February 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Force G729 codec for E1 connection
by William Alianto 08 Feb '13

08 Feb '13

Hi, I just finished a freeswitch server that act as E1 gateway for PBX system. I noticed that all the call from the gateway use PCMU codec before transcoded to G729 in my main SIP server. My current situation is as following : [PBX] --- E1 ---> [Gateway] --- SIP/PCMU ---> [SIP Server] --- G729 ---> [ITSP] What I intended to do is as following : [PBX] --- E1 ---> [Gateway] --- SIP/G729 ---> [SIP Server] --- G729 ---> [ITSP] Is there anyway to send the media from E1 gateway as G729 instead of PCMU? I need to preserve bandwidth for future expansion. Regards, William

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call_queu
by Archana Venugopan 07 Feb '13

07 Feb '13

Hi, I have very basic doubt in call queue and sorry for that query. If I just put the below in dialplan xml and dial say 250 and there is other one who is also dialling 250 will the call be in queue. In /usr/local/freeswich/conf/autoload_configs, default callcenter.conf.xml file is also present. Please let me know if i have to change anything else too to make call in queue ? <extension name="call_queue"> <condition field="destination_number" expression="250" break="on-false"/>  <condition wday="2-6" time-of-day="09:00-17:00" break="on-true">  <action application="limit" data="hash inbound ${destination_number} 3/1 !NORMAL_CIRCUIT_CONGESTION"/>  <action application="playback" data="$${sounds_dir}/greeting.wav"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="callcenter" data="queue@default"/>  <action application="playback" data="$${sounds_dir}/queue_voicemail.wav"/> <action application="answer"/> <action application="set" data="skip_greeting=true"/> <action application="set" data="skip_instructions=true"/> <action application="voicemail" data="default voicemail 2001"/> <action application="hangup"/> </condition> <condition> <action application="limit" data="hash inbound ${destination_number} 3/1 !NORMAL_CIRCUIT_CONGESTION"/>  <action application="playback" data="$${sounds_dir}/tod_voicemail.wav"/> <action application="answer"/> <action application="set" data="skip_greeting=true"/> <action application="set" data="skip_instructions=true"/> <action application="voicemail" data="default voicemail 2001"/> <action application="hangup"/> </condition> </extension> Regards, Archana

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Use local IPv4 for specific LAN IP address range
by Ben 07 Feb '13

07 Feb '13

Greetings all, A question on NATing in Freeswitch - I have my internal SIP profile to use a static public IP for the ext-sip-ip and ext-rtp-ip. The Freeswitch server is on a local LAN IP, for example 192.168.0.10. External SIP clients can access the server via the external IP, as there's a static NAT on the router to map the public IP to the private IP. I have another SIP client on a different LAN, but behind the same firewall, for example, 10.0.0.10. This client is allowed to talk to the server on its private ip of 192.168.0.10, by allowing inter-vlan traffic to pass through. My challenge is, how do I get Freeswitch to recognise this internal SIP client as being allowed to use the internal IP for SIP/RTP communications, and therefore present the client with internal addresses in the SIP header for SIP/RTP? Due to statically setting ext-sip-ip and ext-rtp-ip, Freeswitch tells the client to go via the public IP. The firewall has a policy to not accept traffic to the public IP if sourced from an internal device. I have explored the NATing options in Freeswitch, but can't find anything that fits my scenario. It seems to be more towards how to enable NAT in certain situations, not disable it. I have also looked at re-writing the SDP, but this re-writes the incoming SDP from the clients. My intention was to re-write the outgoing SDP from the Freeswitch to include the internal IP address as the RTP connect in address. I don't have much Asterisk experience, but I seem to remember a nat.conf file where you could specify subnets that were treated differently. These could be classed as internal subnets, and would be given the private IP in the SDP from asterisk, everything got the public IP. I'm also aware that I could create another SIP profile easily with different NATing/IP options, but I want to find out if there's anything I can do with the above first. Many thanks for the help!

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Log Rollover issue
by Nageshwara Rao Moova 07 Feb '13

07 Feb '13

Hi all, I have modified my default logconf file for rollover to be restricted to 100. But sometimes freeswitch is unable to rename rollover say “cannot rename freeswitch.log.41” and fails. But the fail follows with serious issue of writing all the logs to "freeswitch.log" and ends up filling the disk space. I have not change the default file size i.e 10MB. How does 10MB size overridden by freeswitch? -- regards & thanks -- m nageshwara rao 99891 86280

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FreeSWITCH git repo undergoing maintenance
by Ken Rice 07 Feb '13

07 Feb '13

Hey guys, Yes we know git is hosed up right now... We¹re taking the time to do some clean up and upgrades to the GIT server... Once it comes back online there will be an announcement to this. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Passthrough Calling
by mahesh katta 06 Feb '13

06 Feb '13

Hi List, I am new to this freeswitch, Any body know regarding "passthrough". for example i get call from first ip and passthrough to second ip using media proxy. like: IP====>FS Server====>IP I have GATEWAY: xxx.xxx.xxx.xxx Thanking you. Best Regards, Mahesh Katta

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Freeswitch second internal profile
by taube 06 Feb '13

06 Feb '13

Hi all, I created a second internal profile to connect with the phones to freeswitch via two networks. The internal profile works. I can call out -in and can contact the extension with fs_cli sofia_contact. At the second profile the phones register and i can contact them but when i try to call them, i get the message that the user is not registered, are the any hints when i create a second internal profile? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-second-internal-pr… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Understanding Codecs preferences
by mehroz 06 Feb '13

06 Feb '13

Hi, I need to assign seperate codec list for each directory e,g PCMU, h264 for 1000.xml PCMA , h263 for 2000.xml I see, codec preference is somthing that wil help me. But cant get this on head! Can any one help me step wise what do i need to do to acquire this scenario. By the way, ill be using mod_xml_curl to load codecs for each directory. I believe i can convert the scenario to fir with dymanic direcotry parameters with xml_curl once i get the way to do it with static XML files. If i am wrong, please do help. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Understanding-Codecs-preferen… Sent from the freeswitch-users mailing list archive at Nabble.com.

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lua freeswitch.Dbh sqlite error
by Carlo Dimaggio 06 Feb '13

06 Feb '13

Hi all! I have a lua script in which i have a statement that connects to a sqlite db: local dbh = freeswitch.Dbh("core:/usr/local/temp/database.db"); but i receve in FS CLI this error that refers to a Pgsqlinstead of sqlite: 2013-02-05 09:40:10.379612 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2013-02-05 09:40:10.379612 [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL core! 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:354 Connection failed. DBH NOT Connected. 2013-02-05 09:40:10.379612 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. Can you help me? Best regards,

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call queue
by Archana Venugopan 06 Feb '13

06 Feb '13

Hi, Can you please tell me how can I enable call queue in freeswitch? Regards, Archana

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