August 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Does this configuration make sense?
by Michael Avers 16 Aug '17

16 Aug '17

I'm trying to ditch multiple registrations and go with a concept where a user (say identified internally as extension # 3005) can register any number of devices/softphones using unique SIP credentials for each one, and so extension would end up having these extensions configured: 3005@domain - primary extension, but no one registers it directly. It is used for presence and its dial_string will define what other extensions are supposed to ring 3005_desktop@domain 3005_mobile1@domain 3005_yealink@domain and so 3005's dial_string can define that the Yealink and Mobile1 extensions will ring for the first 10 seconds, and only then 3005_desktop starts to ring as well. They will all have 3005's VM box as their voicemail box. When they call out they will set presence_id to be 3005@domain and also presence_id will be set when they are called via 3005's dial_string. Anybody ever did this kind of set up? The whole idea is to have the user control which of their devices ring, if at all. Because I see more and more users registering on 3+ devices and they don't always want them to all to ring right away. Mike

6 8

Empty Re-Invite after a=inactive
by Andrew Cassidy 15 Aug '17

15 Aug '17

Hi Guys, Just wanted to check a behaviour with you guys before rushing to upgrade FreeSWITCH from 1.4.26-37 We are reciveing a reinvite with a=inactive followed by an reinvite with no SDP. I am being told the one with no SDP should revert the media the the previously established. FreeSWITCH isn't currently doing that, (or passing the empty one back to the endpoint) it's just keeping the call held. Which is the correct behaviour? Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director 03303 880 960 andrew(a)cassidyweb.co.uk <andrew(a)cassidywebservices.co.uk> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>

5 11

Channel variables on a-leg and b-leg
by Paul Mateer 15 Aug '17

15 Aug '17

Sorry. I've had to switch to a different mail account because each time I try to reply I get a "you must be a member to post to this list" message, even though I'm already subscribed. Anyway, I don't have an issue with the use of the a-leg uuid. I think my problem stems from an assumption that the ids would be the same on the client and the server. I have a client UA which is placing a call through the server, which in turn is recording the conversation. The client needs a unique way of identifying the filename used in the recording and the use of the uuid seemed like a suitable way of identifying the file. However I've realised that the call Id returned as a response to the pa call issued is a client-side id and doesn't correlate with the call id used at the server end of things. Is there any way for the client to know the call used by the server, or is there a way for the client to provide a custom id that will be accessible to the server when processing the dialplan? > The answer is all of these uuid’s are being used by the server, 1 for a leg, and one for each b leg. Why do you have an issue with using the a leg uuid? >* On Aug 14, 2017, at 1:08 PM, Paul Mateer <paul.mateer at outlook.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: *> >* Thanks Vladyslav. *> >* After reading your reply I found relevant information under the originating_leg_uuid section of the Channel Variables confluence page. *> >* I tried modifying the Dialplan entry on the server to include both ${uuid} and $(call_uuid} in the name of the session recording file as follows *> >* <extension name="Test Dialplan entry"> *>* <condition field="destination_number" expression="^(4008)$"> *>* <action application="export" data="execute_on_answer=record_session $${base_dir}/recordings/${strftime(%Y.%m.%d_%H.%M.%S)}_${uuid}_.${call_uuid}._${caller_id_number}.wav"/> *>* <action application="export" data="dialed_extension=$1"/> *>* <action application="set" data="call_timeout=30"/> *>* <action application="set" data="hangup_after_bridge=true"/> *>* <action application="set" data="continue_on_fail=true"/> *>* <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> *>* <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> *>* <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> *>* <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> *>* <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> *>* <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> *>* <action application="bridge" data="user/$1@${domain_name}"/> *>* <action application="answer"/> *>* </condition> *>* </extension> *> >* but the name generated had the same Guid for both uuid and call_uuid (which suggests that it's the a-leg). *> > >* My client issues a call to a number using embedded FreeSWITCH as follows: *> >* FreeSWITCH.Native.Api.Execute("pa", "call sofia/gateway/<gateway>/<number> XML <caller Id> <caller name>") *> >* and gets back a response like *> >* SUCCESS:<call number>:<Guid> *> >* The Guid in the response looks like it should be a call Id because it'd the same Guid that appears in the Unique-ID headers of the FreeSWITCH events that the client receives - it just doesn't seem to match the Guid(s) used by the server when it'd generating the recording name. Is there any way for the client FreeSWITCH to know or gain access to the IDs being used by the FreeSWITCH server? *> >* Paul *> >* From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> on behalf of Vladyslav Zakhozhai <v.zakhozhai at gmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* Sent: 14 August 2017 16:41:13 *>* To: FreeSWITCH Users Help *>* Subject: Re: [Freeswitch-users] Channel variables on a-leg and b-leg *> >* Hi Paul, *> >* ${uuid} - is Unique Id of channel. For a-leg and b-leg it will be different (I mean that uuid is not Call-ID). You can try to use call_uuid instead. But I'm not sure about that. *> >* I think that you can use set instead of export and fetch recording data from a-leg. I'm using such approach and do not experience any problems with that. *> >* On Mon, Aug 14, 2017 at 5:01 PM, Paul Mateer <Paul.Mateer at outlook.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <mailto:Paul.Mateer at outlook.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>> wrote: *>* Guys, I'm using the C# interface to FreeSWITCH to add the ability to make and receive calls in an existing software application. *> >* I want to record the conversations on the server, but I need to be able to access the recording after the call is complete. *>* The Call-ID value seemed the perfect solution to this and I added the following action to my dialplan entry. *> >* <action application="export" data="execute_on_answer=record_session $${base_dir}/recordings/${strftime(%Y.%m.%d_%H.%M.%S)}_${uuid}_${caller_id_number}.wav"/> *> >* This seemed to work fine, as I would get recording files created with a Guid in the name, but on closer inspection I found that the Guid used in the filename was not the same Guid associated with the successfully placed call in the application. I assume that this is because the application sees the Call-ID for the a-leg, whereas the dialplan entry is executed in the b-leg and therefore has it's own separate Guid. *> >* Is there any way in which I can expose the Call-ID (or some other custom channel variable) for the a-leg to the b-leg so that it will be accessible within the XML dialplan and therefore be usable when creating the recording file? *> >* Thanks for any suggestions/assistance offered. *> > > >* _________________________________________________________________________ *>* Professional FreeSWITCH Consulting Services: *>* consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <mailto:consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> <http://www.freeswitchsolutions.com/ <http://www.freeswitchsolutions.com/>> *> >* Official FreeSWITCH Sites *>* http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>> *>* http://confluence.freeswitch.org <http://confluence.freeswitch.org/> <http://confluence.freeswitch.org/ <http://confluence.freeswitch.org/>> *>* http://www.cluecon.com <http://www.cluecon.com/> <http://www.cluecon.com/ <http://www.cluecon.com/>> *> >* FreeSWITCH-users mailing list *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <mailto:FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> <http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>> *>* http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>> *> > > >* -- *>* С уважением, *>* Владислав Захожай *> >* _________________________________________________________________________ *>* Professional FreeSWITCH Consulting Services: *>* consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> *> >* Official FreeSWITCH Sites *>* http://www.freeswitch.org <http://www.freeswitch.org/> *>* http://confluence.freeswitch.org <http://confluence.freeswitch.org/> *>* http://www.cluecon.com <http://www.cluecon.com/> *> >* FreeSWITCH-users mailing list *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> *>* http://www.freeswitch.org <http://www.freeswitch.org/> *

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rad_auth failing under load
by Joseph Waite 15 Aug '17

15 Aug '17

Hi Guys I am having a strange issue, when I have more than about 450 concurrent calls, rad_auth seems to fail to authorise calls. I see the Radius request received and replied with RAD-ACCEPT from the Radius server but I get the following error 2017-08-07 18:29:53.202133 [ERR] mod_rad_auth.c:504 RADIUS Authentication failure (RC=1) 2017-08-07 18:29:53.202133 [ERR] mod_rad_auth.c:702 An error occured during radius authorization. I have had a look as ulimit and seems to be fine. core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 128716 max locked memory (kbytes, -l) 64 max memory size (kbytes, -m) unlimited open files (-n) 65536 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) 128716 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited Anyone have any ideas? Regards

3 11

Answer a phone remotely via API
by Peter Steinbach 15 Aug '17

15 Aug '17

Hello, does there exist a command to answer a phone remotely? E.g. send a command via ESL to Freeswitch and then to the Phone? I am not looking for some kind of autoanswer. I just want to let the phone ring, then send some kind of event to the phone and then the phone answers. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

2 2

macOS urgent note for 24x7 folks on Sierra +
by Mario G 14 Aug '17

14 Aug '17

If you have FreeSWITCH running 24x7 on Sierra (and eventually High Sierra), you must update the plist in LaunchAgents to the updated one in the wiki: https://freeswitch.org/confluence/display/FREESWITCH/macOS+24x7+Preparation… <https://freeswitch.org/confluence/display/FREESWITCH/macOS+24x7+Preparation…> I spent months on this with Apple and instead of going into the very long story the short of it is: There was a design change (bug in my opinion) in 10.12.0, 10.12.1, 10.12.2, 10.12.5 and 10.12.6 (notice .3 and .4 were OK). Using the default process type macOS would not give CPU to FreeSWITCH even if nothing else was running. I tested this with other apps and it was not a problem, I could only make this happen with FreeSWITCH. I eventually found that overriding the process type resolved the issue in Sierra and High Sierra. Mario G

1 0

ClueCon Talk: Open Source WebRTC testing suite, slides and resources
by Chad Phillips 14 Aug '17

14 Aug '17

I’ve had a few people email me privately about the content of my talk, so I figured I’d post a resource link: http://tiny.cc/cc-2017-webrtc-testing That contains the slides from the presentation, and a resource section with links to relevant projects/demos. Chad

3 2

Multiple Registrations or unique SIP credentials for each user device?
by Michael Avers 14 Aug '17

14 Aug '17

Hello, We have customers that currently register their single extension on their desk phone, mobile softphone, as well as desktop softphone - and usually they want them all to ring at the same time, which is easy, we just let them have multiple registrations and it gets the job done. However, some customers are asking to have more fine grained control over which of their devices receive the incoming calls and in what order. For example they may want the desk phone to ring or 10 seconds and only then start ringing the mobile app, and the desktop softphone to never ring (they just use it for outbound dialing). What is the best way to achieve this? I thought about assigning unique credentials to each device they register on. This would be fine if I were to direct inbound calls coming on a DID or via an IVR extension, because I can just ring them as if it was a ring group and control all the needed requirements with the bridge string. However, how do I get this to work for local extension to extension calling? Say their extension should be 3005, but they no longer register it directly and instead are issued credentials for 3005_DESK@domain, 3005_XYZ@domain and so forth. I can have 3005 be a dialplan extension that bridges to them, sure. But then you have presence to deal with, basically when 3005_DESK is on the phone 3005@domain presence should indicate that. Is what I'm asking for possible at all, and if so are there any best practices to get this working properly? Hope I was clear enough :) Thanks Mike

2 4

freeswitch 1.6.19 bizarre fax 488 behavior
by Aqs Younas 13 Aug '17

13 Aug '17

I am facing a bizarre behavior while trying to send fax from zoiper to freeswitch. First, i tried to use the dialplan example as shown in documentation page. <extension name="fax_receive"> <condition field="destination_number" expression="^9978$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/> <action application="hangup"/> </condition> </extension> But this resulted in "488 Not Acceptable Here" even though i have set fax_enable_t38 in vars.xml Logs. *https://pastebin.freeswitch.org/view/07224882 <https://pastebin.freeswitch.org/view/07224882>* Second, But i remove few lines from dialplan. <extension name="fax_receive"> <condition field="destination_number" expression="^9978$"> <action application="answer" /> <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/> <action application="hangup"/> </condition> </extension> Everything seems to work perfectly. Third. When i insert a small lua script after answer in above dialplan it again fails with 488. <extension name="fax_receive"> <condition field="destination_number" expression="^9978$"> <action application="answer" /> <action application="lua" data="jack.lua"/> <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/> <action application="hangup"/> </condition> </extension> I just want to run a small lua script before i call rxfax application. Any help would be much appreciated. Best Regards. <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

External forwarding and CDRs - which method to use?
by NYCPhoneService 13 Aug '17

13 Aug '17

The call flow is: External Number -> internal extension -> Forwarding to external number. When I use loopback channel, the outbound CDR leg is generated with 2 second call as it is known. I can do a post call processing in database and Check if is_loopback=1, match inbound leg by bridge_uuid and calculate the total call length. I'm trying to avoid post call processing. For that, I tried using sip_looped_call in public context. Works well if you change the deflect data to ${sip_to_uri} from the default ${destination_number}. However, all custom variables are lost and first leg is written with BLIND_TRANSFER hangup cause. Billsec is 0 which make sense. If I need to bill on inbound leg, what should I do in this case? Any suggestions are appreciated.

1 0