October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Phrases and speech
by Noah Silverman 16 Oct '08

16 Oct '08

Hi, I have a question about setting up speech in FS... I'm far away from the default config, so I just want to add the MINIMUM amount necessary to my config to enable speech. My goal is to have some basic functions like "You have 3 dollars and 27 cents left". (Do I need to build the cepstral code and modules?) Can anyone help me with a simple, clean few lines of XML in the config to enable this? Thanks, -N

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Wrong port on response
by David Aldworth 15 Oct '08

15 Oct '08

Hello - Got an issue with Freeswitch not responding on the port that the initial request was made on. I'm not beyond believing that it is a NAT or router issue except that I can register a Cisco phone from another location or a softphone from the same location without any problem. This Aastra just won't work for some reason. We have connectile-dysfuntion turned on. Otherwise we are using the default profile settings. Auth is on (as you can see from the below). Basically, the Reg request comes from port 41450, but freeswitch responds on port 5060. Again, other UA's work fine, just one Cisco and one Aastra from this site do not. Meanwhile a soft phone from this site, and the same model cisco from another site do not work. SIP dump and external profile are below. Thank you for any help. David U 2008/09/19 16:51:45.145303 63.211.239.34:41450 -> 70.42.223.23:5060 REGISTER sip:atl.teliax.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab. Max-Forwards: 70. Content-Length: 0. To: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>. From: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6. Call-ID: ecae0e043ee65c5c5340749a9d28235f(a)192.168.1.192. CSeq: 288881481 REGISTER. Contact: Aastra Test <sip:pleasehelp(aastra)@192.168.1.192:5060;transport=udp >;expires=300. Allow-Events: talk,hold,conference. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO. Expires: 300. User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ v3.2.8.45. . U 2008/09/19 16:51:45.145493 70.42.223.23:5060 -> 63.211.239.34:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab;received=63.211.239.34. From: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6. To: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>;tag=Sarvm1DjmU3Zg. Call-ID: ecae0e043ee65c5c5340749a9d28235f(a)192.168.1.192. CSeq: 288881481 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="atl.teliax.net", nonce="d3538e86-9d86- dd11-82bf-001143e64915", algorithm=MD5, qop="auth". Content-Length: 0.  <profile name="external">  <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile>

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Managing a conference in JS
by Birgit Arkesteijn 15 Oct '08

15 Oct '08

Hi all, I'm trying to manage a conference call in js. I found an example on the FS wiki: http://wiki.freeswitch.org/wiki/Examples_confcall_js I know that using session.execute("conference", route + "@default"); I can drop a existing (originated) call into a conference. Reading http://wiki.freeswitch.org/wiki/Mod_conference, I see that it's possible to create a conference and controls things from within the conference. How could I do that in JS? In other words: Is it possible to do in one JS script: - create an empty conference - dial someone from that conference (and drop them in) - dial another person from that same conference (and drop them in) - dial a third person ... - dial a Nth person ... - wait until all (or all but one) hang up and drop the conference Cheers, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Help!!--Want to setup Calls per Second
by Krunal Patel 15 Oct '08

15 Oct '08

Hi I am new to Freeswitch. I would like to set CPS dynamically. Would you please tell me how to control as well as setup CPS (Calls Per Second) dynamically? Thanking you in Advance -- Krunal Patel

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mod_conference Name announcement or INTROUSER
by Sheeju Alex 15 Oct '08

15 Oct '08

Hello All, I am working on the process to setup announce user name who join/leave the conference. For this first it was required to setup a process to record the user name while joining the conference, this patch attached with this mail is programmed for this but I have some issues with file format... Conference Syntax: confname+flags{introuser}+1234 Freeswitch log: 2008-10-15 07:29:25 [NOTICE] mod_conference.c:4537 conference_function() Conference member introuser flag set. 2008-10-15 07:29:25 [NOTICE] mod_conference.c:4548 conference_function() Conference 77777 INTROUSER set. 2008-10-15 07:29:25 [NOTICE] mod_conference.c:4550 conference_function() File Path is /tmp/conference_member_77777.wav 2008-10-15 07:29:31 [NOTICE] mod_conference.c:464 create_file() File Path as in create_file is /tmp/conference_member_77777.wav 2008-10-15 07:29:31 [ERR] mod_sndfile.c:144 sndfile_file_open() Error : file format is invalid (0x00010000). 2008-10-15 07:29:31 [NOTICE] switch_ivr_play_say.c:403 switch_ivr_record_file() Hangup sofia/internal/1000@192.168.0.3:5061 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-10-15 07:29:31 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File [/tmp/conference_member_77777.wav] [System error : No such file or directory.] 2008-10-15 07:29:32 [NOTICE] switch_core_session.c:833 switch_core_session_thread() Session 1 (sofia/internal/1000@192.168.0.3:5061) Ended 2008-10-15 07:29:32 [NOTICE] switch_core_session.c:835 switch_core_session_thread() Close Channel sofia/internal/1000@192.168.0.3:5061 [CS_HANGUP] I don't understand why .wav is not valid format, Someone please review this and suggest me the what exactly can fix this issue. Will look forward for reply! Thanks, Sheeju

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mod_conference: Any way to pass conference PIN in the URI?
by Joe Attardi 14 Oct '08

14 Oct '08

Hi all, Part of an app I'm working on has a web UI for managing conferences on a FreeSWITCH instance. One of the goals in this effort is to allow a conference host to invite participants. We had initially used the 'api conference dial/bgdial' commands, but unfortunately we ran into some problems (we needed to provide credentials to our SIP proxy, among other things). So what we're doing now is inviting the person with a REFER. The conference owner enters a phone extension/SIP URI and the remote party's phone rings. When the remote party answers the call, it then REFERs their phone to call the FreeSWITCH conference. If the conference requires a PIN, then, obviously they get prompted for a PIN. Here's the million dollar question: Is there any way to pass along the conference PIN as a URI parameter or something like that? Any help would be greatly appreciated. Thanks in advance, -- Joe

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mod_conference nitpick
by Kristian Kielhofner 14 Oct '08

14 Oct '08

Hello everyone, When you enter a conference without any other users, you hear the "you are the only person in this conference" recording with MOH in the background. Is there a way to make the MOH play after the other recording finishes? I know it's a nitpick but some people seem to be violently opposed to the way it happens now... Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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Re: [Freeswitch-users] Originate command
by Jon Bruel 14 Oct '08

14 Oct '08

Brian, I hope this is what you expect: Testing on CLI, this one works: Originate sofia/gateway/100004444500/201,sofia/gateway/100004444500/205 206 XML internalpreparation And this one does not: Originate user/100008888201@10.3.1.24,user/100008888205@10.3.1.24 100008888206 XML internalpreparation Or rather, it gives the same result as this: Originate user/100008888201@10.3.1.24,user/100008888201@10.3.1.24 100008888206 XML internalpreparation (Last user made identical to first user.) I should remark that the gateway above loops back to the dialplan with the context internalpreparation. /Jon

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Distributed voicemail storage
by Kristian Kielhofner 14 Oct '08

14 Oct '08

Hello everyone, I'm looking to build a distributed voicemail platform with FreeSWITCH. What is the current recommended way to store messages, recordings, etc in a distributed manner? Would a simple NFS share work? Not that I would ever use it, but just for example... Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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Strange sound hear between originate_retries
by Adnan Barakat 14 Oct '08

14 Oct '08

Hi all, Got a strange problem when using originate_retries, if the destination is busy/rejects the call there are 2 low pitched beeps heard, then FS will try dialling again and the normal ring-tone is heard. (these 2 beeps are heard between very retry) I have removed the sounds folder in case it was something in there, but it wasn't. I'm using r9959 - it also used to happen if the destination didn't answer, but since the update it doesn't seem to do it any more. Here is the dialplan I'm using: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="DialPlan"> <context name="public"> <condition field="destination_number" expression="^(.+)$"> <action application="set" data="server_ip=$${local_ip_v4}"/> <action application="set" data="ignore_early_media=true"/> <action application="answer" /> <action application="set" data="progress_timeout=20"/>  <action application="set" data="originate_retries=3"/> <action application="set" data="originate_retry_sleep_ms=2000"/> <action application="set" data="ringback=${uk-ring}" /> <action application="set" data="ring_ready=true" /> <action application="bridge" data="sofia/internal/123456789@domain.com,sofia/internal/987654321@domain.com" /> </condition> </context> </section> </document> Any advice or feedback would be appreciated. Adnan

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