come with new firmware. I have seen wiered things by using olde cfg files



On Friday, February 10, 2012, Mike  wrote:
> Hi,
>
>  
>
> I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer 
> simply stopped functioning. I can downgrade and make it work, upgrading kills 
> it again. There obviously is a difference in how the newer firmware is 
> treating this auto answer sip header.
>
>  
>
> Can anybody tell me if they have Polycom firmware 4.x.x working with 
> auto-answer/paging? Just so I know it's worth my time to investigate, as 
> opposed to knowing it`s a Polycom firmware bug? If so, did you have to make 
> any changes to the SIP header sent to make Polycom phones auto answer?
>
>  
>
> Regards,
>
>  
>
> Mike
>
>  
>
>  
>
>  
>
>  
>
>  
>
>   


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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

Thanks.  Make sense, I'll try rewriting a basic phone config just for that
and see what happens.

 

>From the lack of response I assume nobody has 4.x.x and paging working?

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Sunday, February 12, 2012 12:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

 

Are you using the same cfg files?
If yes I would try rewriting them from scratch using the blank cfg files
that come with new firmware. I have seen wiered things by using olde cfg
files



On Friday, February 10, 2012, Mike  wrote:
> Hi,
>
>  
>
> I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware is
treating this auto answer sip header.
>
>  
>
> Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it's worth my time to investigate, as
opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
any changes to the SIP header sent to make Polycom phones auto answer?
>
>  
>
> Regards,
>
>  
>
> Mike
>
>  
>
>  
>
>  
>
>  
>
>  
>
>   

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[asterisk-users] Polycom IP331 Configuration

[Mark Johnson]

I hope this doesn't already exist, but I couldn't find anything to help.  I am 
installing a brand new Asterisk server, and want to use the Polycom IP331 
phones.  Does anyone have any steps on how to configure these?  I have 
softphones working just fine, but for some reason I can't find a clear step by 
step on provisioning the Polycoms.  Any help is greatly appreciated!

Mark J.
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[C F]

Are you using the same cfg files?
If yes I would try rewriting them from scratch using the blank cfg files
that come with new firmware. I have seen wiered things by using olde cfg
files



On Friday, February 10, 2012, Mike  wrote:
> Hi,
>
>
>
> I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware
is treating this auto answer sip header.
>
>
>
> Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it’s worth my time to investigate, as
opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
any changes to the SIP header sent to make Polycom phones auto answer?
>
>
>
> Regards,
>
>
>
> Mike
>
>
>
>
>
>
>
>
>
>
>
>
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

I seem to recall 3.3.2 had other problems (for me at least) so I only used it 
on some phones.  Their 3.3 branch is at 3.3.4 now though, I would try that.

 

Mike

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim DeVito
Sent: Friday, February 10, 2012 7:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

 

Mike. Yes sip.ld is the firmware. 

I wanted to jump in because i saw you had the phantom ringing problem as well. 
I am running 3.3.1 and thought upgrading to 3.3.2 would solve that problem did 
you still have the problem in 3.3.2? I thought I saw in the release notes for 
3.3.2 that was resolved. I dont have them infront of me but i suppose it is 
time to double check as I plan on upgrading 30 phones in the morning. I did 
test 3.3.2 but the phantom ring seemed so rand i thought i could just no 
reprouduce it. 

Thanks!! 

Jim 

- Original message - 
> It does update the sip.ld file, yes. So does all upgrades, no? 
> 
>
> 
> Mike 
> 
>
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny 
> Nicholas Sent: Friday, February 10, 2012 5:39 PM 
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging 
> 
>
> 
> Did the 4.0.1b update overwrite sip.ld on these phones?   If I recall 
> correctly you have to tweak that file to make auto-answer work correctly. 
> 
>
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt 
> Sent: Friday, February 10, 2012 4:37 PM 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging 
> 
>
> 
>
> 
> On Fri, Feb 10, 2012 at 10:30 PM, Mike  wrote: 
> 
> Hi, 
> 
>
> 
> I just moved many Polycom phones from firmware v3 to 4.0.1b.   Anto-Answer 
> simply stopped functioning. I can downgrade and make it work, upgrading 
> kills it again. There obviously is a difference in how the newer 
> firmware is treating this auto answer sip header. 
> 
>
> 
> Can anybody tell me if they have Polycom firmware 4.x.x working with 
> auto-answer/paging? Just so I know it's worth my time to investigate, as 
> opposed to knowing it`s a Polycom firmware bug? If so, did you have to 
> make any changes to the SIP header sent to make Polycom phones auto 
> answer? 
> 
>
> 
> Regards, 
> 
>
> 
> Mike 
> 
>
> 
>
> 
>
> 
> Hi Mike, 
> 
>
> 
> Is there a compelling reason to put version 4.0.1b on these phones? 
> 
>
> 
> Brian 
> 
>
> 
>
> 
>
> 
>
> 
>
> 
> 
> -- 
> _ 
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
> New to Asterisk? Join us for a live introductory webinar every Thurs: 
> http://www.asterisk.org/hello 
> 
> asterisk-users mailing list 
> To UNSUBSCRIBE or update options visit: 
> http://lists.digium.com/mailman/listinfo/asterisk-users 
> 
>
> 

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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Jim DeVito]

Mike. Yes sip.ld is the firmware. 

I wanted to jump in because i saw you had the phantom ringing problem as well. 
I am running 3.3.1 and thought upgrading to 3.3.2 would solve that problem did 
you still have the problem in 3.3.2? I thought I saw in the release notes for 
3.3.2 that was resolved. I dont have them infront of me but i suppose it is 
time to double check as I plan on upgrading 30 phones in the morning. I did 
test 3.3.2 but the phantom ring seemed so rand i thought i could just no 
reprouduce it.

Thanks!!

Jim 

- Original message -
> It does update the sip.ld file, yes. So does all upgrades, no?
> 
>   
> 
> Mike
> 
>   
> 
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
> Nicholas Sent: Friday, February 10, 2012 5:39 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> 
>   
> 
> Did the 4.0.1b update overwrite sip.ld on these phones?   If I recall
> correctly you have to tweak that file to make auto-answer work correctly.
> 
>   
> 
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt
> Sent: Friday, February 10, 2012 4:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> 
>   
> 
>   
> 
> On Fri, Feb 10, 2012 at 10:30 PM, Mike  wrote:
> 
> Hi,
> 
>   
> 
> I just moved many Polycom phones from firmware v3 to 4.0.1b.   Anto-Answer
> simply stopped functioning. I can downgrade and make it work, upgrading
> kills it again. There obviously is a difference in how the newer
> firmware is treating this auto answer sip header.
> 
>   
> 
> Can anybody tell me if they have Polycom firmware 4.x.x working with
> auto-answer/paging? Just so I know it's worth my time to investigate, as
> opposed to knowing it`s a Polycom firmware bug? If so, did you have to
> make any changes to the SIP header sent to make Polycom phones auto
> answer? 
> 
>   
> 
> Regards,
> 
>   
> 
> Mike
> 
>   
> 
>   
> 
>   
> 
> Hi Mike,
> 
>   
> 
> Is there a compelling reason to put version 4.0.1b on these phones?
> 
>   
> 
> Brian 
> 
>   
> 
>   
> 
>   
> 
>   
> 
>   
> 
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>                             http://www.asterisk.org/hello
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> 
>   
> 

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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

It does update the sip.ld file, yes. So does all upgrades, no?

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, February 10, 2012 5:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

 

Did the 4.0.1b update overwrite sip.ld on these phones?  If I recall
correctly you have to tweak that file to make auto-answer work correctly.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt
Sent: Friday, February 10, 2012 4:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

 

 

On Fri, Feb 10, 2012 at 10:30 PM, Mike  wrote:

Hi,

 

I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware is
treating this auto answer sip header.

 

Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it's worth my time to investigate, as
opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
any changes to the SIP header sent to make Polycom phones auto answer? 

 

Regards,

 

Mike

 

 

 

Hi Mike,

 

Is there a compelling reason to put version 4.0.1b on these phones?

 

Brian 

 

 

 

 

 


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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

Not really, it did fix the "phantom ringing" I had (phone continued to ring
when connected to a caller), which was the main reason to upgrade, but I
believe so would upgrading to 3.3.4. Some pluses for me are:

 

-  It does make booting up MUCH faster

-  There is a "Warning" message when no registrations are successful
(as opposed to just empty phone icons) making remote support easier ("would
you describe the phone icon" gets old fast)

 

But that's about it. Yes, I know if it ain't broke don't fix it, but in the
end my decision was to move FROM 3.3.2 because of the ringing issue, so I
chose 4.0.1b instead of 3.3.4.  It`s a decision I am rethinking, to be
honest.

 

Mike

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt
Sent: Friday, February 10, 2012 5:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

 

 

On Fri, Feb 10, 2012 at 10:30 PM, Mike  wrote:

Hi,

 

I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware is
treating this auto answer sip header.

 

Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it's worth my time to investigate, as
opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
any changes to the SIP header sent to make Polycom phones auto answer? 

 

Regards,

 

Mike

 

 

 

Hi Mike,

 

Is there a compelling reason to put version 4.0.1b on these phones?

 

Brian 

 

 

 

 

 


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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Danny Nicholas]

Did the 4.0.1b update overwrite sip.ld on these phones?  If I recall
correctly you have to tweak that file to make auto-answer work correctly.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt
Sent: Friday, February 10, 2012 4:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging

 

 

On Fri, Feb 10, 2012 at 10:30 PM, Mike  wrote:

Hi,

 

I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware is
treating this auto answer sip header.

 

Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it's worth my time to investigate, as
opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
any changes to the SIP header sent to make Polycom phones auto answer? 

 

Regards,

 

Mike

 

 

 

Hi Mike,

 

Is there a compelling reason to put version 4.0.1b on these phones?

 

Brian 

 

 

 

 

 


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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Brian ipt]

On Fri, Feb 10, 2012 at 10:30 PM, Mike  wrote:

> Hi,
>
> ** **
>
> I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
> simply stopped functioning. I can downgrade and make it work, upgrading
> kills it again. There obviously is a difference in how the newer firmware
> is treating this auto answer sip header.
>
> ** **
>
> Can anybody tell me if they have Polycom firmware 4.x.x working with
> auto-answer/paging? Just so I know it’s worth my time to investigate, as
> opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
> any changes to the SIP header sent to make Polycom phones auto answer? ***
> *
>
> ** **
>
> Regards,
>
> ** **
>
> Mike
>
> **
>


Hi Mike,

Is there a compelling reason to put version 4.0.1b on these phones?

Brian

> **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
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[asterisk-users] Polycom firmware 4.0.1 and paging

[Mike]

Hi,

 

I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware is
treating this auto answer sip header.

 

Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it's worth my time to investigate, as
opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
any changes to the SIP header sent to make Polycom phones auto answer? 

 

Regards,

 

Mike

 

 

 

 

 

 

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