[Freeswitch-users] Hold Music not Playing when pressing Hold on Polycom Phone (Lyle Pratt)
Lyle Pratt
lylepratt at gmail.com
Sun Feb 9 00:33:02 MSK 2014
An update on why this was happening so others will know:
I disabled the Sip ALG on my router and everything now seems to be working
fine.
So, hold music will not work on some Polycom phones behind NAT with ALG
connecting to remote Freeswitch servers.
Thanks,
Lyle Pratt
On Thu, Feb 6, 2014 at 7:04 AM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
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> Today's Topics:
>
> 1. Re: routing incoming SMS? (Privus P)
> 2. Re: Filler Music in Call Recordings (Michael Jerris)
> 3. Re: Hold Music not Playing when pressing Hold on Polycom
> Phone (Michael Jerris)
>
>
> ---------- Forwarded message ----------
> From: Privus P <privus007 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 6 Feb 2014 12:58:35 +0000
> Subject: Re: [Freeswitch-users] routing incoming SMS?
> Hi Donny,
>
> Perhaps you could share your javascript and lua script so that we could
> try to figure out together what isn't working.
> Judging by the apparent lack of response to this thread, it seems to be
> the best way forward to try and resolve this.
>
>
> On Thu, Feb 6, 2014 at 6:50 AM, Donny Hardyanto <hardyanto.donny at gmail.com
> > wrote:
>
>> Hi,
>>
>> I have similar problem. I use gateway to GSM provider that support SIP
>> simple . So far i can send sms from fs to gateway and receive sms in fs
>> from gateway. But I was lost how to send the incoming sms from gateway to
>> client.
>>
>> Also my client support sip simple also. When the sms from client arrive
>> at fs, i dont know how to forward them to gateway.
>>
>> I use Javascript heavily and tried to produce the same thing as lua
>> script in javascript but it always fail. The event fire command does not
>> produce any thing.
>>
>> Can any one show us or point us how to forward sms?
>>
>> Dinny
>> On Feb 4, 2014 8:06 PM, "Peter Villeneuve" <petervnv1 at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Thanks for helping out. Indeed I would love to try out a lua script to
>>> handle SMS.
>>> Can anyone point to a sample script that I can study and play with?
>>>
>>> Any help is much appreciated. I'm still stuck with SMS hitting FS but
>>> not being forwarded to the peer....
>>>
>>> Thanks,
>>> Peter
>>>
>>>
>>> On Sun, Feb 2, 2014 at 5:46 PM, Privus P <privus007 at gmail.com> wrote:
>>>
>>>> Hi Peter,
>>>>
>>>> I'm not really sure why your SMS isn't being routed correctly.
>>>> I'm sure others in this list have a lot more experience and can help
>>>> you out, but have you thought of using a lua script to handle SMS instead
>>>> of relying on the send action?
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Feb 1, 2014 at 2:00 PM, Peter Villeneuve <petervnv1 at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I'm experimenting with GSMopen and I'm having trouble doing something
>>>>> which is likely pretty simple.
>>>>> I have enabled mod_sms and GSMopen in FS. I can send SIP simple
>>>>> messages between 2 registered peers OK.
>>>>> I can see that incoming GSM SMS messages arrive in FS as expected, but
>>>>> I can't seem to get them converted into SIP SIMPLE format and routed to the
>>>>> peer (1000).
>>>>> Once they arrive in FS, I can see in the log:
>>>>>
>>>>> 2014-01-30 20:00:54.840999 [INFO] mod_sms.c:336 Processing text
>>>>> message +4412398746->gsm01 in context default
>>>>> Chatplan: gsm01 parsing [default->basic p2p] continue=true
>>>>> Chatplan: gsm01 Regex (PASS) [basic p2p] to(gsm01) =~ /^(.*)$/
>>>>> break=on-false
>>>>> Chatplan: gsm01 Action send()
>>>>>
>>>>>
>>>>> My chatplan has:
>>>>>
>>>>> <?xml version="1.0" encoding="utf-8"?>
>>>>> <include>
>>>>> <context name="default">
>>>>>
>>>>> <extension name="basic p2p" continue="true">
>>>>> <condition field="to" expression="^(.*)$">
>>>>> <!-- <action application="lua" data="test.lua"/> -->
>>>>> <action application="send"/>
>>>>> </condition>
>>>>> </extension>
>>>>>
>>>>> </context>
>>>>> </include>
>>>>>
>>>>> And in my gsmopen.conf.xml:
>>>>>
>>>>> <configuration name="gsmopen.conf" description="GSMopen Configuration">
>>>>> <global_settings>
>>>>> <param name="debug" value="8"/>
>>>>> <param name="dialplan" value="XML"/>
>>>>> <param name="context" value="default"/>
>>>>> <param name="hold-music" value="$${moh_uri}"/>
>>>>> <param name="destination" value="1000"/>
>>>>> </global_settings>
>>>>>
>>>>>
>>>>> I believe I'm doing something wrong in the chatplan. Is there any
>>>>> transfer action like in the regular XML dialplan?
>>>>> I tried creating in the default dialplan the following, hoping that it
>>>>> would be similar to sofia calls and transfer the incoming SMS to peer 1000,
>>>>> but no dice.
>>>>>
>>>>> <include>
>>>>> <extension name="sms_inbound">
>>>>> <condition field="destination_number" expression="^(gsm01)$">
>>>>> <action application="transfer" data="1000 XML default"/>
>>>>> </condition>
>>>>> </extension>
>>>>> </include>
>>>>>
>>>>> So, basically, how can I route the incoming SMS to peer 1000?
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Official FreeSWITCH Sites
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>>>>>
>>>>
>>>>
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>>>>
>>>>
>>>>
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>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>>
>>
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>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 6 Feb 2014 08:02:03 -0500
> Subject: Re: [Freeswitch-users] Filler Music in Call Recordings
> You would have to make a media bug similar to the one for record session,
> that has additional api commands to target the session by uuid that changes
> a flag in a private struct of weather the recording is paused or not, and
> then inside the callback for the recording, trigger different behavior when
> its paused. As there is no audio being muxed, there is no reason at all to
> use mod_conference.
>
> On Feb 5, 2014, at 7:42 PM, JP <jaykris at gmail.com> wrote:
>
> I'm trying to implement a feature that allows a user to selectively pause
> and resume a call recording via FreeSWITCH's event socket interface. The
> audio file should contain filler music for the duration of each pause
> operation. Is there any simple straight forward way to accomplish this? I
> would prefer avoiding the overhead incurred by using the conferencing
> module for this if possible.
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 6 Feb 2014 08:04:04 -0500
> Subject: Re: [Freeswitch-users] Hold Music not Playing when pressing Hold
> on Polycom Phone
> I don't see the phone actually sending the hold signal there, try turning
> on sip trace and confirming what the phone is actually sending. Compare
> that to your dev environment where it works.
>
> Mike
>
> On Feb 6, 2014, at 12:11 AM, Lyle Pratt <lylepratt at gmail.com> wrote:
>
> Hi Guys,
>
> When I press the HOLD button (or start to transfer someone, join a
> conference, etc) the configured hold music will not play.
>
> I previously had this working in production, but all of the sudden it is
> no longer working I can't figure out why. It works fine in my development
> environment on my local network, but not when I'm registering to Public
> IPs. I first thought it was a Polycom specific NAT issue, but then I tried
> it with a different phone (Cisco) from a different network with no success
> either.
>
> Here is a Pastebin of the Freeswitch console logs when I press the hold
> button on my device and no hold music occurs.
> http://pastebin.com/p6xtwyz7
>
> Does anyone have any ideas on how on earth I messed this up? Its the last
> piece of the puzzle I have in completely getting this working! Any help
> appreciated
>
>
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