- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_conf : play tmp/uuid.wav!userleft on member exit.
by Stephen Dame 08 Aug '14

08 Aug '14

I’ve successfully been able to modify dialplan to prompt caller for name, record it and announce in conference per wiki. solution works great. Adding a few lines of code makes sure they can only enter existing conferences. Now that I have their name in temp wav file, trying to decide best approach to play name!” is leaving conference” into conference when they leave. Before heading down a path. Wanted to seek some advice on some possible approaches. 1) Thought I may be able to set exit sound in conf profile with UUID but not sure variables are expanded at run time. 2) The other approach is to monitor ESL conference events and play back the UUID file when its detected that they have left (separate process) 3) Modify source of mod_conf to hardcode playing of /tmp/uuid when member leaves only if the file exists. Is there an easier way I’m missing? Regards, Stephen From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, August 08, 2014 12:17 PM To: Freeswitch-users Subject: Re: [Freeswitch-users] No audio with WebRTC, cannot find certs Try deleting all the files with dtls in the name and restart to let it regenerate. Also try master rev. On Aug 8, 2014 11:13 AM, "A.B. Carroll" <ben(a)hl9.net <mailto:ben@hl9.net> > wrote: Hello, I'm experiencing similar issues as stated here: http://comments.gmane.org/gmane.comp.telephony.freeswitch.devel/6397 Running FreeSWITCH Version 1.4.7-1~64bit (-1 64bit), Debian Wheezy using the official repos With the following errors: 2014-08-08 15:32:33.116140 [ERR] switch_core_cert.c:210 FP FILE ERR! [...] 2014-08-08 15:32:33.136126 [ERR] switch_rtp.c:2974 audio DTLS cert err [5] I'm really lost here, .. I've tried using the -certs parameter, as per reading the source, and mentioned in the above post. Stock, the directory: /etc/freeswitch/tls/ existed with a file dtls-srtp.pem that is a pub+priv key combined. I tried specifying that directory with the '-certs' argv as well as copying the .pem to .crt, and even having all four files: /etc/freeswitch/tls/dtls-srtp.pem /etc/freeswitch/certs/dtls-srtp.crt /etc/freeswitch/tls/dtls-srtp.pem /etc/freeswitch/certs/dtls-srtp.crt .. exist, with mode +rw all (including the directories themselves) And no luck, same error, it seems Freeswitch just refuses to see the file, unlike the mailing list post I'm referencing. Can anyone offer some advice? P.S. I am using the freeswitch and sipML5 examples -- they work fine with the freeswitch server, so I'm assuming this is the issue. Thanks, A.B. Carroll _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

2 1

TLS cipher list
by sangdrax8 08 Aug '14

08 Aug '14

I see there is a parameter to set "tls-version" but is there a similar variable which allows you to set a list of ciphers that are acceptable? I know nginx and others you can generally specify a list to use, or specific ciphers to deny. Just can't find anything like that so far in the documentation. Thank you!

1 0

No audio with WebRTC, cannot find certs
by A.B. Carroll 08 Aug '14

08 Aug '14

Hello, I'm experiencing similar issues as stated here: http://comments.gmane.org/gmane.comp.telephony.freeswitch.devel/6397 Running FreeSWITCH Version 1.4.7-1~64bit (-1 64bit), Debian Wheezy using the official repos With the following errors: 2014-08-08 15:32:33.116140 [ERR] switch_core_cert.c:210 FP FILE ERR! [...] 2014-08-08 15:32:33.136126 [ERR] switch_rtp.c:2974 audio DTLS cert err [5] I'm really lost here, .. I've tried using the -certs parameter, as per reading the source, and mentioned in the above post. Stock, the directory: /etc/freeswitch/tls/ existed with a file dtls-srtp.pem that is a pub+priv key combined. I tried specifying that directory with the '-certs' argv as well as copying the .pem to .crt, and even having all four files: /etc/freeswitch/tls/dtls-srtp.pem /etc/freeswitch/certs/dtls-srtp.crt /etc/freeswitch/tls/dtls-srtp.pem /etc/freeswitch/certs/dtls-srtp.crt .. exist, with mode +rw all (including the directories themselves) And no luck, same error, it seems Freeswitch just refuses to see the file, unlike the mailing list post I'm referencing. Can anyone offer some advice? P.S. I am using the freeswitch and sipML5 examples -- they work fine with the freeswitch server, so I'm assuming this is the issue. Thanks, A.B. Carroll

2 1

FreeSWITCH Friday BugHunt Reminder!
by Ken Rice 08 Aug '14

08 Aug '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday BugHunt Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken http://ift.tt/vGSsGm http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

Call Rejection Drops Call
by Joel White 08 Aug '14

08 Aug '14

I am having an issue with calls coming from another SIP server into FS. Everything works fine with the call getting to its destination. And also works if the call is ignored and goes to voicemail. However if the call is rejected by a polycom endpoint the call is dropped completely and disconnects without going to voicemail. Here is a sample of the dialplan <action application="answer"/> <action application="bridge" data="sofia/internal/57331(a)192.168.100.25 "/>  <action application="voicemail" data="default ${domain_name} 57331"/> <action application="hangup"/> If someone could point me in the right direction

3 3

Unable to play audio on inbound call after uuid_broadcast bridge
by Douglas Lopes Pereira 08 Aug '14

08 Aug '14

Hi, please, consider the scenario: We have a call management gateway connected to a Freeswitch (1.4.7-1~64bit) box. When we get a call from +15301111234 we receive a CHANNEL_CREATE event we answer it with something like: uuid_answer 2edbc62c-1ef9-11e4-8e8e-0f7c1a2ea61 On the gateway we create an outgoing call to +15301115678 and also bridge the previously received call with this new call created using a broadcast command (since we still don't have the new call switchID): uid_broadcast 2edbc62c-1ef9-11e4-8e8e-0f7c1a2ea61c bridge::{origination_caller_id_number=+15301111234,origination_channel_name=c-yzdyzyk45zy6x62zutc6goq}sofia/external/+ 15301115678(a)66.111.222.3 On the first call I hear ringback tone and the second phone rings. When the second call is answered both calls are bridged and I have two way audio. I then tried to play an audio to the first call using: uuid_broadcast 2edbc62c-1ef9-11e4-8e8e-0f7c1a2ea61c http://myserver/audio.mp3 aleg I does NOT work. I simply get this debug message: [DEBUG] switch_core_session.c:1133 Send signal sofia/internal/+ 15301111234(a)66.111.222.3 [BREAK] If I try playing the audio on bleg it works: [DEBUG] switch_ivr_play_say.c:1693 done playing file http://myserver/audio.mp3 I also works if I use the second call callId pointing to the aleg but fails if I use the second callID pointing to bleg. Am I doing something wrong or could we have a bug on Freeswitch? Regards, Douglas

1 0

Mod_conference questions
by Daniel Ivanov 08 Aug '14

08 Aug '14

I started playing with mod_conference and video + webrtc. It's a bit blurry so far, but i've made it work. I am perfectly happy with the 1-video from the floor. My problem is that i can't control which user gets the floor. I want to have 1 speaker on the floor pre-defined and now they change depending on who speaks(and puts energy). Also, what controls the quality of the video streams? Is it the capabilities in the SDP the client offers that set the bandwidth for the streams?

2 1

esl api issues
by leonhe 08 Aug '14

08 Aug '14

Hi All, Recently I'm writing an Application with ESL in inbound mode. I have two issues: 1) esl_recv_event() a) Each time I call this interface, handle->last_ievent will be updated to be the oldest event. Just like de-queue the oldest event from FIFO. My assumption is right or not? b) After call this interface, may I access "handle->last_ievent" without any protection( without mutex/lock)? If there is any risk of handle->last_ievent is updated during my access, for example some other interfaces are called or new events are received from FS side. Per my experience, seems has such risk. 2) esl_execute() When I read the implementation of the code, is there any possibility to dead lock as following? esl_execute() +-->esl_send_recv /esl_send_recv_timed(_handle, _cmd, 0) +--> esl_mutex_lock(handle->mutex); <<<<< lock for 1st time +--> esl_recv_event_timed(handle, ms, 0, &handle ->last_sr_event) +--> esl_recv_event() +----> esl_mutex_lock(handle->mutex); <<<<<< lock for 2nd time, but preview one is not unlock Can anyone confirm this? BR Leon

1 1

New user question, does it have distrubtion which install like pbxinaflash
by John Joseph 07 Aug '14

07 Aug '14

Hi All, New user here. Would like to get advice on Does FreeSwitch has installation with GUI , like PBX in a flash for asterisk Thanks Joseph John

2 1

Conference auto outcall rings with 10s delay?
by MrX 07 Aug '14

07 Aug '14

Hi All, Here is my problem: we are using the Freeswitch conference(mod_conference), when conference starts, we want it to automatically call group members in, so we use conference_set_auto_outcall like this: <action application="conference_set_auto_outcall" data="user/1000@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1001@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1002@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1003@$${domain}"/> ...... but very often, there are some members which delay significantly(most about 10s, sometimes 3~4s) from conference call(3001) to it rings, while other members rings with no delay. And, if we make a second conference call instantly after last conference ends, the delay almost never happens. The longer the interval is, the bigger is the delay chance. When we tryed to use some packet capture tool to analyze, we found that SIP INVITE packet for the delayed peer was sent 10s after conference call, and there was no SIP packet at all during the 10s delay, So the delay is from serverside. The question is, may there be something wrong with our FreeSWITCH configuration, or the delay is as intended for some purpose? How can avoid the outcall delay and make all peers ring at the same time when conference starts? Thanks and Best Regard, David

8 22