May 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to let freeswitch make a transfer call without dialplan xml?
by Leo Wolf 14 May '19

14 May '19

Hi, I want to make a transfer call on freeswitch. I use event socket to receive calls, do the calling invite logic in the program. Now I want to do the same things for transfering call. But the commands I found for transfer are all depending on dialplan xml. E.x., uuid_transfer, transfer, ..., etc. But what I want is do the transfer call without any xml, even the default. Is there any way to achieve that goal on freeswitch? Sincerely, Leo

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Failed Connection
by Sean Devoy 13 May '19

13 May '19

Hi, My user said "It just never connected" referring to an outbound call. It looks like everything is going great until the RTP is starting and then splat. This all happens in 2 seconds or less. I have tried different gateways and 2 different NAT'ed phone locations. The server is not NAT'ed. These have only started since I updated to FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit). But other things have changed too! I created a paste bin of my log: https://pastebin.com/9nw39DaR The lines in the log that confuse me are bordered by rows of ******. >From my clueless reviews, it looks like it may be related top CODEC (re)negotiation. Thanks for any insights you can share. Sean

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set_audio_level being ignored
by Marcelo Yukio Bressan Hossomi 13 May '19

13 May '19

Hi everyone, I'm trying a dialplan like this (inline, added linebreaks for readability): set_audio_level:write -4, playback:shouts://server.com, set_audio_level:write 4, playback:shouts://server.com But the second audio always plays at level 0. I noticed that: - It works most of the time if I add a sleep:50 before the second `set_audio_level`, but this seems hacky; - I see the log "Removing BUG from ..." right after the second audio starts; it doesn't happen when I have sleep:50 as above. Audios are about 3s long. Using code up to commit 78c189bfcc2. Other non-working scenarios (same behavior): set_audio_level:write -4, playback:shouts://server.com, playback:shouts://server.com set_audio_level:write -4, playback:shouts://server.com, set_audio_level:write 4, set_audio_level:write -4, playback:shouts://server.com Any hint is appreciated!

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forward calls
by Daniel Dimitrov 13 May '19

13 May '19

Hello everyone, I am new to freeswitch and fusionpbx . I need help to forward calls coming from mobile A to local phone (818) and send them to mobile B.Here is what i have tried but it doesn't work. Also calls are coming from my provider in public context then send to my ext in private context . phone:03312 818 private cx | Mobile ---- > FS ----------> Mobile A public B <extension name="myphones" continue="false" uuid="7acf4c28-45a0-47d1-b929-c9de90c73428"> <condition field="destination_number" expression="^(03312\d{3})$"> <action application="set" data="origination_caller_id_number=03312${dialed_user}"/> <action application="set" data="effective_caller_id_number=${dialed_user}"/> <action application="info" data=""/> <action application="bridge" data="sofia/gateway/0a8d2462-403e-4923-ade9-10b24739adec/$1"/> </condition> </extension> appreciate any advice

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Nightmare transfer question
by Ryan Delgrosso 11 May '19

11 May '19

So I have a cluster of FS servers using a common shared DB. There is host weighting and some loosely intelligent dialplan generation to attempt to get related dialogs started on the same instance but I still occasionally encounter the "nightmare xfer" section of sofia.c which basically guarantees this call is doomed. To be clear, this happens when for some reason an endpoint ends up on a call through FS1, then makes an outbound call that for some cacophony of reasons ends up on FS2 and then hit xfer. There isnt anything in the signaling from most endpoints that indicate to me the invite beginning the 2nd leg is implicitly related to the 1st so i cant do any clever routing at the outset. What can i do to try and make this recoverable? Since i dont even get a dialplan request for the refer im struggling with how to affect this in a way that might make the call succeed. Thanks in advance -Ryan

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NOTICE: Jira is down for maintenance. Please be patient.
by Andrey Wolk 10 May '19

10 May '19

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Re: [Freeswitch-users] Occasional Dropped call at 30 seconds
by Sean Devoy 10 May '19

10 May '19

Thanks Nathan. That is a big help. Maybe I can finally see what is really happening. Sean From: Nathan Neulinger <nneul(a)mst.edu> Sent: Friday, May 10, 2019 10:17 AM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>; Sean Devoy <sdevoy(a)bizfocused.com> Subject: Re: [Freeswitch-users] Occasional Dropped call at 30 seconds It doesn't address your actual issue - but for the pcap - as long as you can respond in a reasonable timeframe for the hit, you can use the tcpdump file rotation options "-C size -W count" to just set up a capture that will rotate through files. It'll create a lot of capture, but it will only keep the last N files. -- Nathan ________________________________ From: Sean Devoy <sdevoy(a)bizfocused.com><mailto:sdevoy@bizfocused.com> Sent: Fri, May 10, 2019 9:10 AM CDT To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org><mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Occasional Dropped call at 30 seconds Hi all, I have a problem with one client where the call is fine and they can talk for almost exactly 30 seconds, then the log says originating phone (of FS) hung up. This only happens like 1% to 5% of the time. The phones are Mitel 6867i and 6865i which must use TCP not UDP. The firewall is a $#!#&* SonicWall. I am 99% convinced it is the firewall. It happened once on a Cisco phone using UDP so I increased the firewall UDP timeout. I already have keep-alive set to 30 seconds. It has not recurred on UDP phones since. Any ideas on something to do to Mitel config or SonicWall that might prevent a drop after 30 seconds? I have considered PCAP, but with so few occurrences I am worried about PCAP size before I get a "hit." Thanks, Sean _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales(a)freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu<mailto:nneul@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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PASSed dialplan ignored?
by Sean Devoy 10 May '19

10 May '19

Can anyone explain this to me? 79c47900-43c6-4e2a-be5d-42883371a823 Dialplan: sofia/internal/220@<my.fq.dn) Regex (PASS) [getvoicemail] destination_number(*8686) =~ /^\*8686$/ break=on-false 79c47900-43c6-4e2a-be5d-42883371a823 2019-05-07 18:08:53.382829 [INFO] switch_core_state_machine.c:311 No Route, Aborting 79c47900-43c6-4e2a-be5d-42883371a823 2019-05-07 18:08:53.382829 [NOTICE] switch_core_state_machine.c:312 Hangup sofia/internal/220@<my.fq.dn> [CS_ROUTING] [NO_ROUTE_DESTINATION] How can it be REGEX (PASS) and have no route? Thanks, Sean

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Is there device / extension state functionality at Freeswitch or how is to get User/Gateway state ?
by CpServiceSPb 10 May '19

10 May '19

There is Freeswitch 1.8.5 built from sources worked on x64 Linux. I need to check / get internal users / trunck gateways state BEFORE using it to perform different actions in response of got conditions: 1. the called number is not exists in the directory at all; 2. the called number exist in the directory, but not registered at the time, for example don' t launch softphone; 3. the called number is inuse; 4. the called number is busy; 5. the called number is not answered due timeout 6. the called number is answered. Is there functionality looks like device and / or extension (user) state at Freeswitch ? And if I could still get at least 1 and 2 states and 5 of a user, I can' t get inuse, busy states. How is the best way to get these different states for internal users, for each one and /or for trunk gateways, per each also ?

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Re: [Freeswitch-users] redirect + swapping of ports
by Vrajendra VT 10 May '19

10 May '19

Hi, Still awaiting for the reply. Thanks. On Thursday, May 9, 2019, 3:44:28 PM EDT, Vrajendra VT via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: + freeswitch-users(a)lists.freeswitch.org On Thursday, May 9, 2019, 8:51:16 AM GMT+5:30, Vrajendra VT <vrajendra.vt(a)yahoo.com> wrote: Thank you Jose for the reply. The reason for doing this:Lets just say FS2 will be replaced with N no of FSs in future So, FS1 will redirect to FS2 .... FSN, depending upon which is free (whose load is less).As of now, will take FS1 and FS2 as an example. Since I have swapped the ports. the incoming call (on 5060) is handled by external profile (rather than internal) since I have swapped the ports in vars.xml. Since I am only redirecting from FS1, I have changed only public.xml something like this on FS1 <extension name="public_extensions" continue="true"> <condition field="destination_number" expression="^(\d+)$"> <action application="redirect" data="sip:1011@[IP of FS2]"/> </condition> </extension> After this, I am seeing the following in the logs, 2019-05-09 03:05:29.267994 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 302 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:60 sofia/external/+16617480240@[IP from SIP PROVIDER] Standard HANGUP, cause: NORMAL_UNSPECIFIED 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:852 (sofia/external/+16617480240@[IP from SIP PROVIDER]) State HANGUP going to sleep 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:619 (sofia/external/+16617480240@[IP from SIP PROVIDER]) State Change CS_HANGUP -> CS_REPORTING 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:584 (sofia/external/+16617480240@[IP from SIP PROVIDER]) Running State Change CS_REPORTING (Cur 2 Tot 7538) 2019-05-09 03:05:29.267994 [DEBUG] switch_core_state_machine.c:938 (sofia/external/+16617480240@[IP from SIP PROVIDER]) State REPORTING 2019-05-09 03:05:29.428022 [ERR] mod_cdr_csv.c:156 Error opening /var/log/freeswitch/cdr-csv/Master.csv 2019-05-09 03:05:29.428022 [DEBUG] switch_core_state_machine.c:174 sofia/external/16617480240@[IP from SIP PROVIDER] Standard REPORTING, cause: NORMAL_UNSPECIFIED 2019-05-09 03:05:29.428022 [DEBUG] switch_core_state_machine.c:938 (sofia/external/16617480240@[IP from SIP PROVIDER]) State REPORTING going to sleep 2019-05-09 03:05:29.428022 [DEBUG] switch_core_state_machine.c:610 (sofia/external/16617480240@[IP from SIP PROVIDER]) State Change CS_REPORTING -> CS_DESTROY 2019-05-09 03:05:29.428022 [DEBUG] switch_core_session.c:1665 Session 7537 (sofia/external/16617480240@[IP from SIP PROVIDER]) Locked, Waiting on external entities 2019-05-09 03:05:29.428022 [NOTICE] switch_core_session.c:1683 Session 7537 (sofia/external/16617480240@[IP from SIP PROVIDER]) Ended 2019-05-09 03:05:29.428022 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/16617480240@[IP from SIP PROVIDER] [CS_DESTROY] One thing I observed, "+" missing in source number for second half of logs. After corresponding redirect, I am not seeing any logs on FS2.Could there be issue in FS2 configs? Thanks. On Thursday, May 9, 2019, 12:22:44 AM GMT+5:30, Jose Figueroa <josen.figueroa(a)unixmexico.org> wrote: Hello Vrajendra, There are several ways to achieve this. The question is what is the reason you're doing this? It would be good to understand the scenario.In VoIP, you can use prefixes to redirect incoming calls or outgoing calls to specific gateways or even use sip proxy strategies. In addition, if you share your config here I am pretty sure someone from the mailing list can help you. ThanksJose N Figueroa On Wed, May 8, 2019 at 12:40 PM Vrajendra VT via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: ---------- Forwarded message ---------- From: Vrajendra VT <vrajendra.vt(a)yahoo.com> To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Cc: Bcc: Date: Wed, 8 May 2019 14:19:04 +0000 (UTC) Subject: redirect + swapping of ports Hi, I am a novice in Freeswitch. I have a provider sending me calls on 5060 port only. I have 2 FS's. Calls are landing on FS1. From FS1, I need to redirect to FS2 (lets assume all calls to be redirected) With normal config's (I had done few config changes in default and public), I was able to get some logs albeit (Proxy Authentication error). This may be because by default internal calls are authenticated. Thats why I swapped ports 5060 and 5080 of internal and external profiles. But I am getting 480 error Any help is appreciated. Thanks. Thanks. ---------- Forwarded message ---------- From: Vrajendra VT via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Cc: Bcc: Date: Wed, 08 May 2019 10:40:28 -0700 (PDT) Subject: [Freeswitch-users] redirect + swapping of ports _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales(a)freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales(a)freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com

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