[Freeswitch-users] freeswitch with different codecs

[Bernhard Suttner]

Helo,

I have the following setting:

SIP-Telephone 100 --------> FreeSWITCH <--------- SIP-Telephone 101


Both telephones are registered on FreeSWITCH but the have different
codecs. SIP T-100 has GSM and SIP T-101 has PCMU. 

The FreeSWITCH server has the address 10.10.1.1 and the following
extension configuration: 

<extension name="100">
	<condition field="destination_number" expression="^100$"> 
		<action application="bridge" data="sofia/codecGSM/100%10.10.1.1"/>
	</condition>
</extension>

<extension name="101">
	<condition field="destination_number" expression="^101$"> 
		<action application="bridge" data="sofia/codecPCMU/101%10.10.1.1"/>
	</condition>
</extension>

The definition of the profiles look like this:

<profile name="codecGSM">
	<settings>
	        <param name="debug" value="1"/>
		<param name="rfc2833-pt" value="101"/>
		<param name="sip-port" value="5060"/>
		<param name="dialplan" value="XML"/>
		<param name="dtmf-duration" value="100"/>
		<param name="codec-prefs" value="GSM at 20ms"/>
		<param name="codec-ms" value="20"/>
		<param name="use-rtp-timer" value="true"/>
		<param name="rtp-timer-name" value="soft"/>
	</settings>
</profile>

<profile name="codecPCMU">
	<settings>
		<param name="debug" value="1"/>
		<param name="rfc2833-pt" value="101"/>
		<param name="sip-port" value="5060"/>
		<param name="dialplan" value="XML"/>
		<param name="dtmf-duration" value="100"/>
		<param name="codec-prefs" value="PCMU at 20ms"/>
		<param name="codec-ms" value="20"/>
		<param name="use-rtp-timer" value="true"/>
		<param name="rtp-timer-name" value="soft"/>
	</settings>
</profile>

If I start freeswitch I get the following error:
tport_server_addrinfo(0x809fb48): su_getaddrinfo((null), 5060) for *: Name or service not known
nta: bind((null):5060;transport=*): No such file or directory
nua: initializing SIP stack failed
tport_server_addrinfo(0x80a2db8): su_getaddrinfo((null), 5060) for *: Name or service not known
nta: bind((null):5060;transport=*): No such file or directory
nua: initializing SIP stack failed

How is it possible to handle telephones with different codecs? I thought this is possible with different profiles?

I am very pleased for every hint!

Kind regards,
Bernhard Suttner