November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[freeswitch-users] configure one extencion in many sofphone and ringing in all
by Antony Aguirre Morales 13 Nov '14

13 Nov '14

Hi. I want to set one extention in 2 different softphone and I have been successful, the problem arises when I call the extention only sounds in one of 2 softphone configured. How could make sounds in the 2 softphone simultaneously. Thnks. -- Ing. Antony Aguirre Morales Ingeniero en Sistemas Computacionales. Celular. 8181729669

3 3

Music on hold sample rate mismatch
by Oleg Stolyar 12 Nov '14

12 Nov '14

This might be the easiest question ever asked on this list. I am playing wav files with sample rate 48000 as music on hold for a conference with sample rate 8000. I am getting this warning in the logs: switch_core_file.c:211 File /usr/local/freeswitch/sounds/music/8000/music.wav sample rate 48000 doesn't match requested rate 8000 I can ignore it, right? This will not create any actual problems? I don't think it will even increase the CPU consumption but wanted to confirm.

3 4

Configure an extencion in both devices and ringing in all.
by Antony Aguirre Morales 12 Nov '14

12 Nov '14

Hi, Thanks a lot by your answer. My question was related when i have an extensión and i configure the same extensión in two softphones. When this extensión is reciving a call, one extensión is ringing the second one don't. My question is, how can i configure for ringing both softphones and ringing in both devices? Regards.

1 0

Action in nested condition gets executed even when outer condition is false
by Frank Myhr 12 Nov '14

12 Nov '14

Hi, I have the following extension in a context that gets parsed several times during a call due to transfers and execute_extension. I'm finding that one of the inner actions (the correct one, based on time of day) gets executed even when the outer condition is false. Might not have noticed, except my original version of this extension actually played the greeting files, rather than just exporting the correct one in a variable. Callers were getting greeted multiple times, which was not the intention. For now I've sidestepped the audible problem by playing ${greeting} in another extension that does execute_extension on this one. But the greeting is still getting set multiple times per call. Would appreciate any clues as to why an inner action gets executed even though the outer condition is false. Thanks! Frank <extension name="set_greeting_tod"> <condition field="destination_number" expression="^set_greeting_tod$"> <condition minute-of-day="1-359" break="on-true"> <action application="export" data="greeting=ivr/ivr-hello.wav"/> </condition> <condition minute-of-day="360-690" break="on-true"> <action application="export" data="greeting=ivr/ivr-good_morning.wav"/> </condition> <condition minute-of-day="691-780" break="on-true"> <action application="export" data="greeting=ivr/ivr-hello.wav"/> </condition> <condition minute-of-day="781-1020" break="on-true"> <action application="export" data="greeting=ivr/ivr-good_afternoon.wav"/> </condition> <condition minute-of-day="1021-1380" break="on-true"> <action application="export" data="greeting=ivr/ivr-good_evening.wav"/> </condition> <condition minute-of-day="1381-1440"> <action application="export" data="greeting=ivr/ivr-hello.wav"/> </condition> </condition> </extension>

3 4

How to use mod_skypopen with the new Skype-4.3 without Pulseaudio server?
by Jacek 12 Nov '14

12 Nov '14

Hi I read a tutorial by Giovanni Maruzzelli. http://lists.freeswitch.org/pipermail/freeswitch-users/2014-August/107455.h… These are my notes: Theoretically, Skype 4.3.x is demanding Pulseaudio audio output. I write theoretically, because I noticed that Skype works without a problem thanks to the ALSA library itself Apulse. https://github.com/i-rinat/apulse In connection with the question whether it is possible with the help of this library to connect Skype with FreeSWITCH using eg mod_alsa or directly mod_skypopen? Skypopen once used the ALSA API (version 1.0.x, as I recall). Under ideal conditions, it would be best to libraries based on Apulse do fork - eg library FsPulse, that sound from Skype headed straight to the driver interface Skyopen, eg / dev / skypopen. Unfortunately, I am not a programmer, and I do not know how to change the ALSA library API to OSS compatible with FreeSWITCH. But if someone could create such a library to integrate Skype - FreeSWITCH, it for many years to solve the problem with Skype, and potential problems with possible errors Pulseaudio, Pulse which are patched often even after two years, such as the famous error: 866 http://lists.freedesktop.org/archives/pulseaudio-bugs/2010-October/004195.h… If the cooperation of Skype with FreeSWITCH to be stable and predictable, Pulseaudio in the middle of it is not in my opinion a good idea too. Also would be easier to control operation of Skype by FreeSWITCH, if the sound from Skype went directly to the server, without mediators, where the shape and function of FreeSWITCH developers have no effect. PS My natural language is Polish, sorry for bad English. Yours

1 0

no ring back tone
by kthofer 12 Nov '14

12 Nov '14

Hi there please be gentle with me but i can not find any help in the wiki to solve my problem this is a desperat try to get some answers to the following problem freeswitch is a configured as call back server FS originates the A-leg when answered the B-leg is originated. after A-leg has answered I like to play a short welcome message After the welcome mesage is played the international, original ringbacktone (telco carrier RBT) from the B-leg shall be passed through and the A-leg user should be able to hear it. This is important to hear some carrier specific announcements or coloured/personalized ring back. Right now we play our own ringbacktone even though the carrier provides us with a 183 sdp I did play with the usual Channel variables but as far as I understood the passing through of the ringbacktone is default behaviour. -- With best regards kT

4 5

FS rewrites RTP outgoing port. Client requires same port as incoming, no RX RTP
by Bill Ross 12 Nov '14

12 Nov '14

Folks; FS: 1.4.13 / OpenWrt disable-rtp-auto-adjust (default: false) is at default. FS is re-writing RTP to client port. Client requires same RX port as it is sending on. Can FS "somehow" be configured to use same port as incoming RTP? It is my understanding that rtp-auto-adjust only adjusts IP's if foreign and, ports if outside of rtp-start-port / rtp-end-port range Is this correct? Should I report a bug? Regards; Bill Ross

3 2

Skinny firmware version
by Nick Zaitsev 12 Nov '14

12 Nov '14

Good day to you. I try to set the Cisco 7985 to work with freeswitch. But i have a problem: 7985 requests the firmware version 2014-11-12 14:23:41.660763 [WARNING] skinny_server.c:1762 Device SEP0050600361BC:1 is requesting for firmware version, but none is set. 2014-11-12 14:23:41.660763 [DEBUG] skinny_server.c:1767 [SEP0050600361BC:1 @ 10.4.33.12:27811] Send Version with Version() 2014-11-12 14:23:41.660763 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:27811] Received HeadsetStatusMessage (type=2b,length=8). 2014-11-12 14:23:41.660763 [DEBUG] skinny_server.c:2159 [SEP0050600361BC:1 @ 10.4.33.12:27811] Update headset accessory status (OnHook) I've put into the config file. <param name="skinny-firmware-version" value="ATA030101SCCP04"/> <param name="skinny-soft-key-set-set" value="default"/> but it is still request for the firmware.(I supposed that there is not any difference,which firmware version to set).Moreover, there is "Configuring IP" string on the phone display,however the phone works and i can make calls. Please,give me an advice. Thank you for your time, Best regards. -- Nick Zaitsev ----------------------------------------------------------------------

1 0

Core dump in FreeSWITCH Version 1.2.24+git~20141016T210433Z~30a950b5a9~64bit
by Eduardo Alonso 12 Nov '14

12 Nov '14

Dear list: This is my first mail to the list. I hope that my data and email are well formed. We are using freeswitch as voicemail in a production environment and every week we obtain a core dump like the attached file. The error occurs in the CoreSession::recordFile function when the customer try to record a greeting message. As is normal it's not possible to attach the entire core file, as well I attached the backtrace following the debugging guide in the wiki: https://wiki.freeswitch.org/wiki/Debugging_Freeswitch. I hope that the attached file help you to find the error. If more information or testing is needed please let me know. I'm looking for a similar core in the list, but all errors related with this issue are different, from my point of view. Thank you in advance for your support. Cheers and best regards. -- *Eduardo Alonso Gil* VoIP Systems Engineer @ Quobis <http://www.quobis.com/> | e: eduardo.alonso(a)quobis.com | t: +34902999465

3 4

setting recording codec
by ik 11 Nov '14

11 Nov '14

Hello, I could not find any documentation on this subject. I require to have a recording of an a-law wav file (RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). Can I set a variable that tells the record command what type of codec to store the file with ? I know how to convert a normal wav recording into it using ffmpeg (for example): $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav But I wish to avoid it, if I can use the record app, or use the uuid_record API for it. Thanks, Ido

7 11