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Re: [Freeswitch-users] List URI BLF Config Unique to Broadsoft or Can be Done in Freeswitch?
by Lyle Pratt 22 Dec '15

22 Dec '15

I would be interested in seeing the traces as well and we could possibly help implement this functionality if we could make progress with the traces. Regards, Lyle Pratt

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DTMF Volume too low
by Алексей Морозов 22 Dec '15

22 Dec '15

Hello. We have encountered a problem. We use bind_digit_action application to activate a number of features. To still have ability to pass DTMF signals between A_leg and B_leg, we use send_dtmf application. This is a part of our dialplan. <extension name="Bind_DTMF_action"> <condition field="destination_number" expression="^DTMF_CALL_CONTROL$"> <action application="bind_digit_action" data="dtmf_call_control,*1,exec:execute_extension,attented_xfer XML ${context},both,self"/> <action application="bind_digit_action" data="dtmf_call_control,~^\d,exec:execute_extension,send_dgts XML ${context},both,peer"/> <action application="digit_action_set_realm" data="dtmf_call_control"/> </condition> </extension> <extension name="Send_DTMF"> <condition field="destination_number" expression="^send_dgts$"> <action application="send_dtmf" data="${last_matching_digits}"/> </condition> </extension> All work well but one snag - our customers from time to time complain inability to interact whith IVR. During debugging we have discovered that Freeswitch generates DTMF signals with volume = 13. The problem is that a lot of popular PBX and call managers (for example Cisco UCM) do not recognize DTMF with such low volume. Is it possible to set this value in freeswitch as we need or this parameter is not configurable? -- WBR Alex M.

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List URI BLF Config Unique to Broadsoft or Can be Done in Freeswitch?
by Colton Conor 22 Dec '15

22 Dec '15

I asked this question on the VoiceOPs lists, but have not received a response. Considering jumping from Broadsoft to a FreeSwitch based platform. Wanted to know the following: Is the way Broadsoft implements Busy Lamp Field (BLF) with a list URI unique only to Broadsoft? I believe I am talking about Enhance BLF. In the configuration of BLF on a Polycom phone using Broadsoft there is a Broadsoft List URI field. The same is true with Yealink, Gigaset, Cisco, and other phones. Here is a configure guide for a Polycom 4.3.2 Busy Lamp Field Configuration http://community.polycom.com/polycom/attachments/polycom/VoIP/8977/1/Partne… Here is the config guide for a Yealink with screenshots of what the Broadsoft screens look like. Basically from the Broadsoft interface you can decide which users to monitor, and it it will instantly and dynamically update the List URI with the monitored users. Not need to change anything in the phone configure file (assuming you aready have the list uri in there) http://www.yealink.com/Upload/UCOne/features/Quick_Setup_BLF_List_on_Yealin… On other platforms, freeswitch for example, enhanced BLF is not configured with a list. Instead, each line you want to monitor, needs to be added in the config file. The phone then has to be rebooted, and the new config file has to be loaded. Here is what the freeswitch wiki shows: https://wiki.freeswitch.org/wiki/Polycom_Presence_Setup *Enhanced BLF* provides ringing indication and directed call pick-up capability. Once a productivity license has been installed, the BLF configuration can be added to the phone1.cfg(resp. in a phone-specific settings file like "${MAC}-settings.cfg). The following should be configured at the phone's file (e.g. phone1.cfg or *MAC-address*-settings.cfg): <attendant attendant.reg="1" attendant.ringType="1" attendant.behaviors.display.spontaneousCallAppearances.normal="1" attendant.behaviors.display.spontaneousCallAppearances.automata="1" attendant.behaviors.display.remoteCallerID.normal="1" attendant.behaviors.display.remoteCallerID.automata="1" attendant.resourceList.1.type="automata" and for each user you want to watch: attendant.resourceList.1.address="8000" attendant.resourceList.1.label="FirstUserName" attendant.resourceList.2.address="8002" attendant.resourceList.2.label="SecondUserName" /> And finally, the following line should be added into local-sip.cfg: <call call.directedCallPickupString="**" /> On a FreeSWITCH installation with a default-dialplan, "intercept-ext" in conf/dialplan/default.xml is configured to catch ***${destination_number}*. Pressing the line-button on a ringing extension will cause the phone to call intercept-ext with ** and the ringing extension and the phone will be connected to the caller - a perfect setup for an attendant console.

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Keep alive ping for non registered endpoints
by John Nash 22 Dec '15

22 Dec '15

My setup is .. Opensips A ----> Freeswitch ---> Opensips B In freeswitch dialing plan I am using Bridge command to connect Incoming call from Opensips A to Opensips B Is there any way Freeswitch can ping (using options) both A and B opensips to check if they are alive? . From documentation I see that we can enable ping for all "Registered" endpoints as well as "Gateways" but i am unable to figure out how I can use them in my scenario as no end point registers to Freeswitch and not using gateways also (I simply give IP of opensips B in bridge command)

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Free Switch Outbound calling
by Londre 22 Dec '15

22 Dec '15

Hi, I am trying to make a call from FreeSWITCH to an outside line. For some reason I can receive calls but I can't make any. FreeSWITCH is saying that the reason is no answer yet the destination phone does not ring at all. Nor does it put it through to voicemail. I am not getting an red error logs either. Please advise Londre Blocker Software Developer iMerchant Systems Inc.

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NOTIFY messages and outbound-proxy on a profile
by Igor Olhovskiy 22 Dec '15

22 Dec '15

Hi! May be someone have a clue... I have presence (also and MWI) enabled on a internal profile alonf with outbound-proxy on this profile. But NOTIFY massages goes directly to client's IP addresses, not to outbound proxy. Can this behavior be changed, so outbound-proxy will affect also presence part? -- Best regards, Igor

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Playback is not working
by Vikas Kumar 22 Dec '15

22 Dec '15

Hi All, I am getting this log, but I am not able to hear any sound or audio files are not being played. But logs says : 2015-09-23 10:46:51.722460 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms 2015-09-23 10:46:55.782300 [DEBUG] switch_ivr_play_say.c:1708 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-too-small.wav 2015-09-23 10:46:55.782300 [DEBUG] switch_ivr.c:191 Codec Activated L16@8000hz 1 channels 20ms 2015-09-23 10:46:55.882377 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2015-09-23 10:46:55.882377 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2015-09-23 10:46:55.882377 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms 2015-09-23 10:47:00.482235 [DEBUG] switch_ivr_play_say.c:1708 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav 2015-09-23 10:47:00.482235 [DEBUG] switch_ivr.c:191 Codec Activated L16@8000hz 1 channels 20ms 2015-09-23 10:47:01.602224 [DEBUG] switch_ivr_play_say.c:585 Raw Codec Activated 2015-09-23 10:47:01.602224 [DEBUG] switch_core_codec.c:219 sofia/internal/1012(a)119.9.73.183 Push codec L16:70 2015-09-23 10:47:03.602229 [DEBUG] switch_core_codec.c:244 sofia/internal/1012(a)119.9.73.183 Restore previous codec GSM:3. 2015-09-23 10:47:03.602229 [DEBUG] mod_voicemail.c:1244 Message is less than minimum record length: 3, discarding it. 2015-09-23 10:47:03.602229 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2015-09-23 10:47:03.602229 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-too-small.wav] (en:en) 2015-09-23 10:47:03.602229 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms Did anybody faced this issue??. Please let me know..Its urgent.!!! Thanks vikas Disclaimer :- The information in this e-mail and any attachments is confidential and may be legally privileged. It is intended solely for the addressee or addressees. If you are not an intended recipient, please delete the message and any attachments and notify the sender of misdelivery. Any use or disclosure of the contents of either is unauthorised and may be unlawful. All liability for viruses is excluded to the fullest extent permitted by law. Any views expressed in this message are those of the individual sender, except where the sender states them, with requisite authority, to be those of the specific TIMES GROUP company.

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libmp4v2-2 & deb-multimedia.org
by Gordon Hemping 22 Dec '15

22 Dec '15

I am trying to install freeswitch on a mint-debian VM. I am following the instructions on the Debian 8 Jesie page. apt-get is dying when it can't find libmp4v2-2. I understand that deb-multimedia is no more - what is the work around? Gordon Hemping Technology Unlimited Group - SDVOSB & WOSB 3252 Rosecrans St. San Diego, CA 92110 This email and any files transmitted with it are private and intended solely for the use of the individual or entity to which they are addressed. If you have received this email in error please notify the sender. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the company. Finally, the recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email.

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Alternative to loopback endpoint?
by René Weiss 22 Dec '15

22 Dec '15

Hi, I'm building an ESL conference application which calls/adds members after the conference moderator has entered ("outbound conference calls"). For this I would like to use the existing dialplan rules to decide if the call is internal or has to go over a specific gateway. In other applications I'm using the "loopback" endpoint for this, but it seems that I can't use it here, because if I'm using loopback then it plays the sound of the dialing into the conference. Is there another way to use the dialplan rules from ESL or a way to "silently" use the loopback endpoint? Thanks, René

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Understanding BW test in verto protocol
by Carlos González Florido 21 Dec '15

21 Dec '15

Hello, we are developing an open source, distributed, scalable media processing framework (using excellent Freeswitch as the media core) and we are implementing a Verto protocol proxy as part of it. We have some doubts related to the bandwidth test implemented in 1.6.5. Please tell me if I am correct: 1. First, the client sends "#SPU (number1)" where number1 is the amount of bytes it is going to send next with "#SPB" messages (262.144 in my case). 2. Then it finishes with a "#SPE" message. 3. The server answers then with a #SPU (number2), and seems to send 10x bytes (2.621.440 in my case) What does number2 mean? 4. Then, the server finishes with a "#SPD (number3)". Again, what does number3 mean? More questions would be: - Is it possible to put several (many) of concurrent 'invites' over the same verto connection or you really need to open a new connection for each call? Would it be efficient? - Is it possible to 'insert' a channel into Freeswitch without using any endpoint? (Verto or SIP) I mean, an API for sending an SDP and destination number (over the event socket connection for example) and getting a reply SDP from Freeswitch? Thank you very much in advance, Carlos Gonzalez.

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