March 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Grandstream GXP1625 not working with Freeswitch Version 1.4.26~64bit
by Uday kumar 05 Apr '17

05 Apr '17

Hi, I have multiple Grandstream IP phone in my network. All IP phones are working but only one IP phone is not workint i.e GXP1625 with Freeswitch. I am able to register extension and also incoming call is working but when making outgoing call then only its not working. I tried same IP phone with Asterisk and its working in incoming and outgoing both. I am using this only for extension to extension calls so incoming and outgoing calls only between extensions. Is Grandstream GXP1625 supported with Freeswitch or not? *When I am dialing outgoing call then on CLI getting below logs:-* [NOTICE] switch_channel.c:1077 New Channel sofia/internal/2005101@Server_IP [ab84501a-345e-445c-8d4a-52d4d6594e21] [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/2005101@Server_IP [BREAK] [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/2005101@Server_IP [BREAK] [DEBUG] switch_core_state_machine.c:472 (sofia/internal/2005101@Server_IP) Running State Change CS_NEW [DEBUG] sofia.c:8957 sofia/internal/2005101@Server_IP receiving invite from XX.XX.XX.XX:65476 version: 1.4.26 64bit [DEBUG] sofia.c:9124 IP XX.XX.XX.XX Rejected by acl "domains". Falling back to Digest auth. [DEBUG] switch_core_state_machine.c:491 (sofia/internal/2005101@Server_IP) State NEW [DEBUG] switch_core_session.c:1062 Send signal sofia/internal/2005101@Server_IP [BREAK] [DEBUG] sofia.c:2065 detaching session ab84501a-345e-445c-8d4a-52d4d6594e21 *Output of sofia status profile internal reg 2005101:-* Registrations: ================================================================================================= Call-ID: 926642231-5060-1(a)GG.CCG.HD.CAA User: 2005101@Server_IP Contact: "" <sip:2005101@XX.XX.XX.XX:65476> Agent: Grandstream GXP1625 1.0.4.6 Status: Registered(UDP)(unknown) EXP(2017-03-16 12:25:52) EXPSECS(2409) Ping-Status: Reachable Ping-Time: 0.00 Host: Server_IP IP: XX.XX.XX.XX Port: 65476 Auth-User: 2005101 Auth-Realm: Server_IP MWI-Account: 2005101@Server_IP -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349

6 19

100% CPU usage
by Grant Bagdasarian 04 Apr '17

04 Apr '17

Hi, We're currently experiencing a weird issue with one of our freeswitch servers where the CPU just goes berserk and stays at 100%. There are no active calls at that moment, but FS is still using 100% CPU. I was unable to get a debug using gdb, since freeswitch does not crash. We had this issue on 1.6.13 and then I upgraded to 1.6.15 this morning, but the problem still persists. There is currently only one active application running on this box which is using LUA. Maybe that is causing the issue? We checked the code but there are no weird loops etc. Perhaps we are missing something? Any ideas how to trace this? Regards, Grant

6 22

Call Center Behaviour
by Jai Rangi 03 Apr '17

03 Apr '17

We are using call center module. We noticed one thing, if the agent change status to not available or on break and then after a while come back to change the status to available. His stats (num of calls answered or talk_time) reset to 0. Example, callcenter_config queue list agents sales@domain name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|no_answer_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time 101@domain|single_box||callback|[call_timeout=30]user/130@domain |Available|Waiting|0|0|0|0|0|0|0|1489009000|1488960061|1|0|0|0 102@domain|single_box||callback|[call_timeout=30]user/111@jdomain |Available|Waiting|0|0|60|60|0|1489009008|1489009016|1489009000|1488960061 *|0|1|8|0* Here I see 102, had one call for 8 seconds, Change the agent status to break. session:execute("set", "res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'On Break')}") 101@domain|single_box||callback|[call_timeout=30]user/101@domain |Available|Waiting|0|0|0|0|0|0|0|1489009000|1488960061|1|0|0|0 102@domain|single_box||callback|[call_timeout=30]user/101@domain|On Break|Waiting|0|0|60|60|0|1489009008|1489009016|1489009000|1489009163| *0|1|8|0* Change the agent state to Available On Demand, stats are still there. session:execute("set", "res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}") 101@domain|single_box||callback|[call_timeout=30]user/101@domain |Available|Waiting|0|0|0|0|0|0|0|1489009000|1488960061|1|0|0|0 102@domain|single_box||callback|[call_timeout=30]user/102@domain|Available (On Demand)|Waiting|0|0|60|60|0|1489009008|1489009016|1489009000|1489009252| *0|1|8|0* Now changing to Available and stats are reset ed to 0,0, session:execute("set", "res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available')}") 101@domain|single_box||callback|[call_timeout=30]user/101@domain |Available|Waiting|0|0|0|0|0|0|0|1489009000|1488960061|1|0|0|0 102@domain|single_box||callback|[call_timeout=30]user/102@domain |Available|Waiting|0|0|60|60|0|1489009008|1489009016|1489009000|1489009415|0|0|0|0 Is there way to control this behavior. So basically we can to control the call stats.

2 3

park_after_bridge don't working for busy destination!
by Alexander Haugg 03 Apr '17

03 Apr '17

Hi, Ich try to catch the leg A after the destination is hangup with the cause busy to signalling a busy tone stream to the caller (leg A) But leg A will dropped too. Here is the test dialplan: <context name="H3KSip_SP_outbound"> <extension name="outbound_softphone"> <condition field="${channel_name}" expression="^sofia/H3KSip_B2Bua/.*$" break="on-false"> <action application="set" data="ringback=%(1500,3000,425)" /> <action application="set" data="instant_ringback=true" /> <action application="multiset" data="ignore_early_media=true hangup_after_bridge=false park_after_bridge=true" /> <action application="export" data="nolocal:absolute_codec_string=G722,PCMU,PCMA" /> <action application="bridge" data="{ignore_early_media=true}sofia/gateway/GW_H3KSip/${destination_number}" /> <action application="log" data="ERR last_bridge_hangup_cause=${last_bridge_hangup_cause}" /> <action application="log" data="ERR bridge_hangup_cause=${bridge_hangup_cause}" /> <action application="sleep" data="1000" /> <action application="playback" data="tone_stream://%(480,480,425);loops=3" /> <action application="hangup" data="USER_BUSY"/> </condition> </extension> </context> Whats wrong? Thanks a lot!

4 5

VPX on windows?
by Alexander Haugg 01 Apr '17

01 Apr '17

Hi, in the newest main brunch I can see the vpxlib project, but after the build I did not found the libvpx.dll or a mod_vpx.dll. Is the VPX video support on windows available? Thanks a lot!

3 3

possible to use mod_fifo without parking a caller?
by Dave Horton 01 Apr '17

01 Apr '17

I’d like to use something like mod_fifo, if possible to support my use case: - caller calls in - freeswitch sends 183 (pre_answer) but does _not_ answer the call - via out of bands means, I send information to an “agent” that a caller is waiting (providing uuid, or parking slot) - agent calls in (to the “parking slot” or equivalent) - caller is now bridged with agent - OR, if no agent calls in within X seconds, caller is rejected with a SIP 480 Is something like that possible with mod_fifo, or other module? Dave

5 10

DTMF games for FreeSWITCH
by Michael Nielsen 31 Mar '17

31 Mar '17

I'm looking for play a bit with FreeSWITCH/SIP and DTMF games. Does anyone know of any free one source games for FreeSWITCH which can be played without a GUI, but via the phone?

1 0

something like transferring to an extension when caller abandons?
by Dave Horton 30 Mar '17

30 Mar '17

I’m hoping to find a dialplan-based solution to this problem. I have a call coming in, and I start executing a dialplan with various applications. At some point, if the caller cancels before I have bridged or answered his or her call, I would like the ability to immediately execute some code — e.g. by jumping to another extension (or something like that) at the moment when the caller abandons. I know this could be orchestrated in an event socket app, but as mentioned I would prefer a dialplan-based approach, if possible. Any ideas? Dave

4 16

Executing java application within an HTTAPI application
by Thomas, Tom 30 Mar '17

30 Mar '17

Greetings everyone, I was wondering if mod_java is supported by the "execute" work action in HTTAPI. So something like this within a response document: <document type="text/freeswitch-httapi"> <work> <execute application="java" data="{args_for_java_app}" ..../> </work> </document> I haven't setup a webserver for HTTAPI yet. So I wanted to know before I go down this path that this can be done. I have a java application that I currently execute from the dialplan. I need to know if httapi can use execute to do the same. Thanks a lot in advance.

1 0

mod_conference ghost and outcall flags
by David Ponzone 30 Mar '17

30 Mar '17

There is obsolete information about conference_auto_outcall_flags in Confluence. This variable is now called conference_utils_auto_outcall_flags. I wanted to update it but I don’t have the rights as I used to have on the old wiki. If anyone can do it… Also, I tested the ghost flag on a conf member and the member is then mute/deaf. For sure, it’s a total ghost :) I used to think ghost was not related to mute/deaf but only to the conference members count. Am I missing something ? David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <blocked::http://www.ipeva.fr/> - www.ipeva-studio.com <blocked::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

1 0