May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Mod_db deleting data from database
by info＠shishir.com.np 22 May '13

22 May '13

HI I am setting up the intercept feature, and I went through the wiki on how to accomplish it. The only question I have is once I inserted data on the db with <action application="db" data="insert/realm/key/value"/> how could I delete this data out of the db ? I see the <action application="db" data="delete/realm/key"/> will delete the record, is this the only options that I have to delete the record from the db ? If yes then how could I use it to delete the record when the call gets hangup? Do I have to have this action on every calls that get hangup? Thanks in advance.

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Using FreeSWITCH as a proxy
by Oleg Stolyar 22 May '13

22 May '13

Hi, I am trying to use FreeSWITCH as a SIP proxy. I have the dialplan below which simply sets bypass_media to true and then bridges to another FreeSWITCH server. However, when during the call I shut down the proxy FS, the call is immediately dropped. Why is that? Is there a way to keep it going? I understand that in this case I won't be able to properly send the BYE signal when one of the parties hangs up and that's OK. I tried using redirect and deflect instead of the bridge but those don't seem to work at all - probably because my UAs don't know how to handle redirects. <extension name="proxy_test"> <condition field="destination_number" expression="^(p\w+)$"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/external/1000(a)10.2.182.243 "/> </condition> </extension>

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FreeSWITCH Conference Call Today
by Michael Collins 22 May '13

22 May '13

Hello all. Our community conference call is coming up shortly. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22 We don't have a feature presentation today, but we do have several items to discuss so please join us at 1PM EDT, 10AM PDT. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Node.JS ESL libraries
by Alex Lake 22 May '13

22 May '13

Just beginning to get a bit more advanced with our use of Freeswitch, and we'd like to start integrating it's internal data and status with the outside world a bit more. The idea was to listen in on the event socket and use that information to update an external data model. Looking at the wiki(http://wiki.freeswitch.org/wiki/Event_Socket#Javascript_.2F_Node.js_li…, there seem to be 3 or 4 examples of how this is done, but I wondered if there's stuff that's not there or if one of these methods is proving more popular than the others. Any advice on where we should look first would be appreciated, cheers!

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One Way Audio
by Sean Devoy 22 May '13

22 May '13

Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean

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mod_cdr_csv is not working!
by Shahzad Bhatti 22 May '13

22 May '13

Hi, after loading *mod_cdr_csv, *when i originate test calls the cdr is not appended in Master.csv file most of the time but sometime cdr is appended in the Master.csv that show that mod_cdr_csv is configured. Also when i try to use *cdr_csv rotate* command on fs_cli it also not work. even i also try to do using the perl script example available on following url https://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_in… i want to know why this is happening on my server and how i can fix these issues 1. cdrs are not updated in the csv file; 2. rotate command is not working; any reply is highly appreciated; Regards Shahzad Bhatti

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Routing on the basis of RURI
by Lucas Tehbing 22 May '13

22 May '13

Dear Guys, I have been using Opensips as SBC but now want to move to Freeswitch because of Jitter buffer module and many other interesting features. My Call flow is .... SIP UA (Behind NAT) -------------> SBC (Currently opensips)----->Sip Provider (Configured at SIP UA) When SIP message is received at SBC, SBC checks if RURI belongs to some other domain (if(uri==myself) ) if yes it reads the RURI and forwards (State fully) the message (Register, Invite , BYE all kinds of messages). It also modifies SDP in order to be in media path. I want to implement FS in same scenario. The issue is that I cannot configure RURIs on FS, Freeswitch should read from received Request where to forward the message. I am still in the process to learn Freeswitch. If anyone can point me to right resource it will save lot of time. Regards, Lucas

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Switch on/off call recording
by Ashish gautam 22 May '13

22 May '13

Hi, I have setup a little call center application using mod_callcenter. Now what I want is to record all the calls based on the click of an ON?OFF button on the web app. Is there any variable or something like that which I can set like True or False to switch the call recording on or off? Also I want to know if the recording is possible per agent or not i.e. the call session to be recorded for a particular agent. Please throw some light on this. Thanks in advance. Regards -Ashish

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Notification of an event in the dialplan
by Christopher Aloi 21 May '13

21 May '13

I'm currently using FreeSwitch between two systems as a "rate limiter' with the limit application. I bring calls in, run through a hash to identify a loop - if my hash variable is met, I transfer the call to a new extension where I log the event and return a 503 to the source. If not I return a 302 to the source to send the call on its way. I'd like a way to identify these events to our NOC via email or a SNMP trap but I am not sure the best way to do it. The alert would fall into the 503 extension. What is the best way to accomplish this? I'd like to keep the call flow as simple as possible. Any ideas? Thanks - - Chris

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Separating Signaling and Media Among 2 FreeSWITCHs
by JP 21 May '13

21 May '13

Is there any way to separate the signaling and media for a call across 2 FreeSWITCH servers? Suppose that I would ideally like to transfer a call from one FS server to another, but the user agent doesn't support any kind of SIP methods to accomplish this (ex. REFER, INVITE with "Replaces" header, etc.). Is there a way to re-route the media to another FS server while keeping the signaling anchored on the original FS? Thanks JP

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