[Freeswitch-users] freeswitch with different codecs
Bernhard Suttner
suttner at comdasys.com
Wed Jan 10 07:15:19 PST 2007
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
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