February 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Simple question regarding bridge parameters
by Joel Serrano 08 Feb '17

08 Feb '17

Hi all, I have a simple question regarding {}, [] and bridge application when using multiple groups/failover. For example, saying the following bridge: <action application="bridge" data="sofia/outbound/${destination_number}@ $${SBC1}|sofia/outbound/${destination_number}@$${SBC2}"/> It will try SBC1, if it fails it will try SBC2. Fine till here. If I want to add {} and [] params, how would be the correct format: 1- Once, at the beginning, and it would apply in each destination (in this example 2, but it could apply to any number of destination we add with more | separators): <action application="bridge" data="{absolute_codec_string='PCMU,PCMA'}[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC1}|sofia/outbound/${destination_number}@$${SBC2}"/> 2- Once before each destination: <action application="bridge" data="{absolute_codec_string='PCMU,PCMA'}[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC1}|{absolute_codec_string='PCMU,PCMA'}[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC2}"/> 3- The {} only at the beginning, but the [] before each destination: <action application="bridge" data="{absolute_codec_string='PCMU,PCMA'}[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC1}|[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC2}"/> Maybe they all work, I'm just not sure what is the correct way, or if there are any best practices on this matter. Thanks, Joel.

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disabling param enable-timer
by Ján Füri 08 Feb '17

08 Feb '17

Hi all, I have a question about disabling enable-timer. In my sip_profile i have <param name="enable-timer" value="true"/>. But one of our customer has a problem with it and his calls are ended after 90 sec. I think 90 sec. is time that customer's pbx is set for rtp timer. I don't want to change my sip profile enable-timer to false because it would influence all our customers. Is that possible to disable rtp timer in a dialplan ? BR, Jan Furi

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Allison prompt set for FreeSWITCH
by Brian West 08 Feb '17

08 Feb '17

FreeSWITCHers, Last week we finalized a few things we wanted to accomplish, One was we've been asked to make an Allison Smith prompt set for FreeSWITCH, She's agreed to do the entire sound set and now we have to raise the money to pay her for this. I would like to thank everyone that has donated to our dev week... We are very grateful our community got so involved. Here is the link to the GoFundMe for the Allison prompts (full sound set, including the zrtp words): https://www.gofundme.com/allison-prompts-for-freeswitch The total price for this was a little over $10,000, but we've had someone who shall remain nameless donate almost 50% of this total, Now we have to raise the rest. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest

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Module Loading Timeout Exit on Start-up
by Shane Mitchell 08 Feb '17

08 Feb '17

Hi all, Just a quick question before I start searching through the source. On FS start-up, there seems to be a 30 second timeout FS allows for modules to load, after which FS will exit. I appreciate that modules should take less than this to load, but I'm running into this timeout mainly during development, for various reasons, including when debugging the loading logic of the module (hence debugging has become a bit of a race!). Is there any way to disable or extend this module load timeout? Thanks in advance, Shane Mitchell

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FREESWITCH and Verto Communicator
by _dj 08 Feb '17

08 Feb '17

Hi, everybody! I'm new in FreeSwitch. I installed it via Quick Start FreeSWITCH Demo With Verto Communicator <https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie> . The only difference was that I generated letsencrypt certificate manually. I can call from Linphone to my freeswitch server, but I'm getting the "Waiting for server reconnection" error in 50-80% of times when accessing it via verto communicator from localhost. What's the problem? Thanks in advance. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FREESWITCH-and-Verto-Communic… Sent from the freeswitch-users mailing list archive at Nabble.com.

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uuid_getvar for multiple variables
by Ben Hood 08 Feb '17

08 Feb '17

Hi, I was wondering whether there is a multi-get command for channel variables? For example, something like uuid_getvar for multiple variables? Right now I'm using uuid_dump to get all variables and just filtering the ones that I need. But can I ask FS to give me a list of variables in one API invocation? Many thanks, Ben

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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

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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

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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

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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?

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