April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

BEARERCAPABILITY_NOTAVAIL with 1.6.6 on Debian 8
by Thomas Løcke 31 May '16

31 May '16

Hi all, Using 1.6.2 this works from fs_cli and dialplan: originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx) Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL. Has something changed between 1.6.2 and 1.6.6 that may be the cause of this, and if so, what can I do to fix it? :o) Thomas

5 6

FreeSWITCH using outbound proxy but to bypass media
by Muhammad Naseer Bhatti 28 May '16

28 May '16

I am trying to use an outbound SIP proxy for FreeSWITCH by adding fs_path to the bridge. It’s working fine and I am able to send calls using the proxy. On the proxy (which is also FreeSWITCH) I am receiving the call in a lua script, create a new session with the destination gateway and bridge two sessions (old, incoming and the newly created) The idea is to have multiple FreeSWITCH servers (all public IP address) originating calls while all calls going to the carrier via the proxy. The problem is that all media is also passing through the proxy. Is there a way I can only send signaling through proxy and media to flow directly between the endpoints? I have tried bypass_media and bypass_media_after_bridge but it’s not working. Is this even doable? Thanks. -- Muhammad Naseer Bhatti

3 4

Fax....How I hate you.
by Blake Priddy 24 May '16

24 May '16

Is anyone out there having success with sending/receiving faxes with Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I have referred myself to this document https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was just wanting to see what everyone is doing out there with flowroute for the people who still must use faxing.. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

10 21

Using freeswitch as Generic SIP SBC/RtpProxy without predefined endpoints
by Taisto Qvist 14 May '16

14 May '16

Hi, I am an OpenSIPS/Kamailio user that has spent the whole day trying to figure out which software is most suitable to build an SBC that only acts as a full b2bua with rtp-relay, with an undefined number of destinations(entire domains with lots of unknown proxies) on each side. I've been looking at yate, freeswitch, SEMS, but I am overwhelmed by the gazillions of configuration-files that seem to be needed. I have absolutely NO need of pbx/user functionality, just a B2BUA and rtp-proxy capabilities(with full codec-transparency), with the addition that I must be able to add some custom logic to decide the DNS or IP(v6) address of the next-hop sip-node. All examples I've found on freeswitch SBC config, has predefined gateways/trunks/proxies as "predefined next-hops", and I want the SBC to be able to send anywhere, depending on custom application logic. I'd stick with kamailio/rtp-proxy and the standard topology-hiding module, but unfortunately I need the full topology hiding of a b2bua. Is this easily achievable with Freeswitch? Any help with be immensely appreciated! Regards Taisto Qvist

4 3

API break vs uuid_break
by Marek Slivanský - Livispace s.r.o. 10 May '16

10 May '16

Hi there, sorry if I'm not choosing the right way for asking for something, please point me out, this is my first time using the mailing list. I have a question regarding API command "break" has changed to "uuid_break". I really tried to but I've not found any information about that. We are using "bgapi break <uuid>" to stop playing some sound to a call (leg), e.g. moh, and to continue with next dialplan item. We are using v1.2.stable. Probably since last upgrade to the latest commit in this branch (prior we were using some 09/2012 commit), sometimes (rarely) the command "bgapi break <uuid>" gets ignored by FreeSwitch and sound remains playing forever (local_stream://moh). Could anyone please shed some light on this change (and perhaps add some info why break has been deprecated and consequences of continuing to use it, into wiki)? Thanks in advance for any answer, Marek

3 3

High load kernel
by Agustí Ubalde 06 May '16

06 May '16

Hi all, We are having problems with the CPU load of Freeswitch. Specifically, there is a CPU having a charge of 70% of CPU system, always in the same CPU. Any ideas on the possible cause of this system CPU consumption? Thanks, *PRESENCE TECHNOLOGY* *Agustí Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* <http://www.presenceco.com/> *Follow us on:* *[image: tw]* <http://twitter.com/presencetech>*[image: yt]* <http://www.youtube.com/PresenceTechnology>*[image: in]* <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]* <http://www.slideshare.net/PresenceTech>*[image: fb]* <http://www.facebook.com/presencetech> For additional information, please visit our website *www.presenceco.com* <http://www.presenceco.com/> -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

5 11

Pulse / portaudio / sharing
by Rick Jarvis 06 May '16

06 May '16

I’d like to monitor the audio levels on an audio interface that’s being used by mod_portaudio I’m guessing, but I don’t know, that the best way of doing this would be to use Pulse Audio, either directly or indirectly. I don’t know a huge deal about Pulse Audio though, and while it seems that there’s a mod_pulseaudio available, I can’t find much information on it, and there seem to be quite old unanswered questions about Pulse in relation to FreeSWITCH, on this list. Can anyone give me any suggestions on which way I might head?

2 2

Message is less than minimum record length
by Brian May 05 May '16

05 May '16

Some reason my freeswitch system has suddenly stopped listening for messages. Instead it immediately cuts the talker off and says the message is too short. Any ideas? How do I debug this? i am going to conduct an audio test later on to make sure this is working, last I checked phone calls were working fine however. 2016-04-07 10:09:31.172293 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:1910 done playing file /var/lib/freeswitch/storage/voicemail/default/microcomaustralia.com.au/2000/greeting_1.wav 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2016-04-07 10:09:40.232300 [DEBUG] mod_loopback.c:601 loopback/voicemail-b CHANNEL KILL 2016-04-07 10:09:40.232300 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms 2016-04-07 10:09:44.792345 [DEBUG] switch_ivr_play_say.c:1910 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav 2016-04-07 10:09:45.912334 [DEBUG] switch_ivr_play_say.c:559 Raw Codec Activated, ready to waste resources! 2016-04-07 10:09:45.912334 [DEBUG] switch_ivr_play_say.c:673 Raw Codec Activated 2016-04-07 10:09:45.912334 [DEBUG] switch_core_codec.c:221 loopback/voicemail-b Push codec L16:100 2016-04-07 10:09:47.912303 [DEBUG] switch_core_codec.c:246 loopback/voicemail-b Restore previous codec PCMA:8. 2016-04-07 10:09:47.932384 [DEBUG] mod_voicemail.c:1245 Message is less than minimum record length: 3, discarding it. 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-too-small.wav] (en:en) 2016-04-07 10:09:47.932384 [DEBUG] mod_loopback.c:601 loopback/voicemail-b CHANNEL KILL 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms 2016-04-07 10:09:51.492336 [DEBUG] sofia.c:6858 Channel sofia/external/61419503953(a)203.31.79.18 entering state [terminated][487] 2016-04-07 10:09:51.492336 [NOTICE] sofia.c:7879 Hangup sofia/external/61419503953(a)203.31.79.18 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:707 sofia/external/61419503953(a)203.31.79.18 ending bridge by request from read function 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:780 BRIDGE THREAD DONE [sofia/external/61419503953(a)203.31.79.18] 2016-04-07 10:09:51.492336 [DEBUG] mod_loopback.c:601 loopback/voicemail-a CHANNEL KILL 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:701 sofia/external/61419503953(a)203.31.79.18 ending bridge by request from write function -- Brian May <brian(a)linuxpenguins.xyz> https://linuxpenguins.xyz/brian/

1 2

dial-string pickup possible after group call?
by Jonathan Hunter 04 May '16

04 May '16

Hi Guys, Just wondering if its possible to use the user dial-string command to do call pickup after a call has been sent to a hunt group using the bridge and group_call function? So basically a call to group 5000, sends calls to extension 1000 and 2000 which are in the group. This all works fine but I want to be able to pickup the call by dialling *8 then either 1000 or 2000. I can do this if I am not using the group_call function, and its just a normal inbound call to the extension I want to pickup, so just wondered if its possible to do a pickup of one of the extensions in the group? To start the group_call Im using in my dialplan; Dialing 5000 results in; <action application="bridge" data="${group_call(huntgroup1(a)test.com+A}"/> Which then rings both extension 1000 and 2000 which have the dial-string defined; <param name="dial-string" value="sofia/internal/${dialed_user}@test.com,pickup/${dialed_user}@test.com"/> If I add pickup after group_call, this allows be to pickup extension 5000, but I need to be able to create the pickup channel using the dial-string, am I doing this wrong or does anyone have an idea how to achieve this? And just to be clear pickup works fine for normal inbound calls, its just when implementing group_call. Many thanks Jon

1 1

Verto vs SIP
by Oivvio Polite 02 May '16

02 May '16

Summary: I'm just getting started with VoIP/SIP/WebRTC. Following the FreeSwitch Cookbook I've built a webclient that uses SIP signalling to establish a connection with FreeSwitch and then exchanges media with WebRTC. But I get the impression from a lot of sources that SIP is a bad fit for WebRTC. One datapoint that indicates this is the very existance of mod_verto. Why would you bother writing it if SIP, that has 10+ years of battle testing, was a good fit right? What I don't understand yet is why SIP is a bad fit for WebRTC? Would someone in the know care to muse about that? regards Oivvio

6 12