March 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Overriding SIP cause 480 with 500 Problem
by ItTeam 28 Mar '16

28 Mar '16

Hi Guys! We are having an issue where we call a remote sip client, the sip client has network issues and returns a 500 error code. the freeswitch treats it as 480, but because the client sent 500 the freeswitch overrides it to 500. is there a way to not override the error and leave it as 480? because when it returns 500 the initiating side treats it as if the freeswitch is down. LOG Record: c0c9bfae-a833-426c-a9d7-549f29783a63 2016-03-28 10:41:11.616278 [DEBUG] mod_sofia.c:425 sofia/internal/972549957605(a)fs-stb.kaymera.com Overriding SIP cause 480 with 500 from the other leg c0c9bfae-a833-426c-a9d7-549f29783a63 2016-03-28 10:41:11.616278 [DEBUG] mod_sofia.c:431 Channel sofia/internal/972549957605(a)fs-stb.kaymera.com hanging up, cause: NORMAL_CLEARING c0c9bfae-a833-426c-a9d7-549f29783a63 2016-03-28 10:41:11.636279 [DEBUG] mod_sofia.c:568 Responding to INVITE with: 500 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Overriding-SIP-cause-480-with… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Failed to set SCHED_FIFO scheduler
by Chandramouli P 28 Mar '16

28 Mar '16

Hi, Please find my below deployed environment: Environment: Microsoft Azure OS: CentOS 7.0 (64 bit) FreeSwitch Version: 1.6.6~64bit ( 64bit) I installed FS and able to make test calls. When I execute "systemctl status freeswitch" command, I am seeing below errors: *ERROR: Failed to set SCHED_FIFO scheduler (Operation not permitted)ERROR: Could not set nice level* Can anybody tell me what could be the issue and how to solve this? Thanks in advance, Chandra.

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playback timeout when using multiple files
by Nishad Musthafa 28 Mar '16

28 Mar '16

Hi, >From the wiki and through experimenting with FS, I know it's possible to playback multiple files using the playback_delimiter channel variable and then something like  <action application="set" data="playback_delimiter=!"/> <action application="set" data="playback_sleep_val=500"/> <action application="playback" data="/ram/sr8k.wav!/ram/swimp.wav"/> It is also possible to set a timeout for a single file using <action application="playback" data="{timeout=15000}file_string://${sound_dir}/music/8000/music1.wav"/> Is it possible to merge the above two? I am looking at setting a single timeout for multiple files. Is this possible? Nishad

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Relay 407 challenge to UE
by Fabio Margarido 28 Mar '16

28 Mar '16

Hello there. I'm working in a scenario in which FreeSWITCH acts as a media server and all SIP signaling is proxied through Kamailio. Every call that comes in to FreeSWITCH is simply bridged back to Kamailio and on to destination network. Registration bypasses FreeSWITCH, so 401 REGISTER challenges are dealt with by the UE directly. Now I need to talk to a network which demands INVITEs are authenticated as well. when the target network responds with a 407 Proxy Authentication Required, FreeSWITCH terminates the dialog and fs_cli shows the following error: [ERR] sofia_reg.c:2583 Cannot locate any authentication credentials to complete an authentication request for realm '"<whatever>"' Is there any way I can instruct FreeSWITCH to relay the 407 challenge back to the UE, so the authenticated INVITE can also be blindly bridged by FreeSWITCH? Thanks.

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how to register a number using gateway
by amani mansour 28 Mar '16

28 Mar '16

Good morning , I need to configure a test number in a distant freeSWITCH and register it from my freeswitch . i don't need to use a sip phone to register this number . the configuration is done but the registration i don't know how to do it , someone of you tell me to make a gateway but really i don't inderstand how to make it . if some of you can can help me i will be very very happy . thanks , with regards Amani

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Postgress protocols
by Abdul Hakeem 26 Mar '16

26 Mar '16

Hello, I would like to connect FS to Postgress via protocols other than ODBC, can anyone share an idea how to implement this ?. Regards, AH

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Re: [Freeswitch-users] mod_callcenter agent may stuck "In a queue call" forever
by TNTClaus 26 Mar '16

26 Mar '16

Hi. I faced a problem with mod call centre strange behaviour. I have a queue with the strategy "ring all", queue has got 30 agents, all of them always have status available, but only 6 agents can be really active (e.g.: registered and connected) at a time. Those 6 simultaneously answer calls (e.g. can answer 200 on the same INVITE request before CANCEL arrives). *Sometimes an agent might get stuck in the state "In a queue call"* while he is not in a call. In that case waiting agent stops getting calls. CLI command: > *callcenter_config agent set state <agent> waiting* changes state of an agent at DB, but call center still won't offer calls to him. The only fix I found is to reload mod_callcenter, but during reload all active calls would be dropped. Calls are coming one by one (maximum call rate I had ever seen at this site was ~600 calls/hour). But I didn't notice any correlation with the call rate. Agent may get stuck at any time of day with any call rate ranging between ~150 calls/hour to 400 calls/hour. Most of calls last from 5 to 20 seconds. I tried to switch from sqlite to odbc postgresql, but that changed nothing. Currently switch is connected to ODBC Postgresql. *OS version:* Linux local.pbx 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u4 (2016-02-29) x86_64 GNU/Linux *FS version:* FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit) *sqlite package:* llibsqlite3-0:amd64 3.8.7.1-1+deb8u1 amd64 SQLite 3 shared library *PostgreSQL / ODBC packages:* odbc-postgresql:amd64 1:09.03.0300-1 amd64 ODBC driver for PostgreSQL postgresql-9.4 9.4.6-0+deb8u1 amd64 object-relational SQL database, version 9.4 server *XML configs:* *callcenter/queues/queue.xml:*<queue name="dispatchers"> <param name="discard-abandoned-after" value="300"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="120"/> <param name="moh-sound" value="$${hold_music}"/> <param name="strategy" value="ring-all"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rules-apply" value="false"/> <param name="time-base-score" value="queue"/> </queue> *callcenter/agents/agent.xml* <agent name="agent@local" type="callback" contact="user/agent@local" status="Available" max-no-answer="0" wrap-up-time="1" reject-delay-time="1" busy-delay-time="1" no-answer-delay-time="1"/> that's a typical agent. They are all same except name and contact attributes. *callcenter/tiers/tier.xml* <tier agent="agent@local" queue="dispatchers" level="1" position="1"/>

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Register numbers in freeSWITCH without using sipphone
by amani mansour 25 Mar '16

25 Mar '16

Hi everyone , I would like to configure and register a sip number in freeSWITCH B from freeSWITCH A without using a sipphone . Concerning the configuration i did it .But to register this number i don't know how ? Is there someone who can help me please ???? for example if there is a file in which i can found how to register numbers ? i know that to have the list of registred number in this freeSWITCH ,you can do that : ./fs_cli sofia status profile internal reg . thank you . :) with regards Amani mansour

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outbound_caller_id vs origination_caller_id_number
by Markus Bönke 25 Mar '16

25 Mar '16

In the default configuration, the variable outbound_caller_id is set to 0000000: <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/> I understood that this value is used on an originate if when no callerid is set. fs_cli -x 'originate {originate_timeout=20}sofia/gateway/MYGATEWAY/+1111111111 &park' But also changing that value in var.xml does not change the signaled callerid, freeswitch uses also 0000000 as callerid. I think because this value is also hardcoded in switch_types.h as default. If I want to set a callerid on originate, I have to set the variable origination_caller_id_number like: fs_cli -x 'originate {originate_timeout=20}[origination_caller_id_number=022323232323]sofia/gateway/MYGATEWAY/+11111111 &park‘ So what is the idea behind the variable outbound_caller_id which is preset in vars.xml? Thanks Markus

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Configure Freeswitch with two external IPs ...
by evolvingnetworks 25 Mar '16

25 Mar '16

I have a SIP trunk provider that can provide two different services of SIP trunk, but they need two different IPs - is it possible to setup Freeswitch with two external IPs and register one SIP trunk trough one IP and the other through the other IP so calls made out go out the appropriate external IP? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Configure-Freeswitch-with-two… Sent from the freeswitch-users mailing list archive at Nabble.com.

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