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update callee_id_name on pickup (using intercept)
by Piccinin Francesco 09 Feb '17

09 Feb '17

Hi all, I'm experiencing the following issue on pickup groups. We would like to update callee_id_name on pickup groups (using intercept). Please consider the following example: Extensions A and B are in the same pickup groups. Extension C is calling extension B. Extension A perform a pickup using the code *8 (please find below the instersting part of our dialplan.xml). On extension A display we see extension B callee_id_name and callee_id_number. We expect to see extension C number and name. Which channel variable to we need to update? Find below an export of our dialplan.xml Thanks. regards  <condition field="destination_number" expression="^\*8$" break="on-true"> <action application="answer"/> <action application="set" data="intercept_unbridged_only=true"/> <action application="intercept" data="${db(select/$${hostname}_last_dial_${domain_name}/${pickup_group})}"/> <action application="lua" data="functions/clean_limit_pickup.lua ${pickup_group} ${domain_name}" inline="true"/> <action application="sleep" data="2000"/> </condition> Francesco Piccinin Innovative Services Telecommunications Insiel S.p.A. via San Francesco d'Assisi, 43 34133 - Trieste (TS) Italia tel +39 040 3737752 mob +39 335 1329440 francesco.piccinin(a)insiel.it<mailto:francesco.piccinin@insiel.it> www.insiel.it<http://www.insiel.it> rispetta l'ambiente: se non ti è necessario non stampare questa mail AVVISO DI RISERVATEZZA Informazioni riservate possono essere contenute nel messaggio o nei suoi allegati. Se non siete i destinatari indicati nel messaggio, o responsabili per la sua consegna alla persona, o se avete ricevuto il messaggio per errore, siete pregati di non trascriverlo, copiarlo o inviarlo ad alcuno. In tal caso vi invitiamo a cancellare il messaggio ed i suoi allegati. Grazie. CONFIDENTIALITY NOTICE Confidential information may be contained in this message or in its attachments. If you are not the addressee indicated in this message, or responsible for message delivering to that person, or if you have received this message in error, you may not transcribe, copy or deliver this message to anyone. In that case, you should delete this message and its attachments. Thank you.

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freeswitch responding to other port for register request and fail to register
by devang nathwani 09 Feb '17

09 Feb '17

Hello, Freeswitch is getting registration request from ip:random_port and freeswitch responding to ip:5060 and that is why As shown in image 91.18.19.90:50319 is sending register request to freeswitch server 38.201.12.30 but freeswitch is responding to 91.18.19.90:5060 and device is not able to register, what is wrong here? [image: Inline image 1]

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3-party call where 2 people should not hear each other
by Oivvio Polite 09 Feb '17

09 Feb '17

I'm looking at a problem for a potential client. They want to do three party PSTN calls where all audio is NOT routed to all parties. Two parties (A and B) are in the same room but are using separate handsets rather then a speaker phone. The third party (C) is at a remote location. C should hear audio from A and B A and B should both hear audio from C but not from one another. Since they are both in the same room hearing the audio from one another in the handset as well is just distracting. My client has previously experimented with Twilio and figured out that their API does provide a solution.

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Problems with privacy settings on invite
by Christoph Russow 09 Feb '17

09 Feb '17

Hi All, i recently got informed by a collegue that it seems im not setting privacy stuff correct on our outbound calls. After investigating the mentioned call i see the following invite on the inbound leg (i xxxxxed out the last 5 digits of every phone number): INVITE sip:4980058xxxxx@88.198.223.120:5060 SIP/2.0 Via: SIP/2.0/UDP 217.110.38.229:5060;branch=z9hG4bK0dB7a5469c2220f98ee From: "Anonymous" <sip:Anonymous@Anonymous.invalid>;tag=gK0d356f75 To: <sip:4980058xxxxx@88.198.223.120> Call-ID: 1863167222_96798667(a)217.110.38.229 CSeq: 21390 INVITE Max-Forwards: 26 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: "Anonymous" <sip:Anonymous@217.110.38.229:5060> P-Asserted-Identity: <sip:499132xxxxx@217.110.38.229:5060> Privacy: id Supported: timer,100rel Session-Expires: 1800 Min-SE: 90 Content-Length: 262 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 29264 8032 IN IP4 217.110.38.229 s=SIP Media Capabilities c=IN IP4 217.110.38.228 t=0 0 m=audio 19174 RTP/AVP 18 8 100 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=sendrecv a=ptime:20 and on the Outbound Leg i see freeswitch sending the following invite to our outbound carrier: INVITE sip:49151708xxxxx@80.84.31.98 SIP/2.0 Via: SIP/2.0/UDP 88.198.223.120;rport;branch=z9hG4bKFyctt2ZcKvBaD Max-Forwards: 25 From: "Anonymous" <sip:499132xxxxx@88.198.223.120>;tag=eDe24j5Hp5cvm To: <sip:49151708xxxxx@80.84.31.98> Call-ID: 736df04f-64b0-1235-ad92-b8ac6f7db265 CSeq: 102726678 INVITE Contact: <sip:mod_sofia@88.198.223.120:5060> User-Agent: Emtex Voice Switch v1.9 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 251 X-FS-Support: update_display,send_info P-Asserted-Identity: "Anonymous" <sip:499132xxxxx@88.198.223.120> v=0 o=FreeSWITCH 1486105285 1486105286 IN IP4 88.198.223.120 s=FreeSWITCH c=IN IP4 88.198.223.120 t=0 0 m=audio 20200 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 my system is starting the outbound leg with the following command: bridge {origination_uuid=0043c244-7f46-4484-a4a1-fa8863f22321,sip_cid_type=pid,origination_privacy=hide_name:hide_number:screen}sofia/default/49151708xxxxx@80.84.31.98 The questions now are: Is there any way to prevent freeswitch from pulling the informations from the P-Asserted-Identity header into the From: header in the invite? Am i doing something wrong while setting the privacy options (any option that cancel each other out)? What is the preferred way of getting a fully anonymous invite (caller number nowhere mentioned) but correctly set P-Asserted-Identity with the caller number (required by german carriers by law) (like the inbound leg)? If you need any additional informations from me just ask. Best regards Christoph Russow -- ________________________________________________________________ EMTEX GmbH Christoph Russow Software Engineer Bischof-Otto-Weg 9 D-91086 Aurachtal Geschäftsführer: Markus Enzinger Tel. +49 9132 7490 0 Sitz der Gesellschaft: 91086 Aurachtal Fax. +49 9132 7490 900 Amtsgericht Fürth: HRB6804 ________________________________________________________________

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Simple question regarding bridge parameters
by Joel Serrano 08 Feb '17

08 Feb '17

Hi all, I have a simple question regarding {}, [] and bridge application when using multiple groups/failover. For example, saying the following bridge: <action application="bridge" data="sofia/outbound/${destination_number}@ $${SBC1}|sofia/outbound/${destination_number}@$${SBC2}"/> It will try SBC1, if it fails it will try SBC2. Fine till here. If I want to add {} and [] params, how would be the correct format: 1- Once, at the beginning, and it would apply in each destination (in this example 2, but it could apply to any number of destination we add with more | separators): <action application="bridge" data="{absolute_codec_string='PCMU,PCMA'}[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC1}|sofia/outbound/${destination_number}@$${SBC2}"/> 2- Once before each destination: <action application="bridge" data="{absolute_codec_string='PCMU,PCMA'}[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC1}|{absolute_codec_string='PCMU,PCMA'}[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC2}"/> 3- The {} only at the beginning, but the [] before each destination: <action application="bridge" data="{absolute_codec_string='PCMU,PCMA'}[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC1}|[park_after_bridge=true,park_timeout=7]sofia/outbound/${destination_number}@ $${SBC2}"/> Maybe they all work, I'm just not sure what is the correct way, or if there are any best practices on this matter. Thanks, Joel.

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disabling param enable-timer
by Ján Füri 08 Feb '17

08 Feb '17

Hi all, I have a question about disabling enable-timer. In my sip_profile i have <param name="enable-timer" value="true"/>. But one of our customer has a problem with it and his calls are ended after 90 sec. I think 90 sec. is time that customer's pbx is set for rtp timer. I don't want to change my sip profile enable-timer to false because it would influence all our customers. Is that possible to disable rtp timer in a dialplan ? BR, Jan Furi

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Allison prompt set for FreeSWITCH
by Brian West 08 Feb '17

08 Feb '17

FreeSWITCHers, Last week we finalized a few things we wanted to accomplish, One was we've been asked to make an Allison Smith prompt set for FreeSWITCH, She's agreed to do the entire sound set and now we have to raise the money to pay her for this. I would like to thank everyone that has donated to our dev week... We are very grateful our community got so involved. Here is the link to the GoFundMe for the Allison prompts (full sound set, including the zrtp words): https://www.gofundme.com/allison-prompts-for-freeswitch The total price for this was a little over $10,000, but we've had someone who shall remain nameless donate almost 50% of this total, Now we have to raise the rest. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest

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Module Loading Timeout Exit on Start-up
by Shane Mitchell 08 Feb '17

08 Feb '17

Hi all, Just a quick question before I start searching through the source. On FS start-up, there seems to be a 30 second timeout FS allows for modules to load, after which FS will exit. I appreciate that modules should take less than this to load, but I'm running into this timeout mainly during development, for various reasons, including when debugging the loading logic of the module (hence debugging has become a bit of a race!). Is there any way to disable or extend this module load timeout? Thanks in advance, Shane Mitchell

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FREESWITCH and Verto Communicator
by _dj 08 Feb '17

08 Feb '17

Hi, everybody! I'm new in FreeSwitch. I installed it via Quick Start FreeSWITCH Demo With Verto Communicator <https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie> . The only difference was that I generated letsencrypt certificate manually. I can call from Linphone to my freeswitch server, but I'm getting the "Waiting for server reconnection" error in 50-80% of times when accessing it via verto communicator from localhost. What's the problem? Thanks in advance. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FREESWITCH-and-Verto-Communic… Sent from the freeswitch-users mailing list archive at Nabble.com.

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uuid_getvar for multiple variables
by Ben Hood 08 Feb '17

08 Feb '17

Hi, I was wondering whether there is a multi-get command for channel variables? For example, something like uuid_getvar for multiple variables? Right now I'm using uuid_dump to get all variables and just filtering the ones that I need. But can I ask FS to give me a list of variables in one API invocation? Many thanks, Ben

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