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Mod_CallCenter - Strange Behaviour - Instant Abandonment of calls
by Joel White 09 Jun '14

09 Jun '14

I am noticing a strange issue where calls are being placed in a CallCenter queue, then instantly being put into an abandoned state. Here is the callcenter.xml configuration  <queue name="helpdesk@default"> <param name="strategy" value="longest-idle-agent"/> <param name="moh-sound" value="$${hold_music}"/>  <param name="announce-sound" value="it_sounds/it_helpdesk-queue-announcement.wav"/> <param name="announce-frequency" value="180"/> <param name="time-base-score" value="system"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="0"/> <param name="max-wait-time-with-no-agent-time-reached" value="20"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> </queue> And here is when a call comes in and instantly gets put into a Abandoned state 2014-06-06 14:04:02.442348 [DEBUG] switch_core_session.c:871 Send signal sofia/internal/56716 [BREAK] 2014-06-06 14:04:02.442348 [NOTICE] mod_dptools.c:1225 Channel [sofia/internal/56716] has been answered 2014-06-06 14:04:02.442348 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/56716 [BREAK] 2014-06-06 14:04:02.442348 [DEBUG] switch_channel.c:3647 (sofia/internal/56716) Callstate Change EARLY -> ACTIVE 2014-06-06 14:04:02.442348 [DEBUG] sofia.c:5828 Channel sofia/internal/56716 entering state [completed][200] EXECUTE sofia/internal/56716 callcenter(helpdesk@default) 2014-06-06 14:04:02.442348 [DEBUG] mod_callcenter.c:2493 Member 56716;phone-context=phones.mycompany <56716> joining queue helpdesk@default 2014-06-06 14:04:02.442348 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/56716 [BREAK] 2014-06-06 14:04:02.442348 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/56716 [BREAK] 2014-06-06 14:04:02.442348 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/56716 [BREAK] 2014-06-06 14:04:02.442348 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/56716 [BREAK] 2014-06-06 14:04:02.462306 [DEBUG] sofia.c:5828 Channel sofia/internal/56716 entering state [ready][200] 2014-06-06 14:04:02.462306 [DEBUG] mod_local_stream.c:498 Opening Stream [moh/8000] 8000hz 2014-06-06 14:04:02.462306 [DEBUG] switch_ivr_play_say.c:1305 Codec Activated L16@8000hz 1 channels 20ms 2014-06-06 14:04:02.742353 [DEBUG] switch_rtp.c:3757 Correct ip/port confirmed. 2014-06-06 14:04:03.582343 [DEBUG] mod_snmp.c:70 systemStats OID-suffix requested (1) freeswitch@internal> callcenter_config queue list members helpdesk@default queue|system|uuid|session_uuid|cid_number|cid_name|system_epoch|joined_epoch|rejoined_epoch|bridge_epoch|abandoned_epoch|base_score|skill_score|serving_agent|serving_system|state helpdesk@default |single_box|e92b24e0-eda4-11e3-9b8a-77fa9a892911||56716|56716;phone-context=phones.mycompany|1402077829|1402077829|0|0|1402077831|0|0|57504@default |single_box|Abandoned helpdesk@default |single_box|f0fa1898-eda4-11e3-9b95-77fa9a892911||56716|56716;phone-context=phones.mycompany|1402077842|1402077842|0|0|1402077841|0|0|56708@default |single_box|Abandoned +OK 2014-06-06 14:04:06.602338 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/56716 [BREAK] 2014-06-06 14:04:06.622340 [NOTICE] sofia.c:737 Hangup sofia/internal/56716 [CS_EXECUTE] [NORMAL_CLEARING] 2014-06-06 14:04:06.622340 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/56716 [KILL] 2014-06-06 14:04:06.622340 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/56716 [BREAK] 2014-06-06 14:04:06.622340 [DEBUG] switch_ivr_play_say.c:1708 done playing file local_stream://moh 2014-06-06 14:04:06.622340 [DEBUG] mod_callcenter.c:2630 Member 56716;phone-context=phones.mycompany <56716> abandoned waiting in queue helpdesk@default Not sure what I am missing that would cause this unexpected behaviour

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FreeSWITCH - MySQL
by Joel White 09 Jun '14

09 Jun '14

Just a question. I want to attempt to use MySQL as the DB for FreeSWITCH, many of the other systems use MySQL and I would like to keep the data in the same place. The question is, what character set and collation should I use? The FreeSWITCH site explicitly states to NOT use utf8 Thank you in advance

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(no subject)
by Dmitriy Shumaev 09 Jun '14

09 Jun '14

I'm using FS for PTT (Push-To-Talk) services. To start talking FS sends reINVITE (I use "uuid_media_reneg <_uuid>" command for this). I need to add "a=sendonly" to this reINVITE. How it can be done? With best regards, Shumaev D.A., programmer @ KBR Ltd.

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How to set ptime in SDP
by Dmitriy Shumaev 09 Jun '14

09 Jun '14

I'm trying to send 40 ms of speech in each RTP frame when working with a specific/certain gateway . By means of modifying [var.xml]: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA@8000h@40i,PCMU@8000h@40i"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA@8000h@40i,PCMU@8000h@40i"/> , it is possible to call FROM this gateway (gateway does not send ptime in SDP). But, when the call comes from the other client with ptime set to 20 in SDP, outgoing INVITE messange TO this gateway contains ptime set to 20. How can I set ptime to 40 (it's desirable to do so only for one specific gateway). With best regards, Shumaev D.A., programmer @ KBR Ltd.

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transfer_after_bridge
by Andre 08 Jun '14

08 Jun '14

HI, can I call transfer_after_bridge from execute_on_answer? If so, what is the syntax to do that? Thanks

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mod_pocketsphinx: SpeechTools.jm error
by Mark Campbell-Smith 08 Jun '14

08 Jun '14

HI, I'm trying to test pocketsphinx and have been looking at the pizza example from the wiki. When I try to execute the test, I get the following error in the log: 2014-06-05 13:29:10.176189 [ERR] SpeechTools.jm:188 Exception: SyntaxError: Unexpected identifier (near: " console_log("debug", "----Hit score " + result.interpretation.@confidence") It doesn't appear that SpeechTools.jm has been changed for a while. Any ideas?

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Freeswitch FMS server as a endpoint
by info＠shishir.com.np 07 Jun '14

07 Jun '14

Hi all, I've been scratching my head with an issue - I am trying to pass (one way) audio from Freeswitch to FMS server directly via mod_rtmp. I'm able to connect to freeswitch from FMS server ; execute the dialplan ; able to receive audio on the FMS server. But the received audio from freeswitch is very choppy and lagging. I noticed I'm passing audio one way (from freeswitch to FMS) only and not other way around is this causing the issue ? I found similar behavior on FLEX client too, if I comment out the publish method. But if I publish the audio everything is normal. Is there a work around to receive one way audio only from freeswith to FMS without loosing the audio quality? Thanks Shishir

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Custom wording with SIP call rejections
by Victor Chukalovskiy 07 Jun '14

07 Jun '14

Good day, Is there a way to do custom rejection responses in FS like below? And in particular, to use them as rejection responses from limit application? "503 Mass Calling Event" "603 Don't do this" This would be helpful to let customer see specific CAC filter a call was rejected by. Thanks! -Victor

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Codecs in SDP
by Ričardas Stoma 06 Jun '14

06 Jun '14

I have created the following extension: <extension name="my_test"> <condition> <action application="bridge" data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/ 1001(a)192.168.1.78"/> <action application="hangup"/> </condition> </extension> then i call to this extension, freeswitch dials to 1001(a)192.168.1.78 and when i check the content of SDP, included in a SIP packet, i see something like this: v=0 o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65 s=FreeSWITCH c=IN IP4 192.168.1.65 t=0 0 m=audio 17262 RTP/AVP 8 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 why my codecs from 'absolute_codec_string' are not included in SDP? I would expect to see these lines included: a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000

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Codecs in SDP
by Ričardas Stoma 06 Jun '14

06 Jun '14

I have created the following extension: <extension name="my_test"> <condition> <action application="bridge" data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/ 1001(a)192.168.1.78"/> <action application="hangup"/> </condition> </extension> then i call to this extension, freeswitch dials to 1001(a)192.168.1.78 and when i check the content of SDP, included in a SIP packet, i see something like this: v=0 o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65 s=FreeSWITCH c=IN IP4 192.168.1.65 t=0 0 m=audio 17262 RTP/AVP 8 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 why my codecs from 'absolute_codec_string' are not included in SDP? I would expect to see these lines included: a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000

2 1