December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Console Dialing in Freeswitch
by Baskar 11 Dec '08

11 Dec '08

*Hi,* *It is possible to dial outbound through console dialing. Yes means me How?* *Another question whether there is any api command for console dialing.* * Thank you with regards, N.Baskar *

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conference module question - prompts
by jonathan augenstine 11 Dec '08

11 Dec '08

I am trying to modify the behavior of the playing of prompts when someone enters the conference. When I enable the conf-welcome prompt and a new participant enters the conference, the prompt is played to the conference and everyone hears the welcome. Is there a any way to modify the configuration so that the welcome prompt is only heard by the participant entering the conference? Thank you. Jonathan jaugenstine(a)gmail.com

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Audio routing problem between FS and Voxeo
by mszlazak＠aol.com 10 Dec '08

10 Dec '08

I’m trying to route calls from X-lite <--> FS (Nov. 6 2008 svn) <--> Voceo (Prophecy) to use Voxeo’s ASR instead of FS’s built in PocketSphinx/ASR. All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi router that connects a laptop to my desktop in case that matters but I’m not using the laptop for any of this. I’ve set up an extension to bridge calls to Voxeo. Here is the entry in file conf\dialplan\default.xml: <extension name="Doctors Office"> <condition field="destination_number" expression="^2007$"> <action application="info"/> <action application="bridge" data="sofia/internal/sip:Doctorsoffice@127.0.0.1:5068"/> </condition> </extension> I hear one ring then a hang up. No errors in the FS console and Voxeo’s logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeo’s Prophecy ASR called “Doctorsoffice.” Any ideas? Thanks. Mark.

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Hardware requests.
by Brian West 10 Dec '08

10 Dec '08

Does anyone have a Polycom 320 they would like to donate to the project? I need one to trouble shoot a problem. Please contact me off list. Thanks, Brian West

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Audio routing problem between FS and Voxeo
by mszlazak＠aol.com 10 Dec '08

10 Dec '08

Brian, I'm still having the same audio problem when bridging/routing to Voxeo using the latest snapshot. Help! Also, it doesn't look like mod_spidermonkey_teletone.dll was put into this latest snapshot. Has it been changed since November? I could use that older version of this dll. Listen to 350+ music, sports, & news radio stations – including songs for the holidays – FREE while you browse. Start Listening Now!

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how to force a MINIMUM call duration
by Frank ＠ Impact 10 Dec '08

10 Dec '08

How can FS force a Minimum call duration for a FS caller (someone calling out of FS)? We have a carrier that penalizes us with a surcharge for short duration calls (sound familiar?). So when a FS caller (not a call center or predictive dialer) calls a cell phone and gets a ring tone or calls an answering machine, the FS caller hangs up because they do not want to leave a message. But they do this in less then a few seconds after the call is answered. This becomes a short duration call and bang the surcharge applies. It is actually cheaper to pay for a longer call time (6 seconds in this case) and avoid the short duration surcharge. But the FS caller does not know this. So, how can FS hold the connection to the called party open for at least the minimum amount of time I need to avoid the short call charge. even though my FS caller has already hung up the phone on his end? I would like to do this in the xml dialplan if possible. Thanks -Frank

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incoming call routing <domain>
by ccav 10 Dec '08

10 Dec '08

Cable modem <----> nat router <----> fs fs is set as DMZ on nat router so all packets get there. My ipv4 address is 192.168.0.x The nat router holds the public IP. Public IP is a registered domain sparkz.tv so addressable from the internet cloud. Since fs is DMZ, all requests for sparkz.tv or sip.sparkz.tv are resolved and so IP routing is good. So I'm trying to get external sip/soft phones registered and routed properly. The domain/server set in the phone client is sip.sparkz.tv:5080, since the wiki says they need to be set that way??? I have created a conf/directory/sip.sparkz.tv.xml and a conf/directory/sip.sparkz.tv where I have users registration info. conf/directory/sip.sparkz.tv.xml was copied from default.xml and has: </params> I have modified conf/sip_profiles/external.xml and added an <alias name="sip.sparkz.tv"/> External phones are registering and are visible under sofia status profiles external and sip.sparkz.tv Calls outbound from the phones are being routed properly. Calls inbound to their DID's are not. Calls to softphones on the local private net 192.168.0.x register and route properly. vars.xml sets domain to ip_v4... the default.xml dialplan seems to iif the profile to either nat or default.. so I end up with the call going to DID(a)192.168.0.x rather than the registered interface... I'm routing the calls in the dialplan to bridge to user/$1@$${domain} but $${domain} is set to ip_v4 so it's wrong... Any clues what I need to do next to get them routing properly? I want to be able to support multiple domains. how do I do this correctly? -- View this message in context: http://www.nabble.com/incoming-call-routing-%3Cdomain%3E-tp20928933p2092893… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Jitter + Packet Loss
by Jonathan Palley 10 Dec '08

10 Dec '08

I'm curious to start a discussion on being able to query a channel and get statistics on the incoming jitter and packet loss (calculated from the RTP, not RTCP). Is this on the roadmap? Is it hard to do? Would be very useful for us indeed! Thanks - JP

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Question Regarding ASR/TTS and CMU OSS Projects
by Andrew Gilbert 09 Dec '08

09 Dec '08

Curious if anyone has practical real world input on training CMU based ASR engines (Sphinx, PocketSphinx) and / or creating and tuning voices for the TTS related components. Just trying to understand how hard it is, what the realistic gap is to use these tools in real world applications.

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Sending SIP calls via outbound proxy
by Erick Johnson 09 Dec '08

09 Dec '08

Hi There, I'm trying to get freeswitch to originate all SIP calls through an outbound proxy. When I use the originate API command to create a call to a telephone number I see the SIP packets getting to my proxy just fine. However if I originate a call to a SIP address then proxy server is bypassed, instead FS is directly messaging the addressee. Here is the command that I'm trying to use that behaves unexpectedly: originate sofia/gateway/proxy/alice(a)bar.com &echo() However this command produces the results I'm expecting: originate sofia/gateway/proxy/15551234 &echo() Here is the result of my sofia status: freeswitch> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia@X.X.X.X:5070 RUNNING (0) proxy gateway sip:ejjohnson_ippx@ejjohnson.org NOREG ================================================================================================= 1 profile 0 aliases I have also tried setting the sip_invite_domain channel var through the {} Could you let me know what I'm doing wrong? Much appreciated, Erick J

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