January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

unable to dial out troug siemens Hie9200 softswitch
by Meftah Tayeb 13 Jan '10

13 Jan '10

hi dear all, we have a siemens HIE 9200 softswitch that support SIP, SIP-T and H.248 we are trying to interconnect freeswitch with it if HIE9200 dial out troug freeswitch, is passing the call, but without RTP but if the freeswitch dial out troug the HIE9200, the call is unable to pass with error 500 the trace is here: http://siplabs.net/tracebin/fs-siemens-500.pcap http://siplabs.net/tracebin/fs-siemens-rtp.pcap unfortunatly the freeswitch don't support H.248 otherwise i will control the media gateway (HIG 1100) thanks for any help

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Hello , about calculate numbers
by Chia-Yen Wu 13 Jan '10

13 Jan '10

Hello everyone, Is there anyway to calculate number in dialplan.xml? like example: <action application="set" data="var=1"/> and with an action , I can do : var=var+1 I have tried mod_expr, but failed to catch the variable, like below: <action application="set" data="var=${expr($var+1)}"/>

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Bypass_media mode
by Mark Campbell-Smith 13 Jan '10

13 Jan '10

Hi! Hi! I am calling from extension 2010 to extension 1000. Both have ip addesses 192.168.1.x. In the 2000 series dialplan (a separate context) I have the following to try to enable bypass_media. <extension name="Local_1000_9xxx"> <condition field="destination_number" expression="^(10[01][0-9]|9\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=true"/> <action application="transfer" data="${dialed_extension} XML default"/> </condition> </extension> Is this how bypass media should be enabled? This fails fo me (the calls hang up and no audio). The debug trace is in http://pastebin.freeswitch.org/11737 What have I done wrong? Thanks

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Defunct process in ESL testserver example
by MohammedShehzad 13 Jan '10

13 Jan '10

Hello everybody, I am creating a C program of ESL outbound for call processing. I am using testserver.c example, and till now it seems fine. But i noticed that every call testserver process, a new process is being created which I can see in Linux system with below command: For example, when I make two calls and even after hangup, I saw three process like below: ps -A | grep testserver 9345 pts/2 00:00:00 testserver 9350 pts/2 00:00:00 testserver 9357 pts/2 00:00:00 testserver This get increased for every call i make. I did some workout and placed below two lines (close & exit) at the end of mycallback function, (as I found them on ivrd.c file): esl_disconnect(&handle); close(client_sock); exit(0); } But after that the process becomes defunct/zombie 9440 pts/2 00:00:00 conflisten 9442 pts/2 00:00:00 conflisten <defunct> 9452 pts/2 00:00:00 conflisten <defunct> Will anybody please suggest me how can I eliminate this process, which remains in memory even after call hangup? Thanks for any response. MohammedShehzad

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ShoreTel to FS connection
by John 13 Jan '10

13 Jan '10

I am trying to utilize a freeswitch system as a SIP trunk for ShoreTel. I have had to change the port from 5080 to 5060. When I dial through, I see on the FS console show the call; however I can't seem to change the public.xml dialplan to effect any changes in the behaviour. How can I tell which dialplan the external call is using? 2010-01-12 13:04:17.705548 [NOTICE] switch_channel.c:613 New Channel sofia/external/+19705551212@192.168.155.13:5060 [aa2f2fde-ffb5-11de-9590-015cd8b8c273] 2010-01-12 13:04:17.705548 [INFO] mod_dialplan_xml.c:408 Processing Extension 1002->5551212 in context public 2010-01-12 13:04:17.705548 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/external/+19705551212@192.168.155.13:5060 [CS_EXECUTE] [NORMAL_CLEARING Thanks John

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Channel Variables with spaces
by Brian Engert 13 Jan '10

13 Jan '10

I'm trying to set a channel variable with a space in it when I run the originate command. The command only seems to work when I escape the space with a \ however that backspace get's passed on. Some examples are originate {fax_ident=1231231234,fax_header=testing\ spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff) originate {fax_ident=1231231234,fax_header=testing spaces}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff) The first one will go through just fine but the fax at the other end gets "testing\ spaces" the second one gives me "2010-01-12 16:35:18.41325 [ERR] switch_ivr_originate.c:990 Parse Error!" I think this could also be done as a profile but I would rather do it from the call. I've looked around the wiki and have not seen any examples that use a space when doing the originate call. - Brian

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PRI Goes Down Upon Calling Cell Phone
by Jerry Richards 12 Jan '10

12 Jan '10

I am having a problem where my PRI goes down after attempting to call a cell phone from an internal phone. I am running Freeswitch (freeswitch-1.0.5-20100112-0400.tar.gz) with an A101D Sangoma Card and wanpipe driver: wanpipe-3.5.10.tgz. I am not using libpri. The scenario is as follows: 1) Call desktop phone 3491006 2) Call completes normally 3) Call cell phone 4181432 (see http://pastebin.freeswitch.org/11767) ***** call fails; cell phone rings, but does not complete ***** 4) Call desktop phone 3491006 ***** call fails; far-end does not ring ***** 5) Any subsequent calls through PRI fail I notice the call to my cell phone receives a Q.931 PROGRESS message instead of Q.931 ALERTING. Has this issue already been identified as a bug? Do you know if this is a FS issue? Or Wanpipe driver issue? Best Regards, Jerry

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Cisco SPA5xx SCA
by Peder 12 Jan '10

12 Jan '10

Anybody know the settings on the SPA5xx to make SCA work? I've tried all sorts of combinations of shared/private on the ext and phone and set the Server type to Broadsoft, etc and it just never seems to work. Do I need the BLF info? Do I set shared/private on the phone, or ext, or both? Do I need to enable SCA Line ID Mapping? Or SCA Barge-In Enable? Or both? Peder

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How to provide dynamic directory information?
by Mike van Lammeren 12 Jan '10

12 Jan '10

Hello! I'd like to be able to have FreeSWITCH check a database for authorization, every time a user registers. There are some great examples on the wiki, which use either MOD_XML_CURL or Lua to dynamically provide a dialplan, but I cannot find an example for providing a directory. I am developing an application that will have thousands of users, and will run on multiple FreeSWITCH servers behind a load balancer. Ideally, FreeSWITCH would only look-up directory information, specifically, username and password, whenever a user attempts to connect. The directory information will be changing regularly, as users are added or removed from the system. Is this possible with FreeSWITCH? Or can only dialplan information be provided dynamically? I've written a script in Lua that provides the XML data, such as that found in the example /freeswitch/conf/directory/default/ folders, and I try to call it with this bit of XML in /freeswitch/conf/directory/default.xml: <groups> <group name="default"> <users> <param name="xml-handler-script" value="/directory.lua"/> <param name="xml-handler-bindings" value="users"/> </users> </groups> Is this the right approach? Am I going about this the right way? I would appreciate any tips that anyone can provide! Thanks! Mike van Lammeren

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Fax Codecs
by Joseph L. Casale 12 Jan '10

12 Jan '10

While faxing is working, I am seeing these in the log when I attempt to use rxfax: switch_ivr_play_say.c:1154 Codec Activated L16@8000hz 1 channels 30ms switch_core_io.c:652 sofia/internal/1002(a)192.168.13.1 receive message [TRANSCODING_NECESSARY] So later I see: switch_core_codec.c:128 sofia/internal/1002(a)192.168.13.1 Restore previous codec PCMU:0 So I am actually not clear on what's happening? I thought I had the ua set to PCMU but I guess fs is seeing the incoming sip session as L16 which I read is bad for fax? What's the significance of the last line where it states its restoring the old codec? Can anyone shed some light on what's actually happening? Thanks! jlc

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