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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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Proto specific hangup cause issue
by Apostolos Pantsiopoulos 09 Dec '08

09 Dec '08

The proto_specific_hangup_cause is missing on one of the two call legs. When the caller hangs up it is missing from the a-leg CDR. When the callee hangs up it is missing from the b-leg CDR. Is this nornal? And another question : what piece of info could inform me about who hanged up? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Catching hangups
by Joe Bain 09 Dec '08

09 Dec '08

Hi, I'm writing an IVR in Lua and am having problems dealing with hangups cleanly. Very often session:ready() reports true long after I have hung up and the hangup hook function I have set doesn't get called either. It seems to report that the session is active indefinitely in some cases where a loop keeps trying to get some dtmf key presses. Is there any trick to using session:ready() or the hangup hook that I might have missed? On a slightly related point I can't seem to access the session properties, e.g. session.caller_id_num has a value of nil. Any thoughts here? Thanks in advance, Joe Bain

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root privs for mod_fax
by Gabriel Kuri 09 Dec '08

09 Dec '08

I've been experimenting with mod_fax and discovered it doesn't appear to receive faxes unless freeswitch is running as root? it fails trying to open the tiff file for writing (see the logs below). I'm using the dialplan as prescribed in the wiki without any changes and the user the freeswitch process is running under has privs to write to /tmp, but it still fails to receive faxes. I haven't tried sending any faxes yet. I'm running r10609. any ideas? 2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Get document at 14400bps, modem 7 2008-12-09 00:29:41 [WARNING] mod_fax.c:133 spanfax_log_message() WARNING T.30 Cannot open target TIFF file 'rxfax.tiff' 2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Changing from state 17 to 3 2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Tx: DCN with final frame tag 2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Tx: ff 13 fa 2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 HDLC carrier down in state 3 2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D _TX 2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX Set rx type 0 2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX Set tx type 4 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Disconnecting 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX Set rx type 0 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX Set tx type 1 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Changing from state 3 to 2 2008-12-09 00:29:44 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Send complete in phase T30_PHASE_E, state 2 2008-12-09 00:29:44 [DEBUG] mod_fax.c:163 phase_e_handler() =============================================================== =============== 2008-12-09 00:29:44 [DEBUG] mod_fax.c:176 phase_e_handler() Fax processing not successful - result (41) TIFF/F file cannot be opened. Gabe

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Two major flaws: Could they be fixed?
by Dennis 09 Dec '08

09 Dec '08

hi, we are fighting with two flaws in fs and would be happy, if they could be fixed. we are using socket outbound. 1.) hangup a call in ringing state: this worked in one of the last fs versions, but suddenly does not work anymore. let's say, we have an inbound call and do 3 originates to different targets. all 3 targets are in ringing state. the target, which answers first, will be bridged with the inbound call, the other two (still ringing) targets should be hung up. we do not want fs to hang up the other two originates automatically. we want to hang up the other two originates by sending the hangups. we set "hangup_after_bridge=false" and "park_after_bridge=true". we do sendmsg hangup uuid. but the originates are first hung up, when they are answered. when they are in ringing state, the hangup will do nothing (anymore). as i said, it worked before, so i assume, that something has changed in the latest trunks. 2.) sendmsg uuid *whatever* can cause to excute the command on the wrong uuid: let's say, we have an inbound call and an outbound call - at least we thing we have it ;-). now we do for example sendmsg outbound_uuid hangup to hangup the outbound call. but, if the uuid of the outbound call does not exist (because there was a problem or something), the inbound will be hung up instead. the same happens with all sent messages to an uuid, which does not exist. if we want to do a playback for the same outbound, the inbound will hear it, if the outbound_uuid does not exist. perhaps this is a feature, but i think that it would be nicer and more reliable, if the sendmsg is only executed on the given uuid. if the given uuid does not exist, nothing should happen or even nicer, an event with an error should be sent to the socket. thanks dennis

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