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bridge and originate - difference between opening a channel and ring ready
by Paul Muaddib 30 Oct '18

30 Oct '18

Hi, I am trying to understand the difference between opening a channel and ring ready Following example: <action application="bridge" data="<originate_timeout=90,ignore_early_media=true>user/A, user/B, [leg_delay_start]user/C"/> When I call multiple endpoints with bridge in parallel and delay one of them with leg_delay_start, then I can see in the log files that freeswitch opens up three channels and delays the ring ready for C. But when at the same time user/C gets called from someone else while leg_delay_start is not over yet, can user/C still answer the second call? And which call reaches the user first if the user lets pass by leg_delay_start from the first call? When I do the whole thing with enterprise originate, it looks much more efficient since freeswitch is creating the channels only after the time for originate_delay_start has passed by. What should be used when? Best regards, Paul

2 1

Firefox WebRTC ICE error
by Social Boh 29 Oct '18

29 Oct '18

Hello list, I'm trying to use WebRTC on FreeSWITCH 1.8.2 with Firefox. While with FreeSWITCH 1.6.2 all works correctly with FreeSWITCH 1.8.2 I can't connect the call from browser to phone and viceverda. I think the most relevant lines are: [DEBUG] switch_core_media.c:4309 sofia/internal/1010(a)mydomain.org no suitable candidates found [ERR] mod_sofia.c:2493 CODEC NEGOTIATION ERROR. SDP: This problem is on 62.0.3 and 63 firefox versions. No problem with Chrome. Thank you Regards -- --- I'm SoCIaL, MayBe

1 1

filter * DTMF when using bind_meta_app and gateway that supports inband DTMF
by Bipin Patel 29 Oct '18

29 Oct '18

hi, i have a SIP gateway which doesn't support info or rfc DTMF so im forced with using inband so in my dialplan i use start_dtmf_generate to translate from info/RFC used by IP phones to inband so gateway can send that ahead, the problem i face is im using bind_meta_app to start a on demand recording using *2 so when the local extension presses *2, the * DTMF still goes to the gateway inband and 2 is never heard by callee. What i want is to filter out the complete *2 so the callee doesn't hear those at all but i dont want to drop all dtmf from going to gateway. Is there a way to filter out selective DTMF like the * or any other star codes used in bind_meta_app etc? -- Regards, Bipin

1 0

Incorrect example for 'originate' with socket and async full
by . 29 Oct '18

29 Oct '18

Hi, I wanted to report an error/typo in the confluence wiki related to 'originate', and its use together with the socket application. Apologies, I don't have access to the wiki, but thought this was worth passing on, so here it is: - https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_comma… The wiki example for ring_ready also includes the socket application and is written as: - originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929 &socket(127.0.0.1:8082 async full) however this hangs forever if used as written. Eventually, after several hours of hair tearing and googling, I stumbled on this thread: - https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.h… Indeed this is the problem. The correct syntax for the socket component is: - &socket('127.0.0.1:8082 async full') - Specifically single quotes around the command so that it is treated as a single string. After making this change there is no more hanging on the outbound calls and all events come through in a timely fashion. Cheers, Joe

4 5

transfer_fallback_extension + transfer_on_fail
by Paul Muaddib 28 Oct '18

28 Oct '18

Hi, I setup the following: When someone calls an extensions the phone rings for 30 seconds. After that the phone keeps ringing but also the phones at front desk, here 10 and 11 When the user is busy (USER_BUSY) or when the extensions does not exist (NO_PICKUP) the caller will be transferred to the front desk. This is working with transfer_on_fail Now I would like to accomplish the following: When someone does a blind transfer and the callee does not answer after 30 seconds, how can I transfer the callee back to the extension that initiated the transfer? In the documentation I found the channel variable transfer_fallback_extension. But there is not explanation to it and how does this work together with transfer_on_fail? Best regards, Paul <extension name="Extension"> <condition field="destination_number" expression="^([1-9][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="transfer_on_fail=NO_PICKUP,USER_BUSY,NO_ANSWER,SUBSCRIBER_ABSENT BRIDGE_FAILED_EXTENSION xml"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bind_digit_action" data="get_digits,~^([1-9][0-9])$,exec:lua,get_digits.lua,peer,self"/> <action application="digit_action_set_realm" data="get_digits"/> <action application="bridge" data="<ignore_early_media=true>[leg_timeout=30]user/${dialed_extension}, pickup/global, pickup/${dialed_extension}"/> <action application="bridge" data="<originate_timeout=90,ignore_early_media=true> user/${dialed_extension}, user/10, user/11, pickup/global, pickup/${dialed_extension}"/> <action application="hangup"/> </condition> </extension> <extension name="BRIDGE_FAILED_EXTENSION"> <condition> <action application="log" data="INFO ######## Extension failed - ${originate_disposition} ########" /> <condition field="${originate_failed_cause}" expression="(NO_PICKUP)|(USER_BUSY)|(NO_ANSWER)" break="on-true"> <action application="transfer" data="operator"/> </condition> <condition field="${originate_failed_cause}" expression=".*" break="on-true"> <action application="transfer" data="operator"/> </condition> </condition> </extension>

1 0

enterprise originate + originate_delay_start
by Paul Muaddib 28 Oct '18

28 Oct '18

Hi, I am trying to accomplish this bridge command but I am facing two problems. <action application="bridge" data="<originate_timeout=90,ignore_early_media=true>${group_call(werkstatt+A)},pickup/global,pickup/werkstatt:_:{originate_delay_start=25000}${group_call(buero+A)}“/> 1. When I set originate_delay_start to 25000 the group gets not called at all. If I set it to 10000, it takes about 18 seconds and not 10 seconds. 2. Even though I set ignore_early_media=true I don't hear any ringing. Best regards, Paul PS: Why is the value for originate_delay_start millisconds and for leg_delay_start it is seconds?

1 0

Identify voice quality issues using CDR records
by sukitha jayasinghe 28 Oct '18

28 Oct '18

Dear all, I have serious voice quality issues in my freeswitch 1.6.20 installation. In CDRs I found that the voice quality percentage is 100 and mos is 4.5, but the quality is very bad for one side. I have pasted the audio part of the CDR below. What mean by the skip packet count. Is it possible to identify the issue by looking at these statistics? "audio" : { "inbound" : { "raw_bytes" : 90298, "media_bytes" : 90298, "packet_count" : 1235, "media_packet_count" : 1235, "skip_packet_count" : 1184, "jitter_packet_count" : 0, "dtmf_packet_count" : 0, "cng_packet_count" : 0, "flush_packet_count" : 0, "largest_jb_size" : 0, "jitter_min_variance" : 137.097046, "jitter_max_variance" : 315.2, "jitter_loss_rate" : 0, "jitter_burst_rate" : 0, "mean_interval" : 27.274788, "flaw_total" : 0, "quality_percentage" : 100, "mos" : 4.5 },

1 0

Mod_spy part2
by Alex Goulis 27 Oct '18

27 Oct '18

Hi all...Now that I've managed to get mod_spy to work, I've noticed it only allows a perpetual listen and gives none of the dynamic dtmf or the ability to whisper on the eavesdrop app when it initiates it. I've set all the correct eavesdrop parameters before calling userspy, but they dont have any effect.Is there a way to get the eavesdrop to whisper or enable the dtmf functions to work using userspy?Thanks!

1 0

mod_spy, does it only work for registered users?
by Alex Goulis 27 Oct '18

27 Oct '18

Hi all... Does the userspy application only target registered users or can you use it to spy calls to let's say calls to a DID?

2 4

XML unescaping
by Sergey Safarov 27 Oct '18

27 Oct '18

Hello When need to assign variable value with XML symbols requires escaping is suggested to use dialplan contractions like example <action application="set"><![CDATA[ sip_h_Call-Info=<urn:nena:uid:callid:72270801514820004831: lng.001.experient.co.us>;purpose=nena-CallId]]></action> But more elegant will be <action application="set" data="sip_h_Call-Info=<urn:nena:uid:callid:72270801514820004831: lng.001.experient.co.us>;purpose=nena-CallId"/> I want add this functionality to FS. Could you give me reference to FS code where is used escaping/unescaping. Sergey

1 0