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Rejecting undesired codecs in bypass media mode
by Victor Chukalovskiy 29 Jan '13

29 Jan '13

Hello, _Call scenario:_ call comes in with undesired codec only in SDP. (e.g. PCMA) In my sip profile I have: inbound-bypass-media="true" inbound-codec-negotiation="greedy" inbound-codec-prefs="PCMU,G729" _What happens:_ regardless of the codec in SDP, Freeswitch executes dialplan and sends the call to the B-leg. It also happens regardless of wether I use "late" or "early" negotiation _What I would like to see:_ FS rejecting the call right away if SDP does not match inbound-codec-prefs. Should not even hit the dial-plan. Any clue on how to achieve this? Many thanks, -Victor

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fs_encode and segmentation fault
by Guillermo Ruiz Camauer 29 Jan '13

29 Jan '13

I am trying the following command: fs_encode -l mod_sangoma_codec greeting.wav greeting.G729 which returns: 31663 Segmentation fault If I try to convert to PCMU it works fine: fs_encode -l mod_sangoma_codec greeting.wav greeting.PCMU Making and receiving calls with G729 works fine. A similar command on a machine with a DIGIUM card works fine: fs_encode -l mod_dahdi_codec greeting.wav greeting.G729 The only difference I see is the card and the module being loaded. Any idea of what could be happenning? Do I need to load some other module for the Sangoma card? Thanks in advance -- Guillermo

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Kill lua script from CLI
by Carlo Dimaggio 29 Jan '13

29 Jan '13

Hi all, is there any way to kill a lua script from FS CLI? I have some scripts that loop internally and I want to stop them. Regards,

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Postgres in the Core
by Andrew Cassidy 29 Jan '13

29 Jan '13

So, I've got postgres set up in the core as my main database. But what does this mean for other modules that each allow you to specify their own odbc-dns options, such as sofia, callcenter, voicemail, etc? -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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continue_on_fail does not works with limit?
by Muhammad Naseer Bhatti 29 Jan '13

29 Jan '13

I am using limit_execute to bridge a call. <action application="set" data="continue_on_fail=false"/> <action application="limit_execute" data="hash outbound first_vb03_local 1 bridge sofia/internal/19198@10.211.55.18:5080" /> <action application="limit_execute" data="hash outbound second_vb05_local 1 bridge sofia/internal/19198@10.211.55.20:5090" /> My continue_on_fail is set to false since I don't want to try the next gateway if the call is not tried because there is 1 call already first gateway. Since the call is never tried with the first gateway because of limit, there is no FAILURE from the gateway for continue_on_fail to react. Is there a way to make limit work with continue_on_fail? Or is there a better way to send multiple gateways in the dial plan where I only want to try the next gateway in the list if first gateway's limit it reached? Thanks, -- Muhammad Naseer Bhatti

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segfault in libfreeswitch.so.1.0.0
by Mimiko 29 Jan '13

29 Jan '13

Hello. Randomly I get this in kern.log: kernel: freeswitch[3136]: segfault at 8d00000081 ip 00007fbe75c90130 sp 00007fbe705c66d0 error 4 in libfreeswitch.so.1.0.0[7fbe75b68000+277000] Its so random that I can't find why. Usually this happen once a day at random time. There are days when this does not happen. echo 1 > /proc/sys/fs/suid_dumpable ulimit -c unlimited As per wiki, core files can not be located. Ulimit is enabled before starting freeswitch. But as FS is started under user from basedir directory, and that user does not have write permission to this directory, therefore no core files are created. The FS version is: FreeSWITCH Version 1.3.8b+git~20121130T003427Z~b35ac6b3bd (git b35ac6b 2012-11-30 00:34:27Z) Maybe something was changed from that version. In jira changelog I didn't see any related to this. How to track this? Is there a variable to set for core files directory dump? Thank you. -- Mimiko desu.

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[mod_sofia] Is the SIP Presence (SIMPLE) functionality limited to 2 states on FreeSWITCH?
by kkgp20＠gmail.com 29 Jan '13

29 Jan '13

Good morning, I am trying to understand and make working the SIP Presence functionality (SIMPLE) with the FreeSWITCH server. I have noticed that FreeSWITCH seems to support only two states: registered and not registered (online, offline). I have installed two SIP Clients: PortGO and Zoiper Communicator, both have to functionality to set a user status like: busy, away, in a call, on the lunch etc. By the Wireshark software I have seen that changing the user status on PortGO does not cause to send any SIP NOTIFY or PUBLISH messages - which seems to be a bug, but the Zoiper was doing it. So then I have noticed that when I have changed the user state on one of the clients, the second client all the time was receiving the NOTIFY message from FreeSWITCH with "Available" status (no matter which state I have chosen, despite offline). Only in case when I have switched to offline, the client have received "Not registered" message. My question is if it is possible to configure the FreeSWITCH server to support other states? It means any text which is going inside the <dm:note/> XML tag? Because now it seems that this tag is ignored and FreeSWITCH is sending the Available. Thank you for help.

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Direcotry Status with mod_xml_curl
by mehroz 29 Jan '13

29 Jan '13

Hi, I have been working with asterisk previously, recently came over Freeswitch platform. I dig up for the similar fucntionality of asteriks-realtime and found MOD_XML_CURL as the nearest match for dynamic directory(extension) creation. One good thing about asterisk-realtime was, it was compact in a single table and direct conversation with Asterisk and DB is possible , whereas in Freeswitch its through Webserver and curl requests. Watever, i accepted that! but one more thing i am missing is the status of SIP Registration (registered or not). Again, in asterisk realtime i can see the RTP port assigned to extension when it registers, this way i can see the current status reading the DB , and port number. Now the question is , how can i see this with mod_xml_curl ? and also, if any one can help me with the above mentioned grudges and crushes! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Direcotry-Status-with-mod-xml… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Remove inband DTMF on record_session
by Tamas Jalsovszky 29 Jan '13

29 Jan '13

Hello, Is there a way to to remove inbad DTMF from the recorded wav file (conditionaly)? If not, how could be that done the easyiest way and a way it could get into FS git? As FS has a great DTMF detector and it has probably all the stuff doing such a feature, I think it would be much better than using an external tool for removing DTMF from finished wav file - and it would consume more I/O probably. Regards, Jalsot

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Help with originating calls
by Gary Foreman 29 Jan '13

29 Jan '13

Hi, I'm very new to Freeswitch and having a bit of difficulty with the originate command. Essentially I'm using the CLI to originate a call however the process seems to be that the originating extension is first called, then once answered the actual call takes place. My application will take care of whether the agent is available so I want to skip the first part and have the extension dial without having to be called first, this is the way the Avaya systems I have experience with work. I've looked at the sip_auto_answer option however my sip client doesn't support auto answer. I'm just not sure on the correct way of achieving what I'm looking to do, maybe some kind of bridge would be better? Thanks in advance for your help

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