- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_openmrcp
by David Knell 16 Nov '08

16 Nov '08

Hi - Is anyone out there using mod_openmrcp? Be interested in finding someone further down the learning curve than I am at the moment.. Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

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How many calls can FreeSwitch bridge simultaeously
by George Howitt 16 Nov '08

16 Nov '08

My FreeSwitch app needs to bridge a lot of calls, and process all the media. I'm using PCMU, and every inbound call gets bridged to another outbound call. There shouldn't be any transcoding. How many concurrent calls can FreeSwitch support under this scenario? Can someone point me to any performance measurements for FreeSwitch? Thanks.

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Re: [Freeswitch-users] E-mail as user id and bridging
by Jan Kubr 16 Nov '08

16 Nov '08

> You can't have two @'s in a uri. You do know its already domain > based.. you have domains that have users inside them? so you can dial user/brian(a)domain.com > already without any extra thought? You can have multiple domains in > the directory... with users in each domain. These domains don't point to the Freeswitch box though. They are complete random from people from the Internet. So the e-mail might be james(a)yahoo.com while obviously yahoo.com doesn't point to my Freeswitch box. <domain name="yahoo.com"> <user id="jan.kubr"> won't work then because the user can't log in. Basically I want the user to use his e-mail to login with her SIP phone. This works fine when the user id is james(a)yahoo.com (in a domain that is the IP of the FS box), but then it is not possible to bridge a call that user. Jan

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SIP multi domain and FS as SIP transparent B2BUA
by Iñaki Baz Castillo 16 Nov '08

16 Nov '08

Hi, I've a SIP multi domain environment based on OpenSer, so subscribers use OpenSer as registrar and proxy. I want to use FreeSwitch (it's my first time with it) as an application server / B2BUA, but I need it to work in SIP multi-domain way, this is, if for example I use the voicemail server of FS, there will be voicemails for users called: - alice@domain1 - alice@domain2 - bob@domain1 ... This is unfeasible with Asterisk that just handles the username part of the Request URI and From/To headers. How possible is it with FS? is it designed to handle multidomain in SIP? I have other questions more related to pure SIP protocol and FS as transparent B2BUA: alice ---(leg_a)--- FS (B2BUA) ---(leg_b)--- bob - If alice adds a custom header ("Subject: hello!"), will this header appear in leg_b to bob? - If alice sends a custom in-dialog request (INFO request with no standar Content-Body), will FS forward it to bob by just changing the needed data (From_tag, To_tag, Call-ID, Via, CSeq...)? - When alice sends the initial INVITE to bob (through FS), if bob replies a "strange" response code ("497 Strange response"), will FS respect it and forward upstream to alice? - Is it possible FS to act as a SessionTimer endpoint with alice and bob after the call (both legs) has been established? I don't want FS to handle media (so I'll use "media_bypass" mode) but I need FS controling the call status by sending in-dialog requests, hopefully using SIP Session Timers (RFC 4028). Thanks a lot for all your replies, I hope to enjoy FS :) -- Iñaki Baz Castillo

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Problem with multiple meta-app in the latest build
by Doug Blacksone 16 Nov '08

16 Nov '08

Hi, In my dialplan, I have fresswitch so that user can hit *1 to start recording and *2 to end recording, as follows: <action application="bind_meta_app" data="1 ab s execute_extension::record XML features"/> <action application="bind_meta_app" data="2 ab s execute_extension::stoprecord XML features"/> <action application="bind_meta_app" data="3 ab s execute_extension::att_xfer XML features"/> With the latest build, freeswitch starts to give warning message when *2 is pressed after *1. 2008-11-16 22:23:42 [DEBUG] switch_ivr_play_say.c:456 switch_ivr_record_file() Raw Codec Activated 2008-11-16 22:23:52 [DEBUG] switch_rtp.c:1618 switch_rtp_dequeue_dtmf() RTP RECV DTMF *:2000 2008-11-16 22:23:53 [DEBUG] switch_rtp.c:1618 switch_rtp_dequeue_dtmf() RTP RECV DTMF 2:2000 2008-11-16 22:23:53 [DEBUG] switch_ivr_async.c:1477 meta_on_dtmf() sofia/internal/1000(a)192.168.1.101 Processing meta digit '2' [execute_extension::stoprecord XML features] 2008-11-16 22:23:53 [WARNING] switch_ivr_async.c:2035 switch_ivr_broadcast() Channel [sofia/internal/1000@192.168.1.101][execute_extension::stoprecord XML features] already broadcasting...broadcast aborted This *1 and then *2 mechanism has been working before, until I upgraded to the latest build. Does anyone know what may be causing the problem and how I can fix it? Thanks, Doug

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Originate & dialplan entry.
by Johny Kadarisman 16 Nov '08

16 Nov '08

Is there an options to originate call using dialing pattern in the dialplan, instead of putting the complete gateway url syntax. For example, I setup a dialplan ^9(1]d{10}) pattern that will dial out using certain gateway profile. Could I do something like, originate 918001231234@default 5000.

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Hunt-* variables and Meta-app
by Juan Backson 16 Nov '08

16 Nov '08

Hi, I am using Meta-app when bridging too calls using "ab" as the allowed-executor. When the a-leg presses *1, the Hunt-Username refers to a-channel's username and when the b-leg presses *1, the Hunt-Username also refers to a-channels's username. Is this normal? I am simply looking for ways that I can capture the actual user of either a- or b- leg that is pressing the *1 key. Is it possible? Thanks in advance for any help. JB

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ACD Queues
by Ken Treis 16 Nov '08

16 Nov '08

Hello all, I managed to set up and play with FreeSWITCH a little (thanks to some great help from Brian via IRC a couple of months ago, thanks Brian!) but I am having trouble figuring out if FreeSWITCH is able to support automatic call distribution. I'm new to FreeSwitch but fairly familiar with Asterisk, where mod_queue does everything I need -- except work properly on EC2. Is there any similar solution for FreeSWITCH? I have looked through the documentation on FIFOs but for our application I need the caller the software to dial agents when a call comes in, rather than having the agents wait on the line until somebody arrives. I also need the caller to be able to press a key to leave a message instead of continuing to wait in queue. If mod_fifo can do both of those things, then I'd be set. Any pointers would be greatly appreciated. -- Ken Treis Miriam Technologies, Inc.

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Question on CDR CSV
by Shelby Ramsey 15 Nov '08

15 Nov '08

Hello, Looking at the cdr-csv files ... it would appear that the call gets logged to 2 files: -- an accountcode.csv file -- and the master.csv file Is this correct? Just want to make sure I don't miss or duplicate calls in the CDR records ... Thanks! SR

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Compile error and possible solution
by Gonzalo Servat 15 Nov '08

15 Nov '08

Hi, I just tried doing a make current (I was on 10231 and updated to 10416) and mod_python failed to compile. I can't remember exactly what the error msg was as it's beyond what my terminal's scroll buffer is configured to store but it was something along the lines of "cc1 will treat all warnings as errors" and then an error msg similar to: switch_core.h:1686:function declaration isn't a prototype This happened while building mod_python. I think it was mod_python.c which includes "switch.h", and in switch.h line 103 it includes switch_core.h, and in switch_core.h line 1686 I see this line: SWITCH_DECLARE(void) switch_cond_next(); Apparently if switch_cond_next() takes no parameters, 'void' goes in there. After some googling, in C/C++ if you put nothing in the parentheses it means it takes an unlimited number of arguments while void means none. Why would it complain if that's the case? I had a look at the switch_code_next() function definition in switch_time.c() and it looks like its not supposed to take any arguments so I put 'void' inside the parentheses in line 1686 of switch_core.h and it builds fine now. Is this the correct fix? - Gonzalo (znoG #freeswitch)

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