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@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@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
It looks like you are missing the sip-ip and rtp-ip settings. Also, if you are using the same ip for both profiles, you will need to use different ports. I can't say without knowing more of what you need to do, but you probably won't need separate profiles for the different codecs, those can negotiate with the end device. Mike
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Bernhard Suttner Sent: Wednesday, January 10, 2007 10:15 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] freeswitch with different codecs
<snip>
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@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@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
The two telephones have different codecs and FreeSWITCH have to transcode them. So I have created two profiles. Does FreeSWITCH create for every profile one server or is it possible to tell FreeSWITCH that these profiles are only client profiles and used to call other telephones? Maybe the configuration of the local telephones is wrong. I thought, if the phones are registered on FreeSWITCH they can establish a local call to the peer. They are registered to FreeSWITCH with the following profile: <profile name="local"> <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@20ms"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="10.10.1.1"/> <param name="sip-ip" value="10.10.1.1"/> <param name="accept-blind-reg" value="true"/> </settings> </profile> In the Example 5 on http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML there is a similar case. Kind regards, Bernhard Suttner On Wed, 10 Jan 2007 11:48:46 -0500 "Michael Jerris" <mike@jerris.com> wrote:
It looks like you are missing the sip-ip and rtp-ip settings. Also, if you are using the same ip for both profiles, you will need to use different ports. I can't say without knowing more of what you need to do, but you probably won't need separate profiles for the different codecs, those can negotiate with the end device.
Mike
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Bernhard Suttner Sent: Wednesday, January 10, 2007 10:15 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] freeswitch with different codecs
<snip>
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@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@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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participants (2)
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Bernhard Suttner
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Michael Jerris