[Freeswitch-users] Polycom configuration problems?

Tue Jun 23 10:46:14 PDT 2009

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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-- 
-Rupa
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