[Freeswitch-users] Possible to disable core codecs?

Peter Olsson peter.olsson at visionutveckling.se
Fri Apr 13 10:06:59 MSD 2012 

Just pipe it to a file. fs_cli > file.txt.

However, without seeing any logs there are really only two options, the remote node doesn't offer GSM codec, or GSM codec is not loaded in FS (IIRC mod_spandsp handles this codec).

/Peter

13 apr 2012 kl. 07:45 skrev "Malay Thakershi" <mthakershi at gmail.com>:

> When I do "sofia global siptrace on", my console just doesn't stop. I
> get text so fast, I can't even paste the originate command. So I can't
> produce what will hep you help me on this.
> 
> Is there a way I can divert the whole thing to a text file?
> 
> Thanks.
> 
> Malay
> 
> On Thu, Apr 12, 2012 at 10:55 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:
>> Could you collect the full siptrace of one call and paste into pastebin?
>> 
>> To do so, type at the cli:
>> 
>>        sofia global siptrace on
>> 
>> Then make 1 call and collect whatever you got at the screen, and paste into
>> pastebin.
>> 
>> Then, you'll probably want to disable siptrace:
>> 
>>         sofia global siptrace off
>> 
>> Regards,
>> Vitalie
>> 
>> 
>> 2012/4/12 Malay Thakershi <mthakershi at gmail.com>
>>> 
>>> Thanks.
>>> 
>>> Before I do what you said, I have another problem. Sorry about that.
>>> 
>>> I use different mechanism to make outgoing calls (originate calls from
>>> ESL). So I do not use bridge.
>>> I tried to put "{absolute_codec_string=GSM}" in originate call along
>>> with other parameters I pass but it throws error.
>>> 
>>> My originate call looks like:
>>> originate
>>> {absolute_codec_string=GSM,ignore_early_media=true,origination_caller_id_name=CH,origination_caller_id_number=1877XXXXXXX,originate_timeout=45,chivr_outbound=true,other_var=1}sofia/gateway/vitelity-outbound/1972XXXXXXX
>>> 1056 XML default
>>> 
>>> Error thrown on the log: -ERR INCOMPATIBLE_DESTINATION
>>> 2012-04-12 19:14:01.589739 [NOTICE] sofia.c:6332 Hangup
>>> sofia/sipinterface_1/1972XXXXXXX [CS_CONSUME_MEDIA]
>>> [INCOMPATIBLE_DESTINATION]
>>> 
>>> Call without absolute_codec_string works fine.
>>> 
>>> The way this works is: call gets originated -- it goes to default dial
>>> plan extension 1056 -- mod_managed is called from this point to handle
>>> call flow.
>>> 
>>>    <extension name="CH1">
>>>          <!-- Calls from public.xml will come with 1056 extension and will
>>> be handled here -->
>>>      <condition field="destination_number" expression="^1056$">
>>>        <!-- codec trial setting -->
>>>        <!-- <action application="set" data="absolute_codec_string=GSM"/>
>>> -->
>>> <!--            <action application="log" data="codec debug: codec_string
>>> is
>>> ${codec_string}" />
>>>                <action application="log" data="codec debug:
>>> originator_codec is
>>> ${originator_codec}" />
>>>                <action application="log" data="codec debug:
>>> absolute_codec_string
>>> is ${absolute_codec_string}" />
>>>  -->
>>>                <action application="sleep" data="1000" />
>>>        <action application="managed" data="clsAsmtApp" />
>>>      </condition>
>>>    </extension>
>>> 
>>> As you can see, I tried setting absolute string from here too but it
>>> didn't work.
>>> 
>>> Let me know what can be done if you can.
>>> 
>>> Thanks.
>>> 
>>> Malay
>>> 
>>> 
>>> On Wed, Apr 11, 2012 at 5:48 PM, Vitalie Colosov <vetali100 at gmail.com>
>>> wrote:
>>>> Lets forget for a minute about vars.xml and what is there (when you use
>>>> absolute_codec_string=GSM, it overrides any settings from vars.xml), and
>>>> concentrate at the OUTGOING call from FreeSWITCH to your sip provider
>>>> first.
>>>> 
>>>> Try to change your bridge to the following string:
>>>> <action application="bridge">
>>>> 
>>>> data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/>
>>>> 
>>>> (obviously use your sip provider and phone number :)
>>>> 
>>>> And do one outgoing call to this sip provider.
>>>> 
>>>> Then, reply back whether the call was successful or not (did you hear
>>>> the
>>>> voice and whether the other party did hear you). And paste the log to
>>>> the
>>>> pastebin (http://pastebin.freeswitch.org/) and reply back with the link.
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>>>>> 
>>>>>> 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
>>>>>> use
>>>>>> G729 right now as it needs the license). By OUTGOING I mean call from
>>>>>> FreeSWITCH to your provider, and not back.
>>>>> Yes. I want to try some codec that will use less bandwidth and see how
>>>>> much audio quality is compromised. So I chose GSM because G729 has
>>>>> license requirements.
>>>>> 
>>>>> I want this to work when someone calls my DID or my FS program calls
>>>>> someone via DID. So both for incoming calls as well as outgoing calls.
>>>>> 
>>>>>> 2. If #1 is true, then inbound-codec-negotiation is not much useful
>>>>>> at
>>>>>> this point. Because this setting is for INCOMING leg, and not for
>>>>>> OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but
>>>>>> still
>>>>>> transcode to the OUTGOING PCMA/PCMU by some reason (this is what we
>>>>>> need
>>>>>> to
>>>>>> find out - why).
>>>>> I am not very familiar with internals of FS so having lot of trouble
>>>>> figuring this out.
>>>>> 
>>>>> Weird thing is, even if I do not have any mention of G729, it is being
>>>>> negotiated and used. Here are my configuration lines:
>>>>> vars.xml
>>>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
>>>>>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>>>>> 
>>>>> sip profile has:
>>>>> <param name="inbound-codec-negotiation" value="greedy"/>
>>>>> 
>>>>> dial plan
>>>>>        <extension name="CH">
>>>>>      <condition field="destination_number" expression="^105\d$">
>>>>>                <action application="set"
>>>>> data="absolute_codec_string=GSM"/>
>>>>>                <action application="sleep" data="1000"/>
>>>>>                <action application="managed" data="clsAsmtApp"/>
>>>>>      </condition>
>>>>>          </extension>
>>>>> 
>>>>> 
>>>>>  <extension name="vitel-outbound">
>>>>>   <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>         <action application="set" data="absolute_codec_string=GSM"/>
>>>>>     <action application="set"
>>>>> data="effective_caller_id_number=18770001111"/>
>>>>>     <action application="bridge"
>>>>> data="sofia/gateway/vitelity-outbound/$1"/>
>>>>>    </condition>
>>>>>  </extension>
>>>>> 
>>>>>> 3. Try using absolute_codec_string before bridging your call to your
>>>>>> SIP
>>>>>> provider:
>>>>>> 
>>>>>> <action application="set" data="absolute_codec_string=GSM"/>
>>>>>> <action application="bridge"
>>>>>> data="sofia/gateway/your_provider/12345678901_example"/>
>>>>>> 
>>>>> I did as you said. But found what I described above. If I remove
>>>>> "greedy" setting, system simply uses PCMU codec like before.
>>>>> 
>>>>> This thing is so confusing. Even my SIP provider is saying they can't
>>>>> help out with FS because some guy quit who knew the configurations.
>>>>> 
>>>>> Thanks for help.
>>>>> 
>>>>> On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100 at gmail.com>
>>>>> wrote:
>>>>>> 
>>>>>> Lets confirm the following statements now:
>>>>>> 
>>>>>> 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
>>>>>> use
>>>>>> G729 right now as it needs the license). By OUTGOING I mean call from
>>>>>> FreeSWITCH to your provider, and not back.
>>>>>> 
>>>>>> 2. If #1 is true, then inbound-codec-negotiation is not much useful
>>>>>> at
>>>>>> this point. Because this setting is for INCOMING leg, and not for
>>>>>> OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but
>>>>>> still
>>>>>> transcode to the OUTGOING PCMA/PCMU by some reason (this is what we
>>>>>> need
>>>>>> to
>>>>>> find out - why).
>>>>>> 
>>>>>> 3. Try using absolute_codec_string before bridging your call to your
>>>>>> SIP
>>>>>> provider:
>>>>>> 
>>>>>> <action application="set" data="absolute_codec_string=GSM"/>
>>>>>> <action application="bridge"
>>>>>> data="sofia/gateway/your_provider/12345678901_example"/>
>>>>>> 
>>>>>> 
>>>>>> And reply back if this changes anything...
>>>>>> 
>>>>>> Vitalie
>>>>>> 
>>>>>> 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>>>>>>> 
>>>>>>> Under my sipinterface_1 profile that I use for these calls, I have:
>>>>>>> 
>>>>>>> <param name="inbound-codec-negotiation" value="greedy"/>
>>>>>>> 
>>>>>>> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:
>>>>>>>> 
>>>>>>>> On Wed, 11 Apr 2012 13:31:45 -0500
>>>>>>>> Malay Thakershi <mthakershi at gmail.com> wrote:
>>>>>>>> 
>>>>>>>>> Understood your point about G722. But if my list has GSM at
>>>>>>>>> higher
>>>>>>>>> priority
>>>>>>>>> over PCMU/PCMA and far-end SDP supports GSM then why isn't the
>>>>>>>>> call
>>>>>>>>> using
>>>>>>>>> GSM?
>>>>>>>>> 
>>>>>>>>> I don't know where G726 comparisons are coming from. If I haven't
>>>>>>>>> mentioned
>>>>>>>>> it in vars.xml and far-end SDP doesn't support it then why would
>>>>>>>>> it
>>>>>>>>> even
>>>>>>>>> consider that codec?
>>>>>>>> 
>>>>>>>> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>>>>>>>> (inbound-codec-negotiation):
>>>>>>>> 'generous' permits the remote codec list have precedence and 'win'
>>>>>>>>   the codec negotiation and selection process
>>>>>>>> 
>>>>>>>> JY
>>>>>>>> --
>>>>>>>> Don't drop acid -- take it pass/fail.
>>>>>>>>                -- Seen in a Ladies' Room at Harvard
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
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>>>>>>>> 
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>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
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>>>>>>> 
>>>>>>> 
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>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
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>>>>>> 
>>>>>> 
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>>>>> 
>>>>> 
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>>>>> 
>>>>> 
>>>>> 
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>>>> 
>>>> 
>>>> 
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>> 
>> 
>> 
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>> 
>> 
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> 
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