November 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Ability to transport multiple audio streams on the same peerconnection
by Martin Hoole 11 Nov '15

11 Nov '15

2 1

uuid_display only affects called party name and not called party number
by Steven Szeto 11 Nov '15

11 Nov '15

I am trying to find a way to update the originator's display with the correct called party name and number. * The call scenario starts with making a call to the originator, and parking him: originate sofia/internal/3004(a)10.47.32.159 &park default default 'John' 3004 * The originator answers the call, and is parked. The phone display shows "John 3004" * The originator is then transferred to the terminating party: uuid_transfer d407f205-7942-442a-adbc-7bd787731d23 bridge:sofia/internal/3006@10.47.32.159 inline * After the transferred call is answered, the originator's display still shows "John 3004". So let's update the display with the terminator's name and number: uuid_display d407f205-7942-442a-adbc-7bd787731d23 "Jack 3006" Issue: the originator's phone display shows "Jack 3006 3004" Is there a way to remove the "3004" from the originator's phone display? Thanks, Steve ________________________________ NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

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@ in username
by Michael Nielsen 11 Nov '15

11 Nov '15

Is @ in username allowed in FreeSWITCH or SIP in general? I've tried to setup FS with such usernames, but my Bria client doesn't allow it.

2 3

Conference using Draytek Vigor ATA-24 adapter
by Yehavi Bourvine 11 Nov '15

11 Nov '15

Hello, If someone is using this product, do you have an idea how to do 3-way conference when the port is defined as "simple supplemental services"? I can find nothing in the docs, nor at the internet... Thanks, __Yehavi:

1 0

mod_callcenter multiserver doesn't work
by Алексей Морозов 11 Nov '15

11 Nov '15

Hello. We have two freeswitch, load balansing schema is implemented. All work well but mod_callcenter. We have recognised that if member was queued by one freeswitch in cli of other freeswitch appears messages: 2015-11-10 14:09:46.068909 [DEBUG] mod_callcenter.c:1108 Updated Agent user1@tl0001 set state = Receiving 2015-11-10 14:09:46.068909 [ERR] mod_callcenter.c:1522 Member 4957211388 <4957211388> with uuid 87916504-879b-11e5-b8ef-47aae95f691f in queue callcenter@tl0001 is gone just before we assigned an agent 2015-11-10 14:09:46.068909 [DEBUG] mod_callcenter.c:1108 Updated Agent user1@tl0001 set state = Waiting and after that member "hangs" in queue for ever. Please, any help with this problem. WBR Alex M.

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mod_conference, api, create
by roy j 11 Nov '15

11 Nov '15

Hi, all. Is there any ability to dynamically create conference room via api with method different from 'conference <confname> dial <dial-string>'? Our goal is place incoming call to room and if there is no that specific room, create it.

2 4

mod_odbc_cdr VS mod_cdr_pg_csv
by Victor Chukalovskiy 10 Nov '15

10 Nov '15

Good day, Are both mod_odbc_cdr and mod_cdr_pg_csv actively used these days? Or is one of these recommended over the other for writing into PG database? I notice mod_cdr_pg_csv doesn't have a confluence page, only old wiki page.... this is somewhat alarming. Thank you! -Victor

2 1

NAPTR and TCP transport
by Stanislav Sinyagin 10 Nov '15

10 Nov '15

If I specify a bridge destination as "sofia/external/USER@DOMAIN", and that domain has a NAPTR record specifying TCP transport as more preferred, FreeSWITCH is still sending a UDP INVITE. If the domain has only a NAPTR record for TCP transport, the call fails without sending any INVITE. It works with TCP if I add ";transport=tcp" to the bridge string. does this look like a bug, or it was set so with some reason?

3 17

Detect hangup
by Shan Randhawa 10 Nov '15

10 Nov '15

Hi, i need some help. Is there anyway to detect that user hangup (means user press endcall button - diconnects ) in lua. I have tried session:getstate() but its not changing. I have also tried to hookup hangup but it also of no avail. Thanks.

3 4

bridging in video conference, bad video quality
by emiliano olivier 10 Nov '15

10 Nov '15

Hello, thanks you in advance for all help provided. I have successfully recompiled the master project of freeswitch Correcly downloaded compiled and configured libav and all needed dependencies. The freeswitch is correcly configured, i have passed 2-3 months studying it, no warning in fs_cli, all work well. my problem concern video quality in video call 720p or 1080p. The relevant information about video conference in xml are this: the video size of the frame generated in trascoding (mux) are sized correcly 1280x720 but the quality of the video is very low and SD. The same problem is present using VP8 codec, or H263+/H263 trascoded via libav or in H264 trascoded via OpenH264 mod The problem is present even with only 2 member in conference. The virtual machine is ubuntu 14.04.1 quad core machine 4GB di ram. can someone help in some manner? i am a bit desperate. thank you Emiliano Oivier -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/bridging-in-video-conference-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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