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Call quality statistics from CDR reports?
by d.campbell 18 Dec '13

18 Dec '13

Greetings freeswitch community, Freeswitch is an amazing piece of work, and I have been able to use it with great success in large-scale scenarios. Much thanks to all the contributors! One feature I'm not finding is a report on call quality for SIP/RTP calls. I see that the CDRs contain several potentially relevant values including: rtp_audio_in_jb_packet_count rtp_audio_in_flush_packet_count rtp_audio_in_largest_jb_size Unfortunately, there is no documentation on exactly what these values mean. Would it be possible to compute packet loss and jitter statistics on the inbound call from these numbers or others? Much thanks, Doug Campbell

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overriding default timer values
by Grant Bagdasarian 18 Dec '13

18 Dec '13

Hello, The SIP RFC defines some timers and their default values. Is it possible to override these values in sip profiles or somewhere else? T1 500ms default Section 17.1.1.1 RTT Estimate T2 4s Section 17.1.2.2 The maximum retransmit interval for non-INVITE requests and INVITE responses T4 5s Section 17.1.2.2 Maximum duration a message will remain in the network Timer A initially T1 Section 17.1.1.2 INVITE request retransmit interval, for UDP only Timer B 64*T1 Section 17.1.1.2 INVITE transaction timeout timer Timer C > 3min Section 16.6 proxy INVITE transaction bullet 11 timeout Timer D > 32s for UDP Section 17.1.1.2 Wait time for response 0s for TCP/SCTP retransmits Timer E initially T1 Section 17.1.2.2 non-INVITE request retransmit interval, UDP only Timer F 64*T1 Section 17.1.2.2 non-INVITE transaction timeout timer Timer G initially T1 Section 17.2.1 INVITE response retransmit interval Timer H 64*T1 Section 17.2.1 Wait time for ACK receipt Timer I T4 for UDP Section 17.2.1 Wait time for 0s for TCP/SCTP ACK retransmits Timer J 64*T1 for UDP Section 17.2.2 Wait time for 0s for TCP/SCTP non-INVITE request retransmits Timer K T4 for UDP Section 17.1.2.2 Wait time for 0s for TCP/SCTP response retransmits

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Restrict SIP registration
by mehroz 18 Dec '13

18 Dec '13

Hi All, We already know that we can allow/deny a SIP end point to get registered to FreeSWITCH on the basis of IP , but what about if we want it to filter "User-Agent" field in REGISTER packets to be the key for registration. Since , we can also use TLS based registration but some soft phones does not have that capability. In that case , is there any way of passing only specified USER AGENT SIP client? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Restrict-SIP-registration-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Conference "kick all"
by Mike Kane 18 Dec '13

18 Dec '13

Hello, I'm having an issue with conference "kick all". When I release a call before the joining member joins the conference, the conference room does not send a canel or bye to the joining member that hasn't answered yet. It will kick all members currently joined in the conference room. Is is functionality supported, or will kick all just react to calls already in progress in the conference room? Thanks Mike

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FW: how to dial from different gateways according prefix numbers?
by netandtech 18 Dec '13

18 Dec '13

From: netandtech [mailto:bill_vus@hotmail.com] Sent: Wednesday, December 18, 2013 5:27 PM To: freeswitch Subject: how to dial from different gateways according prefix numbers? The following is the extension number that I use in the default.xml What I want to achieve is to Route the calls for landlines to different gateways and calls to mobiles from another gateway. <extension name="1002">   <condition field="destination_number" expression="^69(\d+)$"> <action application="transfer" data="sofia/gateway/myvoip/$0"/> </condition> <condition field="destination_number" expression="^22810(\d+)$"/> <action application="transfer" data="sofia/gateway/voipdito2/$0"/> </extension>

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H323 Configuration
by Richard Mace 18 Dec '13

18 Dec '13

Hi, Within the h323.config.xml file there is an entry for: <listeners> <listener name="default"> <param name="h323-ip" value="$${local_ip_v4}"/> <param name="h323-port" value="1720"/> </listener> </listeners> is the "h323-ip" value the IP address of the H323 server? Richard

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Freeswitch channel is down but stays alive
by Tornike Zedginidze 18 Dec '13

18 Dec '13

Hi, In very rare case, some channels stay active till I execute uuid_session_heartbeat or any uuid command. I run Freeswitch 1.2.14 and there is an LUA listener handling calls' dialplans and hangups. In approximately 12 000 calls, 1 channel stays alive even it's state is down. You can see CLI output here: http://pastebin.com/Z8ycPX3g What do you suggest, where can the issue be? Also not only hearbeat, but any uuid command finishes the call. Thanks

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FS sending INFO requests
by Grant Bagdasarian 17 Dec '13

17 Dec '13

Hello, Why does FS keep sending INFO requests? How do I disable these? Regards, Grant

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number translation in freeswitch
by Prakash Ganesan 17 Dec '13

17 Dec '13

Hi, Is it possible to do a 1 to 1 mapping in a database in free switch and get the number translated?. for example 123456789 is mapped to 987654321. if an INVITE comes with 123456789 to FS, it should translate to 987654321 in RURI and To header before it sends the invite out. this trranslation should be for 100,000 number present in the database? is it a vaible solution in FS? Many thanks Prakash

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Fwd: mod_nibble hangup problems
by Jason Sia 17 Dec '13

17 Dec '13

Hi, When the call is cut off by mod_nibble it sends a hangup signal. However, it is resulting to a multiple re invite. Is there a misconfiguration somewhere? I just copied the examples from the user guide. Thanks, Jason

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