[Freeswitch-users] Polycom configuration problems?

Lars Zeb larclap at yahoo.com
Tue Jun 23 14:57:11 PDT 2009


Thanks to Rupa and Chris for this help. I didn't know enough to understand
Chris was pointing me to the Polycom phone rather than FS. I would never
have figured this out.

 

Are Polycoms the only SIP phones which have this feature?

 

Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 23, 2009 10:46 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

Ok, most of us configure the polycoms via a provisioning interface.  usually
ftp or http.

Anyway, when using the web interface, you want to look at:

Goto the web interface, Click on SIP.

Scroll down to the Local Settings section and you need to modify digitmap
and digitmap timeout.  the syntax is in the polycom manuals which you can
donwload from polycom.

On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb <larclap at yahoo.com> wrote:

Via a web browser.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 23, 2009 8:39 AM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

How are you configuring your polycom?

On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb <larclap at yahoo.com> wrote:

I'm sorry Chris, but I don't know where the look for the "global sip.cfg and
mac/phone specific cfg" settings. I also looked for digitmap but could find
nothing.

 

Can you be more specific?

 

Thanks, Lars

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris
Burns
Sent: Monday, June 22, 2009 2:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration problems?

 

Sounds like a config issue in the <dialplan/> tag. Check global sip.cfg and
mac/phone specific cfg. When you are dialing on-hook I don't think it will
use your .digitmap or ...digitmap.timer settings. When you dial off-hook it
sure will.

On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb <larclap at yahoo.com> wrote:

I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines
on the phone. The first two are registered with a SwitchVox, the last with
Freeswitch.

 

When I select the 3rd line and begin to press numbers, pressing the 3rd
digit automatically causes the phone to begin to dial. It does not matter
which three numbers I press, the 3rd one is magic.

 

However, if I do not select a line before dialing and key a 10-digit number
into the phone, then select the 3rd line, it dials out fine.

 

You can see from the debug console output that Processing begins before it
hits any dialplan, so that cannot be the problem. I must have the line
defined incorrectly for Freeswitch.

 

Thanks for any suggestions, Lars.

 

PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078

 

Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux

 

2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl "domains". Falling back to Digest auth.

2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1001 at 192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2]

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1001 at 192.168.10.29) Running State Change CS_NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel
sofia/internal/1001 at 192.168.10.29 entering state [received][100]

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:

v=0

o=- 1245682011 1245682011 IN IP4 192.168.10.101

s=Polycom IP Phone

c=IN IP4 192.168.10.101

t=0 0

m=audio 2254 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

 

2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403
(sofia/internal/1001 at 192.168.10.29) State NEW

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:115:32000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:107:16000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G722:9:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[PCMU:0:8000:20]

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec
sofia/internal/1001 at 192.168.10.29 PCMU/8000 20 ms 160 samples

2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload
to 101

2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376
(sofia/internal/1001 at 192.168.10.29) State Change CS_NEW -> CS_INIT

2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1001 at 192.168.10.29 [BREAK]

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1001 at 192.168.10.29) Running State Change CS_INIT

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1001 at 192.168.10.29) State INIT

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83
sofia/internal/1001 at 192.168.10.29 SOFIA INIT

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111
(sofia/internal/1001 at 192.168.10.29) State Change CS_INIT -> CS_ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1001 at 192.168.10.29 [BREAK]

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1001 at 192.168.10.29) State INIT going to sleep

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1001 at 192.168.10.29) Running State Change CS_ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483
(sofia/internal/1001 at 192.168.10.29) State ROUTING

2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130
sofia/internal/1001 at 192.168.10.29 SOFIA ROUTING

2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1001 at 192.168.10.29 Standard ROUTING

2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing
1001->323 in context default

Dialplan: sofia/internal/1001 at 192.168.10.29 parsing [default->unloop]
continue=false

Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false

Dialplan: sofia/internal/1001 at 192.168.10.29 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false


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-- 
-Rupa


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-- 
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