June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

G.729 transcoder licensing
by Alex Lake 13 Jun '13

13 Jun '13

If we want to just have a play with G.729, can we buy just a couple of channels and then add to them later? And how does it cope with some channels? Presumably it stops offering G.729 when they're all in use? The other question is how does the G.729 licence bind itself to a system? Is this a hardware ID thing, or a Freeswitch ID? Does it work on VMs?

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fs_ivrd vs. outbound event socket
by Ali Pey 13 Jun '13

13 Jun '13

Hello, What are the differences between fs_ivrd and outbound event socket? Why would someone pick one vs. the other? Thanks, Ali Pey

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Mid Call Transfer
by Thiruvallar Raveendran 13 Jun '13

13 Jun '13

Hi everyone, What is the best way to implement below scenario in freeswitch? I have used att_xfer to transfer the calls from B to C. I couldn't implement the below numeric key action in receiver side with att_xref. You help on this issue will be much appreciated. Midcall Transfer Party A - The Caller Party B - The First Receiver Party C - The Second Receiver In a typical scenario, Party A dials a DDI and Party B receives the call. The Midcall Transfer control is now available to Party B. Options: If Party B presses the hash key (#) Party A will be placed on hold and holding music will be played. The hold will be cancelled if Party B presses the star key (*). While Party A is on hold, Party B may dial the number of Party C. After Party C answers the call, Party B can use the numerical keys for the following options: . 1 Key - disconnects Party B and connects Parties A and C . 2 Key - disconnects Party C allowing Party B to dial a new destination (to dial an alternative Party C) . 3 Key - disconnects Party C and reconnects with Party A . 4 Key - toggles between Parties A and C Thanks, Cheers, Thiru

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LUA XML libraries
by Alex Lake 13 Jun '13

13 Jun '13

What LUA XML libraries are people currently using here? I'm interested in simply parsing and manipulating XML documents. Don't really need xpath or suchlike.

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Event Socket Library (ESL) Perl and Yum (Centos)
by Ali Pey 13 Jun '13

13 Jun '13

Hello, Can I install freeswitch on Centos with yum and still get the perl ESL libraries? If not, how can I add ESL perl libraries to an already yum installed freeswitch version 1.2.9? Thanks, Ali Pey

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Is ODBC compiled in or not?
by Alex Lake 13 Jun '13

13 Jun '13

Is there a way to tell if ODBC has been compiled into the FS core? Maybe there's a log file left over from the make somewhere?

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Unable to secure b-leg to user
by Peter Waldheim 13 Jun '13

13 Jun '13

Dear list, I am trying to configure SRTP. This seems to work fine for calls from a local sip user. Clients are registered via TLS and outbound-call a-legs use SRTP. The problem is to secure the call to local sip users, coming in via openzap in this case. On this call's b-leg, I am unable to make freeswitch offer SAVP (in this case then leading to bria refusing the call due to being configured to accept secure calls only). The local sdp: 2013-06-13 13:13:50.040774 [DEBUG] sofia_glue.c:1219 Local SDP: v=0 o=FreeSWITCH 1371101270 1371101271 IN IP4 out.ip.address s=FreeSWITCH c=IN IP4 our.ip.address t=0 0 m=audio 20760 RTP/AVP 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 19462 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000 In the dialplan I tried combinations of these: <action application="set" data="sip_secure_media=true" /> <action application="set" data="sdp_secure_savp_only=true" /> <action application="export" data="sip_secure_media=true" /> <action application="export" data="sdp_secure_savp_only=true" /> <action application="bridge" data="{local_var_clobber=true,sip_secure_media=true}user/$1(a)our.profile"/> Also I tried setting the sip_secure_media variable in the user's profile. Any hints on how to make this work (or further debug) would be greatly appreciated. Best regards, Peter

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G.719
by Brian West 13 Jun '13

13 Jun '13

Anyone have any endpoints to test this codec with? email me off list please. FYI its a 48kHz full band codec. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest

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XML_CURL max connection limit
by Muhammad Shahzad 13 Jun '13

13 Jun '13

Hi, Is there any connection limit perhaps hard coded anywhere in xml_curl module? I am load testing FS with xml_curl to provide config, directory and dial-plan. I observed i start getting HTTP ERROR 0 when i have around 1600 concurrent calls on machine with media. Although i see no sign of system overload either on FS (load average < 1.8) or Apache (load average < 4,75). Both have plenty of RAM and bandwidth free. Is there anyway to get rid of this HTTP ERROR 0? It must be generated by FS internally, since i don't see anything like this error in Apache logs. Below 1,638 concurrent calls @ 50 CPS, i get success rate of 99.999% but as soon as it crosses this mark, success rate drastically drops to 60% and even below (when concurrent calls go beyond 2,000). Here is error console log. ################################################### 2013-06-05 18:09:13.591229 [ERR] mod_xml_curl.c:310 Received HTTP error 0 trying to fetch http://172.16.171.61/vm-users.php data: [hostname=vms01&section=directory&tag_name=domain&key_name=name&key_value=10.1.3.2&Event-Name=GENERAL&Core-UUID=84c917dc-cdde-11e2-9bb2-8b779d6a4e97&FreeSWITCH-Hostname=vms01&FreeSWITCH-Switchname=vms01&FreeSWITCH-IPv4=10.1.3.2&FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2013-06-05%2018%3A09%3A03&Event-Date-GMT=Wed,%2005%20Jun%202013%2013%3A09%3A03%20GMT&Event-Date-Timestamp=1370437743591210&Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id&Event-Calling-Line-Number=1350&Event-Sequence=1776930&action=message-count&key=id&user=989000049598&domain=10.1.3.2] ################################################### Although this setup is more then sufficient for my requirement but i just want to know if i can extend it beyond, since i still have lot of resources free. OS: Debian 6.0.7 x64 Apache: 2.2.16 FreeSWITCH: 1.5.1b+git~20130424T005233Z~66d1755b06 Both FS and Apache are running on separate dedicated machines on same LAN. Thank you. -- Mit freundlichen Grüßen Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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Gateway Call Limits
by Alex Ynema 12 Jun '13

12 Jun '13

Hi, I'm trying to figure out how to limit the number of calls a Gateway is allowed to use. Our Sip provider has provided up with 200 which I need to set within the system somehow. What's the best way to handle it for an outgoing only system. I've been trying to figure out how how to configure 'Rate limiting concurrent outgoing calls via a provider' which is mentioned in the wiki a bit but nothing specific on what to actually do. *Alex Ynema** *| IT Consultant alex(a)opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 <http://www.ynema.net/> <http://www.ynema.net/>IT Consultant for Open Systems Support www.opensystems.net.au

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