March 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

asr_handle codec = (null)
by Bernd Krueger-Knauber 12 May '20

12 May '20

Hi, I'm testing unimrcp but get empty rtp packets from fs to unimrcp server when I want detect speech. The support from unimrcp told me that with my unimrcp setup is everything fine. He pointed me to the following log entry: [INFO] mod_unimrcp.c:3134 asr_handle: name = unimrcp, codec = (null), rate = 8000, grammar = (null), param = uni2 codec = (null) is wrong. But I have no idea what I can do. Everything else is working in this fs installation. Also speech generation via unimrcp is working. I use FreeSwitch 1.8.7. Any hints?

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mod call center tier behaviour
by abubacker 04 May '20

04 May '20

I am working in mod_callcenter Before my call center configuration is like this /referene : callcenter.conf.xml/ /<agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1003@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1003@default" queue="support@default" level="1" position="3"/>  <tier agent="1003@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers> </configuration> / Tier list : /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 +OK/ When I renamed the 1003@default as 1006@default / <agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1006@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1006@default" queue="support@default" level="1" position="3"/>  <tier agent="1006@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers>/ After that I have reload the mod_callcenter http://pastebin.freeswitch.org/14810 The 1003@default is not overwritten , but 1006@default has added newly why 1003@default is still exist /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 support@default|1006@default|Ready|1|3 +OK/ Is it a bug or Is it a default behaviour ? -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer:http://www.bksystems.co.in/email-policy

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FYI - new Amazon Web Services TTS option
by Nathan Neulinger 30 Apr '20

30 Apr '20

https://aws.amazon.com/blogs/aws/polly-text-to-speech-in-47-voices-and-24-l… The quality blows away everything I've tried out previously... Highly relevant option for voice prompts. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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mod_verto: subscribe events for specific accounts
by António Silva 29 Apr '20

29 Apr '20

Hi, Is it possible to subscribe using mod_verto? Right now I’ve enable in mod_verto the presence events but I get all the events for all the accounts, and is fine but i was wondering if it’s is possible to just do a single subscribe like in sip. Thanks for the help. Regards, António

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Mod_conference with many participants. Question for experts
by Gilles SAUVAIRE 10 Apr '20

10 Apr '20

Hello, I am making a conference with 100 participants, on an 8-core server, 16 gigabytes of ram dedicated to my freeswitch. 98 participant are in mute mode and only two participant active. 8 times out of 10 the conference does not work. (very poor quality hatched audio) 2 times out of 10 the conference works for a few minutes then becomes unstable. (very poor quality hatched audio) A htop at that time shows that all the cores are used but not more than 30%. There is also plenty of memory available on the server. <profile name = "000"> <param name = "domain" value = "$$ {domain}" /> <param name = "rate" value = "8000" /> <param name = "interval" value = "20" /> <param name = "caller-controls" value = "none" /> <param name = "moderator-controls" value = "none" /> <param name = "moh-sound" value = "silence" /> <param name = "suppress-events" value = "start-talking, stop-talking" /> <! - Energy level required for audio to be sent to the other users -> <param name = "energy-level" value = "100" /> <param name = "comfort-noise" value = "0" /> <param name = "conference-flags" value = "audio-always" /> </profile> I have already found a solution, just put interval at 60. <param name = "interval" value = "60" /> With this the conference with 100 participants and only 2 active works. I have many qustions 1) It's strange, the conference with all the members in mute attribute, shouldn't it normally do nothing? Mute members should be transparent when it comes to mixing, is that just broadcast? ( conference data = 66123@000++flags{endconf|nomoh|mute} ) Did I make a mistake in my profile? Is it normal ? 2) Is there an easier way to broadcast? That is, a call that "talks" to 90 calls? My idea is to make a conference with 2 or 3 participants and to broadcast this conference to all the other participants since he is not speaking. many thanks to you for your advice and ideas. Gilles

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Problems with TLS after upgrading to Buster
by Walter Behrend 06 Apr '20

06 Apr '20

Hello there, hope someone else also had the problem - and found a solution for it. My "internal" profile has TLS enabled with tlsv1, 1.1 and 1.2 - this worked like a charm on stretch. I'm using the freeswitch-repos. I upgraded to buster and here my problems started. Seems the gentls_cert only creates SHA1 (CA)Certificates - so freeswitch started with openssl error messages "md too weak". Tried at first to bypass this error by setting the tls_ciphers to "DEFAULT:@SECLEVEL=0" but this error still occured. So as a consequence, I modified the gentls_cert script and replaced everywhere the parameter -sha1 with -sha256. This error disappeared now, but the next one is coming up. It seems it does not matter which value I set for "tls_version" - in every case, my TLS enabled port only accepts TLS 1.3 connections. I have the problem that we're also using older phones which only support TLS 1.0. Error message is: tport_tls.c:157 tls_log_errors() TLS setup failed: 14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol I tried with openssl s_client and the parameters -tls1 -tls1_1 and so on - it really only worked for -tls1_3 Any idea about this? setting tls_version to tlsv1,tlsv1.1,tlsv1.2 does not change anything. Also setting the value just to tlsv1 does not help, I verified this with the phones AND with openssl s_client. Still only TLS 1.3 gives results here. Thanks in advance... BR Walter

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customer sip trunk
by Joli Martinez 31 Mar '20

31 Mar '20

Hello, I am buiding a freeswitch box for customer SIP trunking. I know how to setup a provider trunk and also how to setup extensions on Freeswitch. But never setup a customer trunk before. I have looked and maybe I am looking in the wrong place. Can someone point me in the right direction as to how this will look like or an example. I am also looking for dialplan example for inbound and outbound calls. The plan is for the customer to come in through our SBC's to the freeswitch trunking box and register with their pilot number. All numbers allocated to them should ring down that trunk. Not sure how to acomplish this on Freeswitch. Is there a how to or a sample of this? thanks,

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Problem with limit and hiredis (related with sqlite?)
by Jose Fco. Irles Durá 31 Mar '20

31 Mar '20

Hello! I use "limit hiredis" for limit concurrency in a multi-node setup. In some calls, freeswitch don't decrement the counter in redis, and after some time my customers can't call because they reach the concurrency limit. I have searched for "hiredis" lines in the logs without luck, but I have founded this in all nodes: 2020-03-31 14:55:08.766208 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [select * from basic_calls where b_uuid is not null and hostname='sip12' order by created_epoch] 2020-03-31 15:18:38.766210 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [BEGIN EXCLUSIVE] 2020-03-31 15:28:28.166208 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [select call_id,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_host,profile_name from sip_registrations where force_ping=1 and hostname='sip12' and profile_name='internal' and orig_hostname='sip12' and ping_expires > 0 and ping_expires <= 1585661308] 2020-03-31 15:41:58.246211 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [select count(*) from channels where hostname='sip12'] 2020-03-31 15:54:58.246209 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [select count(*) from channels where hostname='sip12'] 2020-03-31 16:08:58.266208 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [select count(*) from basic_calls where hostname='sip12'] 2020-03-31 16:11:58.246220 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [select count(*) from channels where hostname='sip12'] 2020-03-31 16:19:58.246213 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [select count(*) from channels where hostname='sip12'] 2020-03-31 16:35:58.246214 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [BEGIN EXCLUSIVE] 2020-03-31 16:55:38.786210 [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [select * from basic_calls where b_uuid is not null and hostname='sip12' order by created_epoch] In my setup the "core.db" is in ram with that dsn: sqlite:///dev/shm/core.db The FreeSWITCH version is 1.10.2 from deb repositories, I think that in the previous version (1.10.1) this not happens. This night I'll downgrade the version for confirm it. I have readed that in that version (1.10.2) there are some changes in the sqlite backend for use in memory: https://freeswitch.org/jira/browse/FS-11827 Is possible that some lock in the sqlite operations affect to the limit module? Best regards

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freeswitch not honoring /etc/hosts file
by Joli Martinez 30 Mar '20

30 Mar '20

Hello, I have a customer that has a VPN tunnel to my network. I would like this customer to connect to me directly not through the SBC. I have created routing between the Freeswitch box and my registrar server. I have also created the entry in the /etc/hosts file. When I ping the server I get the correct response back. The issue is that Freeswitch is ignoring the /etc/hosts entry and doing its own DNS resolution. How can I tell freeswitch to honor the /etc/hosts file entry first then use DNS? I am running Debian 10 with freeswitch 1.10.2 Thanks,

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getting sql errors when using postgre with stable 1.10 branch
by Babak Yakhchali 29 Mar '20

29 Mar '20

Hi I'm using freeswitch version 1.10.2-release+git~20191231T140119Z~f7bdd3845a~64bit on debian 10. after some days logs show errors like this: 2020-03-28 14:23:35.669184 [DEBUG] mod_pgsql.c:747 Query (SELECT queue,uuid,session_uuid,cid_number,cid_name,joined_epoch,(1585389215-joined_epoch)+base_score+skill_score AS score, state, abandoned_epoch, serving_agent, instance_id FROM members WHERE (state = 'Waiting' OR state = 'Abandoned' OR (serving_agent = 'ring-all' AND state = 'Trying') OR (serving_agent = 'ring-progressively' AND state = 'Trying')) AND instance_id = 'single_box' ORDER BY score DESC) returned PGRES_FATAL_ERROR 2020-03-28 14:23:35.669184 [ERR] switch_core_sqldb.c:1347 ERR: [SELECT queue,uuid,session_uuid,cid_number,cid_name,joined_epoch,(1585389215-joined_epoch)+base_score+skill_score AS score, state, abandoned_epoch, serving_agent, instance_id FROM members WHERE (state = 'Waiting' OR state = 'Abandoned' OR (serving_agent = 'ring-all' AND state = 'Trying') OR (serving_agent = 'ring-progressively' AND state = 'Trying')) AND instance_id = 'single_box' ORDER BY score DESC] [ERROR: current transaction is aborted, commands ignored until end of transaction block ] 2020-03-28 14:23:35.669184 [ERR] switch_core_sqldb.c:1347 ERR: [SELECT queue,uuid,session_uuid,cid_number,cid_name,joined_epoch,(1585389215-joined_epoch)+base_score+skill_score AS score, state, abandoned_epoch, serving_agent, instance_id FROM members WHERE (state = 'Waiting' OR state = 'Abandoned' OR (serving_agent = 'ring-all' AND state = 'Trying') OR (serving_agent = 'ring-progressively' AND state = 'Trying')) AND instance_id = 'single_box' ORDER BY score DESC] [ERROR: current transaction is aborted, commands ignored until end of transaction block this errors keep showing like a loop and it keeps giving errors till I restart postgre and freeswitch is this related to missconfiguration of core dsn? this is the connection string I'm using: pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='freeswitch' options='-c client_min_messages=NOTICE' application_name='freeswitch'

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