August 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch team: thanks for RPMs 1.10
by Richard Chan 14 Aug '19

14 Aug '19

Dear FreeSWITCH Team Many thanks for 1.10 and the RHEL/CentOS 7 packaging.

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mod_sms questions
by Sean Devoy 13 Aug '19

13 Aug '19

Hi, I have found information on mod_sms to be limited. Our provider (Voip Innovations) supports SMS on our DIDs. Here is their document explaining handling options. https://backoffice.voipinnovations.com/Wiki/View.aspx?article=SMS Our client is using Bria Softphone (on Android and iPhone). I assume this is to mask her true cell phone number/identity. Can FreeSwitch route incoming SMS messages on DIDs to specific extensions/registrations? And route from an extension out? Any documentation hints would be greatly appreciated. Thanks, Sean

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Transcoding Bandwidth parameters
by Elliot Journeaux 12 Aug '19

12 Aug '19

Hi, I've noticed when transcoding freeswitch sets a default max bandwidth to 1mb (b=AS:1024). Why is this? And is this configurable? Thanks, Elliot Elliot Journeaux UC Engineer T 0330 058 0000 D 01202 592311 E Elliot.Journeaux(a)hihi.co.uk [http://www.4com.co.uk/signature/HiHi.jpg]<http://hihi.co.uk/> HiHi.co.uk<http://hihi.co.uk> HiHi Ltd One Lansdowne Plaza, 24 Christchurch Road, Bournemouth, Dorset BH1 3NE This message contains confidential information and is intended only for the individual named. Our company accepts no liability for the content of this email, or for the consequences of any actions taken on the basis of the information provided. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the company. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. If you have received this email in error please notify the system manager or the sender immediately and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message or any damage caused by any virus, which arise as a result of e-mail transmission. The recipient should check this email and any attachments for the presence of viruses.

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Changing a client's local streams (devices) in Verto client
by Anton Olofsson 12 Aug '19

12 Aug '19

Does Verto allow for clients' streams to be changed during a call? I'm trying to add the ability for users to change their microphone whilst in a call. When looking at the source code for the Verto client, I see that streams are only read when starting a call. Would it be possible to add additional handling of adding and removing streams to the client, or is there a limitation in mod_verto which restricts this? Thanks, Anton Olofsson

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Unable to start the Freeswitch server
by Veeru D 11 Aug '19

11 Aug '19

Dear Team, While starting Free Switch server getting below error. Not able to find the solution. Please help me to fix the same. 2019-08-05 16:20:23.012205 [ERR] sofia.c:3228 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC) 2019-08-05 16:20:23.012205 [ERR] sofia.c:3238 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system. 2019-08-05 16:20:23.012205 [ERR] sofia.c:3228 Error Creating SIP UA for profile: internal (sip:mod_sofia@183.82.114.188:5060;maddr=10.30.2.118;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC) 2019-08-05 16:20:23.012205 [ERR] sofia.c:3238 Error Creating SIP UA for profile: internal (sip:mod_sofia@183.82.114.188:5060;maddr=10.30.2.118;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system. Sent from Mail for Windows 10

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Re: [Freeswitch-users] Chat Plan Handling
by Sean Devoy 10 Aug '19

10 Aug '19

Thanks Keith, that is helpful if we need to cancel a message. What I ended up with is NOT cancelling the event but modifying it to have the updated information it needs to route correctly. For inbound SMS messages, I just set the "to" value to a registered extension and let the plan exit. The default handling is to then send it. For external (out to my VSP) I had to set several "fields": [cid:image001.png@01D54F76.816FCA20] Hope that helps others, Sean From: k(a)rhizomatica.org <k(a)rhizomatica.org> Sent: Saturday, August 10, 2019 7:56 AM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>; Sean Devoy <sdevoy(a)bizfocused.com> Subject: Re: [Freeswitch-users] Chat Plan Handling On 08/08/2019 16:23, Sean Devoy wrote: > > How can to stop the default handling of an inbound SMS message? > Hi Sean, after some long time trying to do that, then giving up, and coming back to it, various times, I finally went for the source code with a comb and it seems I finally found a way to do it: Please see. https://github.com/Rhizomatica/rccn/commit/db08b977711da1e09e3326db1723f677… Hmm, when making that commit, I referenced: https://docs.freeswitch.org/switch__loadable__module_8c.html#a6389148319476… Which is where I found in this comment: /* The event was handled by an extension in the chatplan, * so the event will be duplicated, modified and queued again, * but it won't be processed by the chatplan again. * So this copy of the event can be destroyed by the caller. */ This is the condition for skipping the default handling: if (switch_true<https://docs.freeswitch.org/switch__utils_8h.html#a890e4243bd6e31e6c68a8383…>(switch_event_get_header<https://docs.freeswitch.org/group__events.html#ga5c3108ab2e8d73d8ef8e6c4122…>(message_event, "final_delivery"))) { do_skip = 1; } so you need the final_delivery event header. Not sure how you add that in an XML dialplan, or in your implementation. Maybe the python in github can give you some clues. and no, stop and all that does not work - It only prevents the chatplan from execing applications. (IIUC) Cheers Keith.

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Chat Plan Handling
by Sean Devoy 10 Aug '19

10 Aug '19

Hi, How can to stop the default handling of an inbound SMS message? I am working on a chat plan strategy. I need to route external (DID) SMS messages to a registered extension. I think I do that with <action application="chat" data="sip |... However, I can't stop the original SMS message from getting an error for no registered user. I have added <action application="stop" data=""/> but the default route is still attempted. The error causes no problems, but it may cause me to miss actual errors when they occur. Thanks, Sean

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SMS Nobody to send to: Profile internal
by Sean Devoy 08 Aug '19

08 Aug '19

Hi, I have a multitenant server that I am trying to get SMS messaging working on. When I send a text "xxx" from one extension to another I get: 2019-08-01 18:02:56.952248 [WARNING] sofia_presence.c:221 Not sending to local box for 666@[fqdn here] 2019-08-01 18:02:56.952248 [ERR] sofia_presence.c:272 Chat proto [sip] from ["667" <sip:667@[fqdn here]>;tag=72173e0a] to [666@[fqdn here]] xxx Nobody to send to: Profile internal Where do I define the profile? What goes in it? The FQDN or context maybe? Thanks, Sean

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Recording Variables in CDR
by Ankit 06 Aug '19

06 Aug '19

Not getting variable record_file_size for channel. I have enabled the user_record filed in extensions configuration. Also what checks are applied for recording_file and record_ms channel variables. -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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CDR Record variables
by ankit jobanputra 06 Aug '19

06 Aug '19

Hello, Getting the record variables inconsistently for freeswitch CDR. recording_file, record_ms and record_file_size. Is there any configuration for setting this up. I have enabled the user_record variable in extension configuration. Not getting record_file_size for normal user-user call.

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