April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Nibblebill for NoSQL
by Dome Charoenyost 22 Apr '10

22 Apr '10

Dear All, I use nibllebill + postgresql for longtime. it' work wel when concurrent about 500 calls.. Now i have 1200 calls . postgresql got many problem and slow. so i have 2 choice. 1. upgrade hardware , add more memory may be 32 GB and change HDD to SSD (or IOdrive ). 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant) I choose 2. and use tokyo tyrant. because it's persistant and support memcache API. Today i create new module base on nibblebill but use memcacahe API connect to tokyo tyrant. it's look fast. and work without any problem. If someone want to use my module I'l acttach to JIRA and create WIKI page. Please let's me know if you want. BG Dome C.

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 46, Issue 281
by Michael De Lorenzo 22 Apr '10

22 Apr '10

Yes I mean reload mod_xml_cdr. I get a success response when I do it. From: João Mesquita jmesquita(a)freeswitch.orgTo: freeswitch-users(a)lists.freeswitch.orgDate: Thu, 22 Apr 2010 18:51:59Subject: Re: [Freeswitch-users] Mod xml cdrYou mean reload mod_xml_cdr, right? JMOn Thu, Apr 22, 2010 at 4:41 PM, Michael De Lorenzo ‪<delorenzodesign(a)gmail.com>‬ wrote:I have mod_xml_cdr installed but I can't seem to get it to make the HTTP post. I don't see any attempts either in the console or in my Fresswitch log about it attempting, nevermind failing. I've tried reloading the module from the console via "reload_xml_cdr" and then processed a call and still nothing. Here's my xml_cdr config:<configuration name="xml_cdr.conf" description="XML CDR CURL logger"> <settings> <param name="url" value="http://url_path/save_cdr" /> <param name="retries" value="5"/> <param name="delay" value="120"/> <param name="log-dir" value="/var/log/cdr"/> <param name="err-log-dir" value="/var/log/cdr/errors"/> <param name="encode" value="base64"/> </settings></configuration>I've also verified that the module is enabled in the modules.conf.xml.Can anyone point me in the right direction? On 4/22/10, freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." >

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Mod xml cdr
by Michael De Lorenzo 22 Apr '10

22 Apr '10

I have mod_xml_cdr installed but I can't seem to get it to make the HTTP post. I don't see any attempts either in the console or in my Fresswitch log about it attempting, nevermind failing. I've tried reloading the module from the console via "reload_xml_cdr" and then processed a call and still nothing. Here's my xml_cdr config: <configuration name="xml_cdr.conf" description="XML CDR CURL logger"> <settings>  <param name="url" value="http://url_path/save_cdr" /> <param name="retries" value="5"/> <param name="delay" value="120"/> <param name="log-dir" value="/var/log/cdr"/> <param name="err-log-dir" value="/var/log/cdr/errors"/> <param name="encode" value="base64"/> </settings> </configuration> I've also verified that the module is enabled in the modules.conf.xml. Can anyone point me in the right direction?

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passing originating IP when configured as a B2BUA
by RR 22 Apr '10

22 Apr '10

Hello list, we have an FS server configured as a call-distributor (using "mod_distributor") to a bank/farm of Asterisk servers. However, since the FS is acting like a B2BUA, it's not passing the originating IP (network_addr) from the originating switch/gateway to the Asterisk farm (as expected). However, we need to have this information in the Asterisk servers. How can we achieve this? Do we need to create/insert a custom SIP header when passing on the call or set a variable or any other way? Any help on the issue will be much appreciated Thanks in advance, RR

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reloadxml not reloading gateway configuration
by RR 22 Apr '10

22 Apr '10

Hello, using version freeswitch@internal> version FreeSWITCH Version 1.0.5pre9 (hacked) and when I change the configuration of a gateway under conf/sip_profiles/external/gateway2.xml configuration and then do a reloadxml, nothing changes within the freeswitch db as seen from 'sofia status'. Is this a bug and has been fixed in newer versions? Also, does doing an 'fs_ctl send_sighup' drop all calls from the system? would doing that reload the gateway configurations? Thanks \RR

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Different SIP profiles for different codecs?
by Ken Fulmer 22 Apr '10

22 Apr '10

We are using the following external sip profile: <profile name="external"> <settings> <param name="accept-blind-auth" value="true"/> <param name="apply-inbound-acl" value="domains"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5080"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="G729,PCMU"/> <param name="outbound-codec-prefs" value="G729,PCMU"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="10.10.3.12"/> <param name="sip-ip" value="10.10.3.12"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="user-agent-string" value="INTEGRAL PARTNERS"/> </settings> </profile> The dial plan "public" context has two entries, one for PSTN access and one for call routing to an internal PBX. This is the dial plan for PSTN call routing: <extension name="LCL"> <condition field="destination_number" expression="^(\d{10})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$1(a)172.16.15.11"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="ignore_early_media=true" /> </condition> </extension> <extension name="LD"> <condition field="destination_number" expression="^1(\d{10})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$0(a)172.16.15.11"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="ignore_early_media=true" /> </condition> </extension> <extension name="911"> <condition field="destination_number" expression="^(911)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$1(a)172.16.15.11"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="ignore_early_media=true" /> </condition> </extension> <extension name="INTL"> <condition field="destination_number" expression="^011(\d+)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$0(a)172.16.15.11"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="ignore_early_media=true" /> </condition> </extension> And this is the entry for internal call routing to a PBX: <extension name="PBX"> <condition field="destination_number" expression="^(205314849[0-9])$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$1(a)10.10.3.10"|data="sofia/external/$1(a)10.10.3.11"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="ignore_early_media=true"/> </condition> </extension> So, here's my question: We'd like to be able to lock down the codec as 711 for the internal leg going to the PBX and 729 for the external leg to the PSTN. We have transcoding setup and it's working fine. How can we use two SIP profiles to hard code the codec in each direction? I've seen an example in the dial plan section, but didn't understand how to implement it. Thanks, Ken Fulmer

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SIP Agent (Android SIP client)
by Phillip Jones 22 Apr '10

22 Apr '10

Hi there, Has anyone got the android app SIP Agent working with FreeSWITCH. I have the app saying it is registered correctly, but when I make a call it is in the public context and not default. I just wondered whether anyone has had any success. thanks pj (PS: I have sipDroid working very nicely and iSIP on the iPhone also)

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Re: [Freeswitch-users] Windows Build of latest git version
by Peter Olsson 22 Apr '10

22 Apr '10

When using git in Windows you must make sure to set core.autocrlf=false in the git configuration. The suggested way when installing is true, and it causes problems with some LF to CRLF conversions. /Peter ________________________________ Från: Phillip Jones <pjintheusa(a)gmail.com> Skickat: den 22 april 2010 00:16 Till: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Ämne: [Freeswitch-users] Windows Build of latest git version Hi All, I just downloaded the lasted version from git, and I am having real problems compiling in VS2008 for Windows. FreeSwithCoreLib, mod_PortAudio and others are fine, but when building libsofia_sip_ua_static I get 905 errors, eg: Error 601 error C2065: 'sip_supported_hash' : undeclared identifier c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c 279 libsofia_sip_ua_static Error 602 error C2099: initializer is not a constant c:\Projects\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sip_feature.c 279 libsofia_sip_ua_static etc etc I have previously compiled successfully many times before, although not on this machine. Anyone else experience this and know how to fix? TIA Phil !DSPAM:4bcf792732932989516006!

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$25 - Robust Affordable Speech Recognition
by Kashif Kahn 22 Apr '10

22 Apr '10

Dear All, All those who have wanted to try speech recognition with Freeswitch but found the software cost too expensive or the recognition accuracy unsatisfactory, I encourage you to try our Vestec Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license for the standard engine - is available for only $25. Additional ports (channels) licenses can be purchased for $99/port. Of course, the engine comes with a Freeswitch connector, thereby allowing a Freeswitch user to bypass engine API and interact directly via Dialplan. Best regards, -Kashif Kashif Kahn VP, Business Development Vestec, Inc. Waterloo, ON Canada phone: (519) 885-7615

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Stopping an audio file playing in a XML IVR
by Alberto Escudero 22 Apr '10

22 Apr '10

Hi, Is there any way to stop/exit/jumpo out from the play-file function inside of a XML IVR? <action function="play-file" data="content.wav"/> /aep -- Stopping junk mailers is good for the environment

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