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portaudio callout only PCMA (but not G722)
by Tamer Higazi 12 Feb '20

12 Feb '20

Hi people, I try to figure out why I am only cappable to callout with portaudio with PCMA and not with G722 In the dialplan I have this: <action application="set" data="media_mix_inbound_outbound_codecs=true"/> otherwise I am not capable to call out. Do I need any other modules or codecs which I forgot to compile ? When placing the call I see this: 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101@8000 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:3837 Set Codec sofia/external/08000xxxxxx PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2020-02-12 15:02:57.593179 [DEBUG] switch_core_codec.c:111 sofia/external/08000xxxxxx Original read codec set to PCMA:8 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101@8000 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5909 sofia/external/08000xxxxxx Set 2833 dtmf send payload to 101 recv payload to 101 When I call from my sip hard phone I see this: 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match any ideas ? best, Tamer

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freeswitch Unimrcp Issue
by Arjunstl 12 Feb '20

12 Feb '20

i want to test freeswitch ASR with unimrcp. in which i want to test only my words of grammar, for an example if i speak certain words and if those words are defined into my grammar then it will go ahead else it will not. can you please help me out regarding this requrement? my environment is : uniMRCP Server : Ubuntu 16.04.6 LTS FreeSWITCH Server : Debian GNU/Linux 9.10 (Stretch) my point of concerns are: 1. i'm testing a dialplan in which i'm loading grammar path from the same server of freeswitch. but its not loading. sending grammar-loading-failure. how to use grammar for scripts. i'm stuck, what and how to load the grammar path for testing. is anything i'm missing dering configuration. 2. i want, when i make a test, then it should recognize only those words which i defined into my grammar or phrases. apart from those words, it shouldn't recognize. can you please help me out to resolve these issues. any help or suggestion, i will be greetful. thanks -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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MultiTenant
by Sean Devoy 12 Feb '20

12 Feb '20

Hi, I am setting up new server with Multi-Tenant. I have one working already v1.8 on external IP address, but cannot get the new version 1.10 working on a LAN 10.0.0.n. I have gone over these steps many many times: https://freeswitch.org/confluence/display/FREESWITCH/Multi-tenant. I am just trying to get internal extensions working at this point with a FQDN. On the v1.10 server I get registration errors that say: 2020-02-11 13:16:32.986364 [WARNING] sofia_reg.c:2929 Can't find user [111(a)10.0.0.34] from 10.0.0.182 You must define a domain called '10.0.0.34' in your directory and add a user with the id="111" attribute and you must configure your device to use the proper domain in its authentication credentials. However, the phones are registering with the FQDN, not IP address. What am I missing? Thanks, Sean

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ESL Sending INFO Message for uaCSTA Purposes
by Brent Bailey 11 Feb '20

11 Feb '20

I am trying to send an INFO message via ESL but I am having trouble. It looks like the logic in mod_sofia.c doesn't support external devices behind nat. When I generate the event via ESL as shown below the SIP INFO message has a to-uri with the private nat ip address and the message never reaches the phone. Is there a better way to sent CSTA INFO based messages or is there another way to hand craft SIP messages via ESL? Using sendevent NOTIFY will properly build the SIP message but since it isn't the rfc dictated INFO message it will not work (strangely enough the NOTIFY message is being used for CSTA messages for sendevent SWITCH_EVENT_PHONE_FEATURE). See mod_sofia.c at line 5699 for SWITCH_EVENT_SEND_INFO: https://github.com/signalwire/freeswitch/blob/v1.10/src/mod/endpoints/mod_s… sendevent SEND_INFO profile: internal from-uri: sip:1003@labDomain.com to-uri: sip:1003@labDomain.com local-user: 1003(a)labDomain.com content-type: application/csta+xml Content-Disposition: signal;handling=required content-length: 275 <?xml version="1.0" encoding="UTF-8"?><MakeCall xmlns=" http://www.ecma-international.org/standards/ecma-323/csta/ed3 "><callingDevice>sip:1003@labDomain.com </callingDevice><calledDirectoryNumber>1016</calledDirectoryNumber><autoOriginate>doNotPrompt</autoOriginate></MakeCall> Thank you. Brent Bailey

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 163, Issue 12
by Dragos Oancea 11 Feb '20

11 Feb '20

maybe increase this in switch.conf.xml : <param name="sessions-per-second" value="30"/> On Mon, Feb 10, 2020 at 11:05 PM VERMA Vineet (BCS/PSD) via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "VERMA Vineet (BCS/PSD)" <vineet.verma(a)bics.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 10 Feb 2020 23:05:01 +0000 > Subject: RE: FreeSWITCH-users Digest, Vol 163, Issue 12 > Dears, > > I am facing issue on freeswitch release 1.10 that after reaching to 33 > calls per second rate. > Freeswitch chokes the caps with rate of 16 calls per seconds only. > > Can you please help me to overcome this issue? > > Thanks, > Vineet verma > > > -----Original Message----- > From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> On > Behalf Of freeswitch-users-request(a)lists.freeswitch.org > Sent: 06 January 2020 18:58 > To: freeswitch-users(a)lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 163, Issue 12 > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific than > "Re: Contents of FreeSWITCH-users digest..." > This e-mail cannot be used for other purposes than BICS business use. See > more on https://bics.com/mail-disclaimer > > > > ---------- Forwarded message ---------- > From: "VERMA Vineet (BCS/PSD) via FreeSWITCH-users" < > freeswitch-users(a)lists.freeswitch.org> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 10 Feb 2020 15:05:43 -0800 (PST) > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 163, Issue 12 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com

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Cli command for check specific user registration status
by kumar uppu 11 Feb '20

11 Feb '20

HI All, How to check specific Registered user expiry time are creation time can one help how to achieve this thing. Thanks -- Kumar

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RE: FreeSWITCH-users Digest, Vol 163, Issue 12
by VERMA Vineet (BCS/PSD) 10 Feb '20

10 Feb '20

Dears, I am facing issue on freeswitch release 1.10 that after reaching to 33 calls per second rate. Freeswitch chokes the caps with rate of 16 calls per seconds only. Can you please help me to overcome this issue? Thanks, Vineet verma -----Original Message----- From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: 06 January 2020 18:58 To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 163, Issue 12 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." This e-mail cannot be used for other purposes than BICS business use. See more on https://bics.com/mail-disclaimer

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Refer and Cannot Blind Transfer 1 Legged calls
by David Villasmil 07 Feb '20

07 Feb '20

Hello all, I'm trying to create a new INVITE using REFER from kamailio. Sending FS the first leg (which is answered in the dialplan and sent to a python script.) And then sending a REFER to FS to call another endpoint. I've tried and i'm always getting *Cannot Blind Transfer 1 Legged calls* and getting a NOTIFY with 404 Not Found Should this work? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Dropped calls
by Sean Devoy 06 Feb '20

06 Feb '20

Hi, I have a customer that has reported LOTS of dropped calls over a 6-month period. She claims several a day. I have replaced the phone and the network cabling with no change. I have reviewed many log files. There are no errors reported and the calls all end with a hang-up, mostly from her leg but occasionally the other end. It happens inbound and outbound. It has just occurred to me that if the RTP stopped working, it would appear to be a dropped call to the user who would hang-up. Other users at this site report occasional or rare dropped calls, but the main problem is then receptionist who clearly gets most of the calls. They are NAT connections through a SonicWall firewall that I hate and may be the problem. The server (FS 1.8) is shared with other users/companies and is not NAT. Does anyone have any suggestions for diagnosing this maddening problem? Thank you, Sean

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Can execute_on_media_timeout try another peer?
by David P 06 Feb '20

06 Feb '20

If we lose RTP from a peer and a MEDIA_TIMEOUT occurs on leg B, can execute_on_media_timeout be used to dial that peer again or another one? The "transfer" example at https://freeswitch.org/confluence/display/FREESWITCH/execute_on_media_timeo… implies that this is possible, but the example isn't self-explanatory. I didn't find a confluence entry about "transfer" itself but I did find: https://freeswitch.org/confluence/display/FREESWITCH/transfer_on_fail https://freeswitch.org/confluence/display/FREESWITCH/transfer_to Which should be used for my scenario? We'll want to play an audio recording before doing the retry, too. Should we use https://freeswitch.org/confluence/display/FREESWITCH/transfer_ringback Cheers, David

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