- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_fifo && limit usage
by Борисов, Дмитрий / Dmitriy Borisov 08 Feb '17

08 Feb '17

Hi, All! I have a problem with limiting number of calls to fifo consumer. I have next dialplan to apply limits: <extension name="call-limit" continue="true"> <condition field="${caller_id_number}" expression="^(\d{3})$"> <action application="set" data="auto_hunt=true"/> <action application="limit" data="hash 192.168.1.44-lines $1 1"/> </condition> </extension> <extension name="call-limit" continue="true"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="auto_hunt=true"/> <action application="limit" data="hash 192.168.1.44-lines $1 1"/> </condition> </extension> <extension name="limit_exceeded"> <condition field="destination_number" expression="^limit_exceeded$"> <action application="hangup" data="USER_BUSY"/> </condition> </extension> and next dialplan to register in fifo queue: <extension name="Agent Login"> <condition field="destination_number" expression="^551$"> <action application="answer"/> <action application="set" data="result=${fifo_member(add operators {fifo_member_wait=nowait,fifo_consumer_exit_key=none}loopback/${user_name} 1 10)}"/> <action application="log" data="INFO Add FIFO agent result: ${result}"/> <action application="log" data="INFO User Login:${user_name}"/> <action application="sleep" data="1000"/> <action application="hangup"/> </condition> </extension> In this situation limit is zeroed when loopback channel is hangup (after successful bridge). I trying next variant: <extension name="Agent Login"> <condition field="destination_number" expression="^551$"> <action application="answer"/> <action application="set" data="result=${fifo_member(add operators {fifo_member_wait=nowait,fifo_consumer_exit_key=none}user/${user_name} 1 10)}"/> <action application="log" data="INFO Add FIFO agent result: ${result}"/> <action application="log" data="INFO User Login:${user_name}"/> <action application="sleep" data="1000"/> <action application="hangup"/> </condition> </extension> But in this variant limit is not checked and if operator make a outbound call, he received incoming call when talking. How can I limit number of calls on fifo operator line? -- -- С уважением, Борисов Дмитрий Отдел облачных технологий 1С-Рарус Тел. +7 (495) 231-20-02 доб.: 15-94 -- with best regards, Dmitriy Borisov 1C-Rarus Cloud Services tel.: +7 (495) 231-20-02 add.: 15-94

2 3

India Progress Tones
by Gonzalo Gasca Meza 08 Feb '17

08 Feb '17

Anybody has configured conf/autoload_configs/spandsp.conf.xml For India? Thanks

1 1

Link an incoming call to gateway
by Varsha Agarwal 08 Feb '17

08 Feb '17

Hi, We have multiple gateway registration with internal profile for inbound calls. These registrations could be with same IP address but different extensions/users. In our public dial plan, we need to link an incoming call with the gateway and transfer it to different destination based on which gateway call came over. We tried setting channel variables in gateway but cannot access them in dial plan. They are always empty. Is there a way to achieve this? Sample gateway file <include> <gateway name=gw1> <param name="username" value="xxxxx"/> <param name=realm" value="xxx"/> <param name="from-user" value="1001"/> <param name=register" value="true"/> <variables> <variable name="curGateway" value="gw1"/> <variables> </gateway> <gateway name=gw2> <param name="username" value="xxxxx"/> <param name=realm" value="xxx"/> <param name="from-user" value="1002"/> <param name=register" value="true"/> <variables> <variable name="curGateway" value="gw2"/> <variables> </gateway> </include> ${curGateway} is always empty in dialplan. Thanks, Varsha

2 2

Slow to open the inbound RTP port - wait for 183 response
by martin.hoole＠emailn.de 07 Feb '17

07 Feb '17

I am trying to figure out why Freeswitch is sometimes slow to open the inbound RTP port. Have you run into anything that might affect that aspect of FS's behavior? FS seems to wait for the 183 response following an Invite. But if the remote side starts sending RTP data to the port FS said to use (in the SDP), then the RTP bounces back and the sender might switch to a different source port. And then we can end up with asymmetrical port relationships, and possibly one-way audio. So I'm looking for a mechanism to tell FS to get that RTP port open as soon as possible, and not to wait for the 183 response. Might have something to do with the "ignore-early-media" setting, but not sure yet?

2 2

Change any character # with mod_translate?
by José Lopes 07 Feb '17

07 Feb '17

Hello, I am trying to change on the destination number, the character # to %23. I know that this is possible using lua script, but that it is also possible using mod_translate? Thanks in advance. Best Regards, Jose Lopes

2 2

Japanese Audio Prompts
by Dayton Turner 07 Feb '17

07 Feb '17

Hi list, I'm wondering if anyone has put together (and can share) an audio pack for FreeSWITCH in Japanese? Some googling doesn't uncover much, sadly! -Dayton <http://www.getpostbox.com><http://www.getpostbox.com>

2 1

Transcoding G729 to PCMA fails
by Igor Potjevlesch 07 Feb '17

07 Feb '17

Hello, I don't figure out why I can't make my calls transcoded from PCMA to G729. Here is the scenario: I send the call to external interface with G729. The call is properly sent to the leg B with PCMA. When the call is pick up by leg B, FS sends 488 to leg A after 200K/ACK with leg A. The module mod_com_g729 is properly loaded and has channels available. Thanks for your advice to diagnose this. Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 7

generate deliberate packet loss?
by jungle boogie 07 Feb '17

07 Feb '17

Hi All, I would like to make a sip call through my freeswitch server to a device registered onsip and cause packet loss on either side of the call (not simultaneously, though). Is something I would be able to do through fs_cli? I know a little bit about the uuid_jitterbuffer, but that seems to cause delays with all the packets eventually arriving. Thanks!

3 4

Prevent Call Transfer from a specific ip network(s)
by Lợi Đặng 07 Feb '17

07 Feb '17

Hi guys, I have 2 network-ip list A, and B that has processed similarly in xml_dialplan. Call Scenario is pretty simple: A/B -> FS -> C. For some reasons not listed, I'm using `proxy-refer=true` to proxy REFER msg to C. Using this setting make `force_transfer_context` useless. But I want to apply REFER from A only, REFER from B needed to be rejected with 403/405 or something. All REFERs are within dialog (RFC3515) I know I can place a SIP proxy before FS to achieve this. But I don't want to add a SIP component at the moment. So, can you guide me how to do this with FreeSwitch? rgds, Loi Dang Thanh Phone : +84. 1224.735.448 Email : loi.dangthanh(a)gmail.com

3 3

How are you using FreeSWITCH?
by Ken Rice 06 Feb '17

06 Feb '17

Hey Guys, A lot of you are using FreeSWITCH for some fairly complex things. However that is not always clear to people discovering FreeSWITCH for the first time. I would like to build a list of examples. If you don't mind this info being compiled on Confluence. Please Reply with who, what, and where so we can document it and show off some of the really cool FreeSWITCH based or FreeSWITCH using deployments out there. Thanks Ken

20 24