- FreeSWITCH-users - FreeSWITCH Mailing Lists

discriminate between session answered by users and voice mail automatic messages
by sharigo roma 27 Feb '16

27 Feb '16

Dear freeswitchers, is there a way to understand if a new session originated from a python handler has been actually answered by the remote user or by the operator voice mail with a message like "the customer you are trying to reach is not available at the moment" ? The python code looks something like the following: def handler(session, args): ## I do various things.... and than session_string = str("sofia/gateway/MY_GW/0039123456789”) new_session = freeswitch.Session(session_string) if new_session.ready(): # I get here also in case of automatic voice mail message else: # I supposed i’d get here in case of automatic “unreachable” messages… Thanks in advance Lorenzo

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How to kill call/channel on the reception of CUSTOM avmd::beep
by Aqs Younas 26 Feb '16

26 Feb '16

Hi, I have subscribed to CUSTOM avmd::beep and this is what I get. "Event-Subclass": "avmd::beep", "Event-Name": "CUSTOM", "Core-UUID": "11e64e66-f81b-4d23-a8e1-e6af0eaa0bbc", "FreeSWITCH-Hostname": "debian", "FreeSWITCH-Switchname": "debian", "FreeSWITCH-IPv4": "192.168.10.59", "FreeSWITCH-IPv6": "::1", "Event-Date-Local": "2016-02-25 10:46:17", "Event-Date-GMT": "Thu, 25 Feb 2016 15:46:17 GMT", "Event-Date-Timestamp": "1456415177487145", "Event-Calling-File": "mod_avmd.c", "Event-Calling-Function": "avmd_process", "Event-Calling-Line-Number": "550", "Event-Sequence": "3174", "Beep-Status": "stop", "Unique-ID": "80d33754-b234-47e2-b7c5-16ffa20f359c", "call-command": "avmd" Now I need to hangup that call(beep detected), but could not find any event variable connected to the call. How can i kill the call. Any pointer is much appreciated. This email has been sent from a virus-free computer protected by Avast. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Transcoding and recording
by Oleg Stolyar 26 Feb '16

26 Feb '16

Hi guys, if I record a session where both A and B use the same codec, will transcoding occur? Basically, in this case will the same codecs gain any CPU advantage?

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Call gets disconnected on write frame error
by Chris Young 26 Feb '16

26 Feb '16

Hi all, Recently, we've been experiencing a problem whereby FreeSWITCH disconnects a call after receiving a frame of audio data with a data length of 0. When this happens we see the following messages in the log: [DEBUG] switch_core_io.c:1503 Engaging Write Buffer at 320 bytes to accommodate 0->320 [ERR] switch_core_io.c:1518 Write Buffer 0 bytes Failed! and the call gets disconnected. This normally happens when trying to join a conference but we have also seen it when bridging two calls together. Is the call disconnection by-design or just a side-effect of having received the bad frame of data? Thanks, Chris

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Multiple device registration on single extension
by Shisheer Teli 26 Feb '16

26 Feb '16

Dear Team, Multiple device registrations on a single extension are not working. I made changes in internal profile as "<param name="multiple-registrations" value="true"/>" . but still not working. I am using FreeSWITCH (Version 1.6.5 git 64bit) on Linux -- Regards, Shisheer T

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no audio after hold
by Nick 'tarantul' Novikov 26 Feb '16

26 Feb '16

Hello I have some problem with sound after hold usage. Sometime the sound disappear after callcenter agent shift unhold.For callcenter agents we use sip.js (version 0.7.2). In freeswitch logs I see strange lines ( https://pastebin.freeswitch.org/24557) Can anyone explain, what happened with audio and how to fix it? -- tarantul Dios es Amor

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FreeSwitch second recovery
by Gorbachev Alexey 26 Feb '16

26 Feb '16

I installed FreeSwitch cluster according to official manual -https://wiki.freeswitch.org/wiki/Freeswitch_HA And it works, when I power off first node current calls successfully move to second node and voice disappears only for 3 second. The problem is when I power on first node, server starts FreeSwitch, and FreeSwitch during start up clears calls in database and of cause I can't move current calls back again to first node. Can I move current calls between servers without interruption again? Thank you. Best regards, Gorbachev Alexey, System administrator Linux LEOMAX Group Tel: +7(495)967-7797, ext. 4119, Cell: +7(926)982-9520 Mail to: alexey.gorbachev(a)leomax.ru

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gsmopen problem: INCOMPATIBLE_DESTINATION
by Nandy Dagondon 25 Feb '16

25 Feb '16

Hello guys, I installed mod_gsmopen. When testing incoming call, it hangs up with this error message INCOMPATIBLE_DESTINATION. I understand this is a codec negotiation issue. Looking at the CDR - the gsmopen presented L16 8000 in the SDP. By default, FS does early negotiation. Snippet of my internal sip_profile: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="presence-privacy" value="$${presence_privacy}"/> <param name="inbound-codec-negotiation" value="generous"/> Take note: there is no disable-transcoding parameter which I understand FS is transcodind. In vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h ,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> I tried to add L16 at the end of global_codec_prefs which is unnecessary, isn't it? But it didn't work, too. Finally, I added L16 in the softphone, it worked. I have installed gsmopen before and it worked (transcoding) with IP phones. Is there anything I may have missed? I appreciate for any leads. Thank you, /Nandy

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Getting 503 when starting automatically via systemd
by freeswitch.opencode＠spamgourmet.com 25 Feb '16

25 Feb '16

I'm using Arch Linux (specifically Arch Linux ARM on a Raspberry Pi Zero). I'm successfully done a make and install for freeswitch 1.6.6 and when I run it manually (freeswitch -nonat), it works fine. Also, when I start it manually via systemd (systemctl start freeswitch), it still works fine. However, when I enable freeswitch.service to start automatically and reboot, the freeswitch log shows 503 errors trying to register out to my gateway, and my internal client (an Obi200) gets a 503 when trying to register in to FreeSWITCH as well; this keeps happening repeatedly (waiting 30 seconds and then failing again). However, if I restart the service (systemctl restart freeswitch), connectivity then starts working. Would the 503 be what's reported up if there's no network connectivity either way? I'm using the Debian systemd *freeswitch.service file (with very minor tweaks for my configuration on Arch Linux). Has anyone encountered this problem before (on Debian, Arch Linux, or elsewhere)? The overall .service file is very similar to sshd.service, which doesn't have connection problems, so my guess is there's something specific to FreeSWITCH going on. If this is Arch Linux-specific and I ask in their forums, is there anything different about how FreeSWITCH handles connectivity (perhaps caching some kind of handle) that would help point things in the right direction? Thanks, David

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Freeswitch/FusionPBX sales force integration
by Joseph Waite 25 Feb '16

25 Feb '16

Evening all Does anyone know of an integration with sales force? Looking for click to dial and screen pops primarily. With some kind of call recording integration being ideal. Open source preferred, commercial solutions considered! Joe Waite

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