April 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_say / switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it
by Thorsten Göllner 25 Apr '18

25 Apr '18

Hi, I am using Freeswitch 1.6.19~64bit on CentOs 7 (with FusionPBX 4.2.2) - working fine so far. But when calling a mailbox, I can hear the greeting but can not hear the destination number. I can find this macro is used: <macro name="voicemail_play_greeting"> <input pattern="^(.*)$"> <match> <action function="play-file" data="voicemail/vm-person.wav"/> <action function="say" data="$1" method="pronounced" type="name_spelled"/> <action function="play-file" data="voicemail/vm-not_available.wav"/> </match> </input> </macro> This is leading me to track it down to the function "say". So I configured a simple inbound destination which answers the call and just makes "say : de number pronounced 12345". When calling this number I get the following error: EXECUTE sofia/internal/12(a)mydomain.cloud say(de number pronounced 12345) 2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f33b405a700 Connected. 2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f33b405a700 released. 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it 2018-04-25 15:07:09.732930 [ERR] switch_xml.c:3180 Can't find phrases tag. EXECUTE sofia/internal/12(a)mydomain.cloud hangup() I checked the source code but my coding skill is not good enough. I do not really know, where the problem is. Do you have any hint for me? Best regards, Thorsten

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Linphone Softphone crashing freeswitch
by Jai Rangi 25 Apr '18

25 Apr '18

I have noticed Linphone client app can easily crash free-switch. Noticed few restarts right after. These logs d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.128573 [WARNING] switch_rtp.c:1092 missed 2 6c1e0ae1-dbe2-4630-8d8d-3cd20ba7e639 2018-04-24 13:13:18.348574 [ERR] switch_rtp.c:930 Invalid STUN/ICE packet received 28 bytes 6c1e0ae1-dbe2-4630-8d8d-3cd20ba7e639 2018-04-24 13:13:18.368567 [ERR] switch_rtp.c:930 Invalid STUN/ICE packet received 28 bytes d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.548572 [WARNING] switch_rtp.c:1092 missed 3 d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.668567 [WARNING] switch_rtp.c:1092 missed 1 de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:445 Looking for zrtp-hash de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:400 Deciding whether to pass zrtp-hash between legs de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:408 Found peer channel; propagating zrtp-hash if set de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:323 Deciding whether to pass zrtp-hash between a-leg and b-leg de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:323 Deciding whether to pass zrtp-hash between a-leg and b-leg And freeswitch restarts, de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.968579 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. 2018-04-24 13:13:19.576345 [CONSOLE] switch_loadable_module.c:1540 Successfully Loaded [mod_logfile] 2018-04-24 13:13:19.576605 [CONSOLE] switch_loadable_module.c:1540 Successfully Loaded [mod_blacklist] 2018-04-24 13:13:19.576617 [NOTICE] switch_loadable_module.c:338 Adding API Function 'blacklist' 2018-04-24 13:13:19.577958 [CONSOLE] switch_loadable_module.c:1540 Successfully Loaded [mod_enum] 2018-04-24 13:13:19.577969 [NOTICE] switch_loadable_module.c:250 Adding Dialplan 'enum' 2018-04-24 13:13:19.577987 [NOTICE] switch_loadable_module.c:292 Adding Application 'enum' 2018-04-24 13:13:19.578009 [NOTICE] switch_loadable_module.c:338 Adding API Function 'enum' 2018-04-24 13:13:19.578029 [NOTICE] switch_loadable_module.c:338 Adding API Function 'enum_auto' Is there any setting to ignore ICE and STUN packets.

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Implementing customer callback with mod_callcenter
by Michael Avers 25 Apr '18

25 Apr '18

Hello, Looking for some best practices.. I'm trying to let users exit a mod_callcenter queue to be placed in a callback queue that will (ideally) place them first in line to be connected to an agent when one becomes available. One way I had in mind is to disconnect the caller once we saved the fact that they requested a callback (application logic, doesn't matter here). Then, originate a call that plays in a loop "press 1 to connect" that is added to the relevant mod_callcenter queue with a base score that places it first in line to be picked up as the first priority. When the call is picked up the an agent, they hear the "press 1" playback, and once they actually press 1, we execute a lua script that calls the user that requested the callback and bridge agent and user. In theory this makes perfect sense . Any thoughts? Is there a simpler solution I am overlooking? Thanks Mike

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Voicemail via web interface
by John 24 Apr '18

24 Apr '18

Hello, Can you point me to any additional information about the voice mail via web interface? I have it up and running; however if you click the play button there is no playback, if you click download it will play in MS media player. Thanks John

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Question about large confeerence
by Abaci B 24 Apr '18

24 Apr '18

Hi, I'm trying to set up a large audio conference of over 1000 users where only 1 or 2 users are talking and the rest are just listening, but after 600-700 users the CPU usage gets too high and the audio gets choppy, I understand that mod_conference is not optimized for my use case but was wondering if someone has any suggestions on what can be done to optimize for my use case. I was also thinking that maybe there is a better alternative for what I look for, such as having just the talkers on the conference and the listeners should listen to it via mod_local_stream or maybe use something like eavesdrop for the listeners but not sure if and how much performance I would gain. Thanks for any help or input

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How to add headers to an outgoing REGISTER
by Lars Paulsen 24 Apr '18

24 Apr '18

Hello, I have read the documentation under https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers and as I understood it is possible to add Headers to SIP requests simply by changing the configuration. But the documentation refers to the dialing plan. I tried to add headers to an outgoing REGISTER but failed. Because the documentation refers to the dialplan and changing headers for existing calls, I am not sure if adding headers to a REGISTER request is possible or not. Can anyone please tell me if it is possible or not and if yes, how to formulate the correct "<action application="set" .>" and "<condition>" sections in the configuration. Thanks in advance for all answers. Best Regards, Lars

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Memory Leak in 1.6.20
by Jai Rangi 24 Apr '18

24 Apr '18

Not 100 % sure, but I noticed memory leak in version 1.6.20, anyone else noticed same? Just want to check out before I create a bug report. My quick test was on aws compute t2.medium with 4 GB RAM. Downgrading back to 1.6.16 seems to be working fine. Thank you, -Jai

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RTP reception before sending any RTP packet
by Mickael Hubert 23 Apr '18

23 Apr '18

Hi list, I'd like to know if there's a way to tell FS to wait for RTP reception before sending any RTP packet ? We're facing a problem regarding early media with a PBX, NATed behind a dumb router : actually, if FS first sends RTP to the PBX, without waiting for RTP coming from the PBX, there's no audio. In some cases (related to the provider we use behind FS to establish call to the PSTN) FS isn't the 'first-shooter' of the RTP flow, and everything works fine, but when FS is the first to send RTP, it doesn't work... Because we can't change anything at our providers configs, we'd like to know if FS can "delay" RTP sending until it receives some RTP from the calling party. The schematic is following : PBX ===> FS ===> ...voice backbone... ===> Providers (FS is acting as an SBC) Thanks for your help,

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Mod_callcentre
by Andrew Colin 22 Apr '18

22 Apr '18

Hi Guys Myself and lorenzo from queuemetrics are busy starting with integration of queuemetrics into freeswitch. quick question. how does the module post its data, does it use a json api or how does it work? thanks

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IPv6 endpoints on LAN getting Error Cause: 19 [NO_ANSWER]
by Rajil Saraswat 22 Apr '18

22 Apr '18

Hello, I have two Linphone clients connected to a PBX server on my LAN. Both the sip clients are registered to the internal-ipv6 profile. Unfortunately, i am not able to call between them. I get an error [INFO] mod_dptools.c:3436 Originate Failed. Cause: NO_ANSWER Any hints on why this may be happening? Thanks

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