- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_voicemail prompt nitpick from a user - prompt order is flipped around
by Nathan Neulinger 09 Oct '14

09 Oct '14

Got a report from a user that is more 'support' oriented related to recording voicemail greetings. He noticed that the prompt ordering is flipped in the config menu - everything else in voicemail is: to do A press X to do B press Y to do C press Z "to record a greeting press 1" but as soon as you record a greeting, they flip around to: press X to do A "press 1 to listen to recording" "press 2 to save recording" His comment was that if you are paying attention to the "result -> action", you can get confused and wind up trying to press 3 to save your greeting instead of pressing 2. It's this in the vm lang macros: <macro name="voicemail_record_file_check"> <input pattern="^([0-9#*]):([0-9#*]):([0-9#*])$"> <match> <action function="play-file" data="voicemail/vm-press.wav"/> <action function="say" data="$1" method="pronounced" type="name_spelled"/> <action function="play-file" data="voicemail/vm-listen_to_recording.wav"/> <action function="play-file" data="voicemail/vm-press.wav"/> <action function="say" data="$2" method="pronounced" type="name_spelled"/> <action function="play-file" data="voicemail/vm-save_recording.wav"/> <action function="play-file" data="voicemail/vm-press.wav"/> <action function="say" data="$3" method="pronounced" type="name_spelled"/> <action function="play-file" data="voicemail/vm-rerecord.wav"/> </match> </input> </macro> I can certainly fix this locally, but would there be any objection to a commit to flip these around so the prompts are consistently ordered throughout mod_voicemail by default? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Using fs_encode on G722 audio produces half speed wav file
by Michael Collins 09 Oct '14

09 Oct '14

For those of you familiar with the fs_encode tool I have a question - I can run it on a G722 raw file (such as is produced when using tshark to extract the RTP stream) and it produces a perfect wav file at exactly half speed. I'm using basic syntax: #>fs_encode raw_audio.G722 output.wav Looking at fs_encode's options I tried using -r to specify the sampling rate, but that seems to apply to the input on the codec and anything other than 8000 produces an error. Has anyone used fs_encode to decode G722 (or any other exotic codec) with success? I appreciate your input. -MC P.S. - doing #>sox slow_audio.wav normal_audio.wav speed 2.0 will "fix" the file, so I'm not totally dead in the water. I was just curious if I was possibly not using fs_encode properly. :)

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Re: [Freeswitch-users] fs_path with a 'transport' parameter
by Dan Christian Bogos 09 Oct '14

09 Oct '14

Yuriy, In my case I use t_relay() to send the request out. Maybe make sure you are not sending in custom way? If that does not work for you, I suggest continuing the thread over OpenSIPS mailing list since there might be something in your config which is preventing t_relay to work properly. DanB On 09.10.2014 15:43, freeswitch-users-request(a)lists.freeswitch.org wrote: > Thanks for instruction, DanB, really very appreciate! > but... unfrtunately in my case opensips just added transport=tls ro RURI but stil send to endpoint via UDP.

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Can not see video with low internet connection (audio is ok)
by To Hai Son 09 Oct '14

09 Oct '14

Hi, We are now using Freeswitch for VOIP part of our mobile social network application for Japanese market. We have a problem with Freeswitch as following: Video calls are performed perfectly with high speed internet connection, but if internet connection speed is quite low, we can not see the video, just can hear voice only. We have make an emulation call from PC with low speed of internet connection, the same problem occurred. We guess that there is a parameter in FreeSwitch to limit the internet connection, and server does not allow video stream if internet connection speed is lower than this parameter. We are newbie to Freeswitch, we do not know much about it, so please help us. Any advice will be appreciated. Thanks and Regards, SonTH

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Problem with Sound in Soft Phone (FSClient)
by Manish Talwar 09 Oct '14

09 Oct '14

Hi, We have successfully integrated FreeSwitch with our current web based IVR application by "httapi", Its working fine as per our requirement. We have setup FreeSwitch on our "linux EC2 AWS Server", and its successfully handshaking our IVR application with FreeSwitch. We have registered default "1001" user on FSClient (1.2.3.5) soft phone by SIP Id and we can call 5000 for running "IVR_DEMO" and also able to call "httapi" methods by SIP Id. But we are stuck on one issue i.e. When we call by soft phone, its successfully connected a call and running all commands on FreeSwitch console but there is no Voice coming on soft phone. All other soft phone functionality are working fine except sound problem in soft phone. ? We have got a server Soundcard details by running "aplay -L" command on server, but still we are unable to get any sound from FreeSwitch by soft phones. For reference, Please find the log details as an attachment. Please suggest us if we need to do some configuration setting in Conf file related to Sound issue. Regards, Manish Talwar

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Re: [Freeswitch-users] fs_path with a 'transport' parameter
by Dan Christian Bogos 09 Oct '14

09 Oct '14

Yuriy, You should not need to select the socket from where you send out your request in OpenSIPS. Just do something like this: * Remove previous transport parameter from ruri (for the case it was added by fs_path). Have the feeling it will work also without removing since opensips will overwrite. * Add a new transport parameter with what you got in your private header as transport in ruri params. """if is_present_hf("X-EpTransport") $ru = $ru + ";transport=" + $hdr(X-EpTransport);""" * Remove the private header. * Relay the request as you did previously, things should automatically work out for you due to transport param. Cheers, DanB On 09.10.2014 02:13, freeswitch-users-request(a)lists.freeswitch.org wrote: > Thanks for the good advice, DanB! > I thought about this to be honest. > Yes, FS can add header with X-EndpointTransport: tls and opensisp can detect this. > But... I am still not sure how let opensips know that it should use tls instead of udp for invite to endpoint. > I tried to use > force_send_socket(tls:my_IP:5061); > But looks like I can't convert transport by this. > > Will try to send by > t_relay("tls:1.2.3.4:6677"); > > Probably it is the question more for opensips list in this case but It is related I hope. > > Will be grateful for advice > Thanks

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Re: [Freeswitch-users] fs_path with a 'transport' parameter
by Dan Christian Bogos 08 Oct '14

08 Oct '14

Hi Yuriy, We have solved this issue by adding a custom header with the transport OpenSIPS need to use when sending the call out (extract that info in OpenSIPS, eg: X-EndpointTransport: udp). A more elegant solution would be custom ruri params being sent out by FreeSWITCH, however not sure how doable that would be technically speaking since the transport is automatically written out from the one you set in fs_path. Hope this helps, DanB On 07.10.2014 21:06, freeswitch-users-request(a)lists.freeswitch.org wrote: > I can add transport parameter I need on FS side, but how opensips will > >know that it should use a different transport for INVITE to an endpoint?

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ClueCon Weekly! Today (Wed Oct 8) 1PM Eastern
by Ken Rice 08 Oct '14

08 Oct '14

Hi there! Today Eric Tamme, Protocols Engineer at OnSIP will be joining us today to talk about OnSIPs call architecture and how and why it is different then most. Join us at 1PM Eastern (10 AM Pacific) on https://conference.freeswitch.org/ And don¹t forget tomorrow 2PM Central the Weekly Bug Hunt! For more conference access methods such PSTN please visit https://confluence.freeswitch.org/display/FREESWITCH/ClueCon+Weekly+Conferen ce+call Hear you then!! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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PostgreSQL Database
by Joel White 08 Oct '14

08 Oct '14

I am noticing an issue when I moved sip_profile from internal sqlite db to postgreSQL several tables from the channels table are missing initial_cid_name initial_cid_num initial_ip_addr etc... How do I fix this issue short of manually adding the columns myself?

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how to add to sip To/Contact/Invite uri on outbound call?
by Nathan Neulinger 08 Oct '14

08 Oct '14

Is there any way to add a parameter to the outbound uri via dialplan variables? In particular, I'm wanting to investigate setting a 'tgrp' and 'trunk-context' value on some outbound calls. http://www.markholloway.com/blog/?p=352 -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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