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Hangup Cause using ignore_early_media=false w183/SDP
by Gonzalo Gasca Meza 20 Aug '16

20 Aug '16

Hi Forum, I'm launching calls using ESL originate command. Example: *originate {ignore_early_media=false,bridge_early_media=false,originate_timeout=24,call_timeout=24,execute_on_answer_1='sched_hangup 18 ALLOTTED_TIMEOUT',origination_uuid=920a1b14-j02y-7bi9-kgq0-3c1u-920a291166,origination_caller_id_number=+55XXXXXXXXX,RECORD_STEREO=true,execute_on_answer_2='record_session /var/www/html/recordings/Y90I68RLWX_+55XXXXXXXXX_920a1b14-j02y-7bi9-kgq0-3c1u-920a291166.wav'}sofia/gateway/twilio/+55XXXXXXXXX handle_calls* I disabled ignore_early_media from true to false. What is happening is that when I receive a 183 w/SDP and call is NOT ANSWERED I used to get NO_ANSWER as reason code, now I get NORMAL_CLEARING even though call was never connected. My feeling is that after entering in tone detection and bridge the call is just following rules and end up in hangup rule which by default should send this reason. handle_calls - Dialplan: sofia/external/+XXXXXXXXXX Action answer() - Dialplan: sofia/external/+XXXXXXXXXX Action log(INFO handle_calls application match) - Dialplan: sofia/external/+XXXXXXXXXX Action set(fax_verbose=true) - Dialplan: sofia/external/+XXXXXXXXXX Action set(transfer_ringback=${us-ring}) - Dialplan: sofia/external/+XXXXXXXXXX Action spandsp_start_fax_detect(transfer 'fax' 10 ced) - Dialplan: sofia/external/+XXXXXXXXXX Action spandsp_start_tone_detect(55) - Dialplan: sofia/external/+XXXXXXXXXX Action playback(silence_stream://20000) - Dialplan: sofia/external/+XXXXXXXXXX Action spandsp_stop_tone_detect() - Dialplan: sofia/external/+XXXXXXXXXX Action hangup() I know that during 183 w/SDP many things could happen (BUSY, REORDER, Announcement, RING BACK, etc). What is the recommended way to handle those situations? For now Im using the following conditions to rewrite the reason code in my App: - call_progress_183sdp + ring_tone + no_200OK = NO_ANSWER - call_progress_183sdp + busy|reorder|sit tone + no_200OK = BUSY,INVALID_DESTINATION - call_progress_183sdp + no_tone + no_200OK = INVALID_NUMBER (Probably an announcement) Any pointers in how to make Freeswitch handle the above conditions internally? Thanks

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Conference API
by Mirko Brankovic 20 Aug '16

20 Aug '16

Hi, I'm trying to set up verto vhost to execute lua scrypt on API calls. api = freeswitch.API() That is all working when I use uuid_dump or all uuid commands like this: ret = api:execute("uuid_dump", uuid) ret = api:execute("show", "channels") But if I try to execute conference APIs like: ret = api:execute("conference list") i get invalid command response (INVALID COMMAND!) Is there some security issue where I need to allow verto to access Conference API, or am I doing something wrong with this? -- Regards, Mirko

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freeswitch 1.6 without video support
by Grant Bagdasarian 19 Aug '16

19 Aug '16

Hello, Is it possible to compile and install fs 1.6 without support for video? Or is the 1.4 version the same as 1.6 with bug fixes and security updates? Regards, Grant

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Two FS balancing/failover using DNS
by Yu Boot 19 Aug '16

19 Aug '16

I have an idea to make two FS servers with identical configs and with same host name on DNS. What's wrong with this? Is it working solution? Sent from my ASUS

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Invite request randomly landed to wrong context
by acheraime . 19 Aug '16

19 Aug '16

Hi, I'm experiencing an issue where for some reason, internal authenticated calls land to the public context. This is very random and the next attempt always go to the right context. Any idea of what might be the root cause? Thanks and Happy Friday! -- *Adolphe CHER-AIME*

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Configure freeswitch and opensips for using tls and udp protocols simultaneously.
by Стас Тельнов 19 Aug '16

19 Aug '16

I have freeswitch and opensips working with the mobile client in the conference mode. When using UDP connection everything works perfectly, but when using tls connection the call is interrupted in 30 seconds. Whether to use TLS or UDP connection - it is assigned on the mobile client before initialization of connection with opensips server. Originally I assumed that these problems were caused by the NAT settings, but in that case the problem would be watched irrespective of the connection used - UDP or TLS. Generally such scheme works as it should: +++++++++ udp ++++++++ udp +++++++++ udp +++++++++ + + -----> + + -----> + + -----> + + + phone + + SIP + + free + + SIP + + + <----- + + <----- + switch + <----- + provider + +++++++++ udp ++++++++ udp +++++++++ udp +++++++++ And in such scheme a call breaks in 30 seconds: +++++++++ tls +++++++++ udp +++++++++ udp +++++++++ + + -----> + + -----> + + -----> + + + phone + + SIP + + free + + SIP + + + <----- + + <----- + switch + <----- + provider + +++++++++ tls +++++++++ udp +++++++++ udp +++++++++ SIP and freeswitch are in one local area network (Amazon EC2). SIP provider doesn't support tls in principle, they have 5061 closed. And the BYE packet sends freeswitch, as I understand, from packet headers as I didn't receive the response to ACK in time. There is the packet: BYE sip:8@85.236.*.*:55194;ob;transport=tls SIP/2.0 Via: SIP/2.0/TLS sip0.*.*:5061;branch=z9hG4bKc7a2.7909e7e1.0; received=52.58.*.* Via: SIP/2.0/UDP 172.31.*.*;received=52.58.*.*;rport=5060;branch= z9hG4bKBK82Zg50c2U0p Max-Forwards: 69 Contact: <sip:*7906******@52.58.*.*:5060;transport=udp> To: "8" <sip:8@sip0.*.*>;tag=59221e6a From: <sip:*7906******@sip0.*.*>;tag=j4aX21rv83etN Call-ID: O7E3ktwLPiQWDN2Rism-7g.. CSeq: 95383912 BYE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit Reason: SIP;cause=408;text="ACK Timeout" Content-Length: 0 Having looked on logs, I can tell that the INVITE packet from the mobile client reach freeswitch and provider, but in reverse Trying/Ringing packet doesn't reach. I can't understand at what stage there is a problem. Freeswitch can't respond and transmit the response through opensips, or there is a problem in something else? Who faced similar problem, prompt what settings should be analyzed in order that the above-stated scheme with tls connection start functionning?

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SMS Gateways / Providers
by Don Hawkins 19 Aug '16

19 Aug '16

I'm looking for SMS providers that will allow me to set outgoing number to any one of my DID numbers. I know there are restrictions on this in the US and Canada but others such as Twilio and Plivo have obviously have a provider. Will I have to buy the DID number from the SMS provider or l can I use my own DID numbers from my current provider? Suggestions please... Sincerely, Don Hawkins

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OpenR2/ftmod_r2 Installation on Sangoma/Digium
by Nandy Dagondon 19 Aug '16

19 Aug '16

Hi folks, I'm installing OpenR2/FreeTDM on Sangoma A102. It compiled/installed successfully using Wanpipe drivers. However, I'm expecting ftmod_r2 module to be installed. But, there's none. Is this okay? Meaning the R2 signalling is included in the ftmod_wanpipe module? I'm also installing OpenR2 for a Digium card. I encounter FS "make" errors. The card is not yet installed. Does the card need to be installed to make/install FS modules? Do I need to uninstall the Wanpipe drivers to avoid conflict with the DAHDI drivers? Appreciate for you inputs. Tks, Nandy

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Username with @ in it
by Igor Olhovskiy 18 Aug '16

18 Aug '16

Hi! My users want to use usernames in their phones e-mail-type style. Like username@domain. With FS it works ok. The only problem I’ve faced - it’s when you route this call via proxy like OpenSIPS or Kamailio. Actual problem, when you bridge user@proxy, your URI becomes like username@domain@proxy. So, This RURI is dropped by proxy with reason «cannot be parsed» or so. Actually, they are right, Is there any way of avoid to use two @ in URI? FS 1.6.9 -- Best regards, Igor

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Is there equivalent of "last_bridge_to" but showing "last_originated_leg"
by Victor Chukalovskiy 18 Aug '16

18 Aug '16

Good day, My ultimate goal is for any given a-leg CDR to find the very last b-leg CDR. var "last_bridge_to" allows to find last b-leg for any given a-leg as long as call was actually "bridged". However, if given a-leg had b-leg originated, but no actual media or ring-back happened, variable "last_bridge_to" remains empty. So I can't use that. So is there a single variable on a-leg CDR that contains UUID of the last "originated" leg? I know there are arrays e.g. "originated_legs" or "originate_causes", so worst case scenario I can iterate and pick the last UUID in either one of these arrays. But this means external code or fancy trigger functions upon CDR SQL insert. Trying to avoid that. Many thanks! -Victor

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