September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

For those ubuquiti lovers.
by Blake Priddy 03 Sep '14

03 Sep '14

Would be awesome to throw csipsimple on this or if the app on there works. Ordering 2 for fun :) http://www.ubnt.com/unifi-voip/overview/ -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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RECOVERY_ON_TIMER_EXPIRE + Changing Hangup Cause
by Massimo Varriale 03 Sep '14

03 Sep '14

Hi All! This is my first question...and what question... I'm a skilled developer and I'm almost familiar with telephony in general since it's 15 years experience in telco companies. However I'm still facing a problem I couldn't find any solution yet and i hope someone could help in finding a solution As my present company can't afford an expensive soft switch I'm using Freeswitch 1.4 with a simple dialplan.xml and some scripts to load CDRs into mysql to send incoming calls to my remote peers. At present the live traffic is only 1 incoming IP and 1 outgoing peer, quite easy. On some calls I'm getting "408 Request Timeout" and FS is giving me a RECOVERY_ON_TIMER_EXPIRE cause (IDSN 102). The problem is that as most of the switches are not able to make automatic rerouting based on this cause and so, I will like to use instead the ISDN Cause 18 (I suppose "480 Temporarily unavailable"). Is this possible? I've used this snippet of code that use another relcause but with same concept (taken as is and pasted into the dialplan) but doesn't work as the 408 is given after 183. <extension name="AllDestinations_Inbound"> <condition field="destination_number" expression="^"> <condition field="network_addr" expression="^154\.48\.198\.190$"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=19"/> <action application="bridge" data="sofia/gateway/remote_outbound_peer/${destination_number}"/> <action application="transfer" data="408to503"/> </condition> </extension>  <extension name="408to503"> <condition field="${proto_specific_hangup_cause}" expression="<a href="sip:408">sip:408"> <action application="set" data="sip_ignore_remote_cause=true"/> <action application="respond" data="503"/> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> How to correctly remap causes? If there is no way to change normal behaviour, do you think could be possible also to change the source code and recompile FS accordly? Thank yo so much and sorry for this long post I hope someone could help Max

3 3

Codec Error
by Ali R. 03 Sep '14

03 Sep '14

Hello All, I'm getting the below error while in pass-through mode (proxy_media=true) but I get success when the disable transcoding is set on the sofia profile. [WARNING] switch_core_codec.c:703 Codec G723 Exists but not at the desired implementation. 8000hz 20ms [ERR] switch_core_media.c:2285 Can't load codec? I did hear about broken SIP devices out there but since it works fine when disable transcoding is set, I thought I give it a shot on the mailing list and while at it, this error does not occur in case of another pass-through codec(G729) Here are my settings when in proxy media mode: Sofia Profile: inbound-late-negotiation = true inbound-codec-negotiation = greedy (tried scrooge) I also tried rtp-autofix-timing=true (no luck) DialPlan: proxy_media=true bridge FS version tried: 1.5.13b & 1.2 with proxy-media (failed) http://pastebin.freeswitch.org/23205 with disable-transcoding is set (success) http://pastebin.freeswitch.org/23206 Your help or suggestion is very much appreciated. Best Regards, Ali_R

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Force minimum conference member dinamically
by Antonio Silva 03 Sep '14

03 Sep '14

Hi, Is it possible to set the minimum members in a conference dynamically? for example: conference <confname> min 1 In my case, in doing conference with bridge, from the dialplan: <application="conference" data="bridge:....."> But this forces the minimum number of members to 2, so when one member hangup, the conference is ended. The idea is to still use the bridge conference, but once the conference is created via an external command change the minimum conference members to 1. Thanks, António

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Re: [Freeswitch-users] perl startup scripts
by Preeti Trivedi 03 Sep '14

03 Sep '14

Hi Brian, DBI is used in all three perl scripts and each script will have distinct connections. On Tue, Sep 2, 2014 at 9:04 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. perl startup scripts (Preeti Trivedi) > 2. Re: perl startup scripts (Brian West) > 3. Re: Inband DTMF Woes (Brian West) > 4. Re: Allowing calls with Mod nibblebill and zero balances (DP .) > 5. Re: Freeswitch-1.4.7 source rpm (Ken Rice) > > > ---------- Forwarded message ---------- > From: Preeti Trivedi <sweetpt31(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Tue, 2 Sep 2014 19:53:51 +0530 > Subject: [Freeswitch-users] perl startup scripts > Hi, > > I am facing a problem with FS getting dumped while starting up. > > It worked fine when I had 2 startup-scripts in perl initially. I didn't > face any problem in FS starting up. > I added the 3rd script today and now FS is getting dumped while starting > up few times. Few times, it is starting up without any problem.. The > behaviour is random. > Is there any parameter which I've to tune? any idea? > > FS version: > FreeSWITCH Version 1.5.14b+git~20140804T185031Z~1960533260~64bit (git > 1960533 2014-08-04 18:50:31Z 64bit) > > Output of 'scripts/backtrace-from-core' script: > http://pastebin.freeswitch.org/23197 > > Regards, > Preeti > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 2 Sep 2014 09:43:27 -0500 > Subject: Re: [Freeswitch-users] perl startup scripts > Are you using DBI in any of these scripts? > > > On Tue, Sep 2, 2014 at 9:23 AM, Preeti Trivedi <sweetpt31(a)gmail.com> > wrote: > >> Hi, >> >> I am facing a problem with FS getting dumped while starting up. >> >> It worked fine when I had 2 startup-scripts in perl initially. I didn't >> face any problem in FS starting up. >> I added the 3rd script today and now FS is getting dumped while starting >> up few times. Few times, it is starting up without any problem.. The >> behaviour is random. >> Is there any parameter which I've to tune? any idea? >> >> FS version: >> FreeSWITCH Version 1.5.14b+git~20140804T185031Z~1960533260~64bit (git >> 1960533 2014-08-04 18:50:31Z 64bit) >> >> Output of 'scripts/backtrace-from-core' script: >> http://pastebin.freeswitch.org/23197 >> >> Regards, >> Preeti >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.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 >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian(a)freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > >

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How to place incoming unauth calls to different contexts.
by Pavel 03 Sep '14

03 Sep '14

Hello Everyone ! My /sip_profiles/external.xml looks like: <profile name="external"> <gateways> <gateway name="gw1"> <param name="username" value="username"/> <param name="realm" value="192.168.1.1"/> <param name="password" value="password"/> <param name="proxy" value="192.168.1.1"/> <param name="register" value="false"/> <param name="context" value="gw1"/> </gateway> <gateway name="gw2"> <param name="username" value="username"/> <param name="realm" value="192.168.1.2"/> <param name="password" value="password"/> <param name="proxy" value="192.168.1.2"/> <param name="register" value="false"/> <param name="context" value="gw2"/> </gateway> </gateways> <aliases> <alias name="outbound"/> <alias name="nat"/> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-zrtp-passthru" value="true"/>  <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_tls_port}"/> <param name="tls-passphrase" value=""/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-policy" value="none"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="$${sip_tls_version}"/> </settings></profile> Incoming unauthenticated calls from either gw1 or gw2 keep hitting context "public". Any way to place them in the appropriate contexts (eg. "gw1" and "gw2"). Thanks for your attention. Rergards,Pavel.

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Re: [Freeswitch-users] Can't register more than one user to freeswitch
by Snabel Kabiya 03 Sep '14

03 Sep '14

Hi, now i can't register both of them and i get the error for both users. <include> <user id="4444"> <params> <param name="password" value="4444"/> </params> </user> </include> <include> <user id="1111"> <params> <param name="password" value="1111"/> </params> </user> </include> Snabel, > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 3 Sep 2014 07:36:13 -0500 > Subject: Re: [Freeswitch-users] Can't register more than one user to > freeswitch > You do not need domain tags on the 1111.xml or 4444.xml the domain is > wrapped from a higher level include, the possibly reason 1111 is working > and not 4444 is the domain is listed twice in the XML and the first one > found doesn't contain a user 4444 in it. > > > On Wed, Sep 3, 2014 at 7:18 AM, Snabel Kabiya <snabel(a)lexifone.com> wrote: > >> Hi, >> >> I'm using X-lite to test mod_sms in freeswitch. >> I've created two users on server x.x.x.1 >> >> <include> >> <domain name="x.x.x.1"> >> <user id="1111"> >> <params> >> <param name="password" value="1111"/> >> </params> >> </user> >> </domain> >> </include> >> >> <include> >> <domain name="x.x.x.1"> >> <user id="4444"> >> <params> >> <param name="password" value="4444"/> >> </params> >> </user> >> </domain> >> </include> >> >> I've but both files: 1111.xml, 4444.xml in >> /usr/local/freeswitch/conf/directory. restarted freeswitch after that. >> >> >> i managed to register user 1111 using X-Lite, but when i tried to >> register user 4444, i got this error on freeswitch: >> >> >> >> *2014-09-03 15:05:57.177795 [WARNING] sofia_reg.c:2566 Can't find user >> [4444(a)x.x.x.1] from x.x.x.xYou must define a domain called 'x.x.x.1' in >> your directory and add a user with the id="4444" attribute and you must >> configure your device to use the proper domain in it's authentication >> credentials.* >> >> is there some limitation in freeswitch? do i need to change some >> configuration file? >> >> >> Thanks, >> Snabel >> >> >> Hi, >> >>> >>>> i'm trying to use mod_sms. >>>> >>>> i did the following steps >>>> 1 - used the instructions here to compile the module >>>> http://wiki.freeswitch.org/wiki/Mod_sms#Args >>>> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_chat >>>> >>>> created two users on both servers x.x.x.1, x.x.x.2 >>>> >>>> <domain name="x.x.x.1"> >>>> <user id="2222"> >>>> <params> >>>> <param name="password" value="2222"/> >>>> </params> >>>> </user> >>>> </domain> >>>> >>>> <domain name="x.x.x.1"> >>>> <user id="1111"> >>>> <params> >>>> <param name="password" value="1111"/> >>>> </params> >>>> </user> >>>> </domain> >>>> >>>> created this chatplan: >>>> >>>> public.xml: >>>> <?xml version="1.0" encoding="utf-8"?> >>>> <include> >>>> <context name="public"> >>>> <extension name="public"> >>>> <condition field="to" expression="^(.*)$"> >>>> <action application="log" data="INFO to_chat remote"/> >>>> <action application="reply" data="Hello, you said: >>>> ${_body}"/> >>>> </condition> >>>> </extension> >>>> </context> >>>> </include> >>>> >>>> >>>> but now when i send this from server x.x.x.2 >>>> freeswitch@internal> chat sip|2222@x.x.x.2|external/sip:1111@x.x.x.1|hi >>>> from freeswitch snabel is here >>>> Sent >>>> >>>> i get the following on server x.x.x.1 >>>> 2014-09-03 11:18:18.780102 [INFO] mod_sms.c:336 Processing text message >>>> 2222->1111 in context public >>>> Chatplan: 1111 parsing [public->public] continue=false >>>> Chatplan: 1111(a)x.x.x.1 Regex (PASS) [public] to(1111(a)x.x.x.1) =~ >>>> /^(.*)$/ break=on-false >>>> Chatplan: 1111(a)x.x.x.1 Action log(INFO to_chat remote) >>>> Chatplan: 1111(a)x.x.x.1 Action reply(Hello, you said: ${_body}) >>>> *2014-09-03 11:18:18.780102 [ERR] switch_loadable_module.c:769 Invalid >>>> chat application interface [log]!* >>>> >>>> >>>> what does it mean? what did i miss? >>>> >>>> Thanks, >>>> Snabel >>>> >>>

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Getting Freeswitch to proxy SIPTLS/SRTP
by Tim Smith 03 Sep '14

03 Sep '14

Hi, Does anything need to be done to enable Freeswitch to correctly proxy SIPTLS/SRTP ? i.e. VoIP provider with TLS/SRTP on one side, non-TLS/SRTP capable SIP clients on the other side. Thanks ! Tim

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Can't register more than one user to freeswitch
by Snabel Kabiya 03 Sep '14

03 Sep '14

Hi, I'm using X-lite to test mod_sms in freeswitch. I've created two users on server x.x.x.1 <include> <domain name="x.x.x.1"> <user id="1111"> <params> <param name="password" value="1111"/> </params> </user> </domain> </include> <include> <domain name="x.x.x.1"> <user id="4444"> <params> <param name="password" value="4444"/> </params> </user> </domain> </include> I've but both files: 1111.xml, 4444.xml in /usr/local/freeswitch/conf/directory. restarted freeswitch after that. i managed to register user 1111 using X-Lite, but when i tried to register user 4444, i got this error on freeswitch: *2014-09-03 15:05:57.177795 [WARNING] sofia_reg.c:2566 Can't find user [4444(a)x.x.x.1] from x.x.x.xYou must define a domain called 'x.x.x.1' in your directory and add a user with the id="4444" attributeand you must configure your device to use the proper domain in it's authentication credentials.* is there some limitation in freeswitch? do i need to change some configuration file? Thanks, Snabel Hi, > >> i'm trying to use mod_sms. >> >> i did the following steps >> 1 - used the instructions here to compile the module >> http://wiki.freeswitch.org/wiki/Mod_sms#Args >> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_chat >> >> created two users on both servers x.x.x.1, x.x.x.2 >> >> <domain name="x.x.x.1"> >> <user id="2222"> >> <params> >> <param name="password" value="2222"/> >> </params> >> </user> >> </domain> >> >> <domain name="x.x.x.1"> >> <user id="1111"> >> <params> >> <param name="password" value="1111"/> >> </params> >> </user> >> </domain> >> >> created this chatplan: >> >> public.xml: >> <?xml version="1.0" encoding="utf-8"?> >> <include> >> <context name="public"> >> <extension name="public"> >> <condition field="to" expression="^(.*)$"> >> <action application="log" data="INFO to_chat remote"/> >> <action application="reply" data="Hello, you said: >> ${_body}"/> >> </condition> >> </extension> >> </context> >> </include> >> >> >> but now when i send this from server x.x.x.2 >> freeswitch@internal> chat sip|2222@x.x.x.2|external/sip:1111@x.x.x.1|hi >> from freeswitch snabel is here >> Sent >> >> i get the following on server x.x.x.1 >> 2014-09-03 11:18:18.780102 [INFO] mod_sms.c:336 Processing text message >> 2222->1111 in context public >> Chatplan: 1111 parsing [public->public] continue=false >> Chatplan: 1111(a)x.x.x.1 Regex (PASS) [public] to(1111(a)x.x.x.1) =~ >> /^(.*)$/ break=on-false >> Chatplan: 1111(a)x.x.x.1 Action log(INFO to_chat remote) >> Chatplan: 1111(a)x.x.x.1 Action reply(Hello, you said: ${_body}) >> *2014-09-03 11:18:18.780102 [ERR] switch_loadable_module.c:769 Invalid >> chat application interface [log]!* >> >> >> what does it mean? what did i miss? >> >> Thanks, >> Snabel >> >> >> >

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mod_sms chatplan / unable to find registered user
by Daniel Ivanov 03 Sep '14

03 Sep '14

When i receive an SMS on my chatplan from the external profile(5080 port), the send application refuses to lookup the sofia_contact. What is a correct and secure way to enable users from the internal profile to be enable for lookup on the external?

1 0