July 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Is there a way to do a (signalling only) call timeout on a bridge request?
by Nathan Neulinger 09 Jul '19

09 Jul '19

Is there a way to put a timeout on a bridge that would detect the situation of (media or no media)+(no response to invite) ? It seems like originate_timeout and leg_timeout do not catch this case. I'm wanting to detect failed/non-responsive outbound call, but NOT die on 'taking a long time to answer' which is what it appears that call_timeout will do. The failure scenario that occurred was a bad firewall ruleset resulting in rtp flowing for an outbound call, but the response to the invite never made it back to FS (responding SIP message was dropped at fw, but not the RTP). I'd like to be able to time out the call request, but NOT time out if I'm actually getting an actively ringing call. Suggestions? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 341-6679 System Administrator - Architect (573) 612-1412 System and Desktop Infrastructure Team Manager

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How to get call uuid from b-leg and insert to pgsql
by 王聡 09 Jul '19

09 Jul '19

Hi all, I’m trying to manage cdr in pgsql by mod_cdr_pg_csv, and I hope to record a call uuid to manage multiple recorded channels in one call. A channel variable “variable_call_uuid” seems good to record, and I confirmed it is the correct call uuid in fs_cli by: freeswitch@TEST-FS> uuid_getvar 67534e60-fc91-4e6b-b64e-868d5f3d8b16 call_uuid 67534e60-fc91-4e6b-b64e-868d5f3d8b16 freeswitch@TEST-FS> uuid_getvar 2ab39a17-2f8d-4ac1-81de-b32a990e55c1 call_uuid 67534e60-fc91-4e6b-b64e-868d5f3d8b16 However, when I tried to record this variable into pgsql, it turned into channels uuid in pgsql. The followings had added into cdr_pg_csv.conf.xml: <field var="call_uuid”/> But pgsql record showed: SELECT call_uuid FROM cdr where uuid = '67534e60-fc91-4e6b-b64e-868d5f3d8b16’; > 67534e60-fc91-4e6b-b64e-868d5f3d8b16 SELECT call_uuid FROM cdr where uuid = '2ab39a17-2f8d-4ac1-81de-b32a990e55c1’; > 2ab39a17-2f8d-4ac1-81de-b32a990e55c1 # Which should be 67534e60-fc91-4e6b-b64e-868d5f3d8b16 Any idea or advice on this issue? Regards.

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Custom Sip "To:" header
by Mehdi Shirazi 06 Jul '19

06 Jul '19

Hi I want to set a phone number different than phone number in INVITE request URI in the "To:" header.I tried this dialplan without success: <action application="export" data="sip_to_user=777777"/> <action application="bridge" data="sofia/internal/091XXXXXXXX@10. !XX.1XX.XXX"/> Suggestion please. Regards M.Shirazi

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fax_enable_t38_insist
by Sandro Bordacchini 06 Jul '19

06 Jul '19

Hello everyone. Can anyone point me to the documentation about the variable in subject? Even a brief hint on how and when to use it (and what it is the difference with the other fax_enable_t38* variables) would be greatly appreciated. Thanks a lot in advance. -- SB

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Call center dtmf binding
by Vishal Pai 05 Jul '19

05 Jul '19

Hello Everyone I have one scenario where call comes to call center agent and agent answers the call after that the agent need to dial external number using dtmf in ongoing call. I tried with certain parameter under contact for call center module to send the dtmf but it didn't work. what is the best possible way to achieve this? Thanks Vishal Pai

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Problem with SRTP in HA
by Francisco Carmona Ramírez 05 Jul '19

05 Jul '19

Hi, i have some issues trying to make a HA with SRTP, i have that dialplan <extension name="salientes"> <condition field="destination_number" expression="^(\d{9})$"> <action application="set" data="ringback=${es-ring}"/> <action application="set" data="transfer_rinback=$${hold_music}"/> <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="export" data="rtp_secure_media=optional"/> <action application="export" data="rtp_secure_media_outbound=forbidden"/> <action application="bridge" data="sofia/gateway/providerExt/$1"/> <action application="hangup"/> </condition> </extension> When i make a normal call in my FS1 it goes well, same when i do a normal call in a FS2, then the problem that i find is when i do a normal call in my FS1 (it takes AES_CM_128_HMAC_SHA1_32) and i do a "fsctl crash" in my "fs_cli", when the FS2 recovery the call, the audio drops in both ways, and if I use sngrep to see what is sending in the invite, i see that the crypto is "AES_CM_NULL_AUTH", in addition in the fs_cli on FS2 y found that error 2019-07-04 10:37:38.286877 [ERR] switch_core_media.c:1554 Error! No crypto to parse 2019-07-04 10:37:38.286877 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND 2019-07-04 10:37:38.286877 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_32 2019-07-04 10:37:38.286877 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV 2019-07-04 10:37:38.286877 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_32 Can someone give me some clarity about that problem? How I can solve this?

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FreeSWITCH Weekly Community Conference Call
by Ken Rice 03 Jul '19

03 Jul '19

Hey FreeSWITCHers. Join us today at 12noon US Central Time at https://conference.freeswitch.org/vc and call 888 or via sip:888@conference.freeswitch.org Open Floor discussion of FreeSWITCH and related technologies. This week we'll be talking about Debian 10's pending release! Plus more. Need help with FreeSWITCH? Come on in and ask your question. K

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Get videocall image preview in target endpoint
by Jose David Jurado Alonso 03 Jul '19

03 Jul '19

Hi, I would like to know if there is any way to obtain a preview image of the video intercom (source endpoint) that makes the call to be shown on the mobile (target endpoint). In this way, the target of the call can see who is calling before answering. intercom (with camera) <--> FS <--> mobile (linphone) Thanks. José David Jurado Alonso

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