February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS 1.0.7 crash on originate
by Anirudha Shimpi 04 Feb '11

04 Feb '11

Just downloaded (from git) and installed FS 1.0.7, however, every time I do originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit.

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H.323 implementation on FS
by Vedran Zeljeznak 04 Feb '11

04 Feb '11

hi everyone, do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)? Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch? --- Vedran Zeljeznak

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How to use sofia_gateway_data ?
by David Ponzone 04 Feb '11

04 Feb '11

All, I am still trying to reroute a call in case a gateway is down, but I decided to do that another way. I'd like to detect my gateway is down, and I was trying to use sofia_gateway_data function for that purpose. But I can't figure out the expected args. I try: sofia_gateway_data <gatewayname> var Status or ivar/ovar instead of var I tried with both a gateway I register to, and an IP-auth-based gateway I ping regularly. So I am stuck now, and I can't find any info anywhere on how to detect a gateway is flagged as DOWN. TIA David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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deflect
by Sam 04 Feb '11

04 Feb '11

Hi, When call comes on 1 server and plays an application and after execution of the application the call is bridge to the other server ,but here after bridging the call should refer/deflect to other server, how this can be done ? Here just using the deflect variable is not recommended as there is proxy in between, so once the call is bridge the next step would be deflect the leg totally to another server via proxy. Regards Sam

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Strange behaviour with leg_timeout
by David Ponzone 04 Feb '11

04 Feb '11

It's my time to mess with XXX_timeout variables in FreeSWITCH. What I need to do is to route a call to a gateway, and if I don't have an answer (like not even the 100) within X seconds, I try to route the call to a backup number. I tried it this way: <extension name="gw_then_backup"> <condition field="destination_number" expression="^(XXXXXXXXXX)$"> <action application="bridge" data="{leg_timeout=5}sofia/external/$1(a)a.b.c.d|sofia/gateway/provider.out/YYYYYYYYYY"/> </condition> </extension> My understanding was that a.b.c.d would be tried for 5 seconds, and if no 100 is received, gateway provider.out would then be used. It works...but after the first INVITE to provider.out (which sends back 100/180/183), I can see some more INVITEs sent to a.b.c.d. I am wondering if I am doing something wrong, or if there is an issue. I use latest git from 5 minutes ago. Thanks David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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play_and_get_digits and macros
by Madovsky 04 Feb '11

04 Feb '11

on speech managment wiki page there's explanation of how to write XML for a macro, but where to put this XML code ? I'd like to use a macro in play_and_get_digits, possible ? if yes what's the syntax ? Thanks in advance

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mod_fsk not detecting call waiting caller id
by Anirudha Shimpi 04 Feb '11

04 Feb '11

I have an analog line connected through an SPA3102 forwarded to an extension on FS. Line has call waiting ID on it (verified). I get the tone when call waiting is sent. A sample dialplan is as follows <action application="answer"/> <action application="sleep" data="1000"/> <action application="fsk_recv"/> <action application="set" data="execute_on_fsk=log 'Got FSK [${fsk_phone_num}] [${fsk_phone_name}]'"/> <action application="transfer" data="1002 XML default"/> When call waiting tone is detected I get the following 2011-02-03 17:25:06.089086 [DEBUG] mod_fsk.c:276 sofia/internal/xxxxx@xxxxx processing execute_on_fsk [log 'Got FSK [] []'] EXECUTE sofia/internal/xxxx@xxxx log('Got FSK [] []') 2011-02-03 17:25:06.089086 [DEBUG] mod_dptools.c:1183 FSK [] []' 2011-02-03 17:25:06.269078 [DEBUG] switch_core_media_bug.c:467 Removing BUG from sofia/internal/xxxxx@xxxx If I do a uuid_dump, the fsk_ (except variable_execute_on_fsk) variables are not present. Any idea what I could be doing wrong here? Never used mod_fsk before, however, need to just log call waiting name and number. Any help would be greatly appreciated.

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Portaudio Improvements
by Mitch Capper 04 Feb '11

04 Feb '11

I have submitted a ticket with a patch for portaudio and my improvements. This includes the improvements I had discussed previously during my call for input on the mailing list and is available at: http://jira.freeswitch.org/browse/FS-3006 It was only tested in windows, however most of the changes should not effect default behavior (but please test if you can). To test you will want to enable some things in the portaudio config including: <param name="live-stream-switch" value="true" /> <param name="no-auto-resume-call" value="true" /> <param name="no-ring-during-call" value="true" /> you will then be able to use the new features fully. Many of these changes were made to add better support for softphone's using freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for the conference call in a week and a half it is .net/c# but WPF so sadly windows only currently. jlink and drk are helping so we should end up with a nice installer also for it. If you would like to test it please let me know. ~Mitch

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voice quality in conference
by Madovsky 04 Feb '11

04 Feb '11

is the audio of outbound call can be normalized when it enters in conference ? it seems that some phones with mic wit low sensitivy level have the start of sentence cut or hashed, some "ssss" and low level voice are not heard in conference. if I make a normal bridge with the same phones the problem not appears.. Thanks

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Enterprise bridge not continuing?
by Tim St. Pierre 04 Feb '11

04 Feb '11

Hello, I'm trying to implement a ring group scenario that supports multiple registrations per user. I have been able to get it to work (mostly) by using the enterprise bridge syntax, but when I do, I can't continue on in the dialplan to go to voicemail, for example. If I change the syntax and use the parallell dial comma, voice mail works fine, but only the first registration of each user entry will ring, and I would really like them all to ring. From what I can tell, dialplan execution stops after the originate, rather than continuing on to process the next steps. Is there some sort of variable I have to set to tell it to continue when using enterprise bridge? Here's the XML debug output of the dialplan I'm sending from xml_curl <?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan"> <context name="internal"> <extension name="Intercom" description="Local Extension" continue="true"> <condition field="destination_number" expression="5240"> <action application="pre_answer" data=""/> <action application="set" data="continue=true"/> <action application="set" data="cdr_destination=Internal"/> <action application="set" data="ringback=$${us-ring}"/> <action application="set" data="sip_callee_id_name=Lab 0"/> <action application="set" data="sip_callee_id_number=5240"/> <action application="set" data="progress_timeout=4"/> <action application="set" data="call_timeout=12"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="cdr_amaflags=INTERNAL"/> <action application="set" data="originate_continue_on_timeout=true"/> <action application="set" data="originate_disposition=NO_ANSWER"/> <action application="set" data="ignore_early_media=true"/> <action application="log" data="INFO Adding user 5242to the group"/> <action application="log" data="INFO Adding user 5244to the group"/> <action application="bridge" data="[call_timeout=12,alert_info=${domain};info=alert-internal]user/5240@communicatefreely.net:_:[ignore_early_media=true,call_timeout=12,origination_caller_id_name=Cordles-${caller_id_name},origination_caller_id_number=${caller_id_number},export_vars=accountcode subdomain,alert_info=${domain};info=alert-group]user/5242@${domain}:_:[ignore_early_media=true,call_timeout=12,origination_caller_id_name=Cordles-${caller_id_name},origination_caller_id_number=${caller_id_number},export_vars=accountcode subdomain,alert_info=${domain};info=alert-group]user/5244@${domain}"/> </condition> </extension> <extension name="Voicemail" description="Leave VM message" continue="true"> <condition field="destination_number" expression="5240"> <action application="set" data="playback_terminators=#0"/> <action application="set" data="cdr_destination=VM Box 5240"/> <action application="answer" data=""/> <action application="set" data="vm_busy_greet=0"/> <action application="set" data="vm_unavail_greet=0"/> <action application="set" data="vm_direct__greet=0"/> <action application="set" data="vm_ivr_enable=false"/> <action application="lua" data="vm_leave.lua 5240 lab new"/> </condition> </extension> </context> </section> </document>

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