October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch 1.0.4 problems with music on hold
by Durk de Beer 19 Oct '09

19 Oct '09

Hello, This is something I came across on Freeswitch 1.0.4 First let me explain what I'm trying to do. I want Free-Switch to behave as a proxy so in the settings section of Sofia.conf.xml I use and As fare as I am able to follow the RTP stream it is passing by Freeswitch making it a proxy. So far so good. Now I want to serv music on hold if a call is put on hold. The only way I am able to find for Freeswitch to serv music on hold (MOH) is for it to be in media. Ok we've got a command for this situation namely Ok if I put the last parram command into the Sofia.conf.xml Freeswitch is starting to behave funny. Let me try to explain this by an example. Let suppose Alice is calling Bob without the last command, the “resume media on hold”. If Bob puts Alice on hold al is silent on Bob's and Alice's phone. If Bob takes Alice of hold there's two way audio. Ok no I put the command in sofia.onf.xml, the “resume media on hold”. Now Alice is calling Bob again and Bob is putting her on hold again. You are expecting MOH on this moment but it isn't, all there is are the sounds of silence. Ok mistake made checking local streams, and they are there?. Yes it is. Ok other mistake, didn't put the “hold-music”command in sofia.conf.xml. Noop it's in there so there must be MOH. Codecs then?. Noop there in vars.xml. Ok for me it is time to throw in the towel, I'll find it out later. Ok Bob is now taking Alice of hold and surprise surprise there is only one way audio, between Alice and Bob. I didn't have the time to look with Wireshark where al the steams go so sorry for that. Did anyone came across the same problem with Freeswitch 1.0.4. I've a 1.0.2 running and that is doing everything a ok. Kind regards, Durk

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freeswitch french community
by Meftah Tayeb 19 Oct '09

19 Oct '09

hello folk we are starting a new freeswitch community for french users the newly created Irc channel is #freeswitch-fr freeswitch staf, please can we register this channel officialy to the freeswitch freenode group? also can we build a french documentation and host it somehere? in the wiki? my long time friend, diegoviola i going to help about the freeswitch wiki markup we are waiting for french users to join, and welcome thank toAnthony Minessale for accepting the community welcome! __________ Information from ESET NOD32 Antivirus, version of virus signature database 4520 (20091018) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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Outbound sip address call fails
by Lars Zeb 19 Oct '09

19 Oct '09

I am trying to make a test call to the FreeSwitch conference address, sip:888@conference.freeswitch.org on a Bria softphone. However, it fails with a 404 Not found message. I have struggled with this for a while. Is there something special I must do on the Bria to make this happen correctly? Thanks, Lars http://pastebin.freeswitch.org/10715

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macros for performing repeated dialplan actions?
by Simon J Mudd 18 Oct '09

18 Oct '09

Hi again, I'm trying to build up a soho dialplan to replicate an existing Asterisk setup I want to replace. I'm editing /usr/local/freeswitch/conf/dialplan/default/00_mydomain.com.xml to add some dialplan rules for various destination numbers I'm using. These dialplans may use one of several gateways I have configured. I currently have: <include> <extension name="spanish_mobiles.mydomain.com"> <condition field="destination_number" expression="^(6\d{8})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/asterisk/$1"/> </condition> </extension> <extension name="spanish_fijos.mydomain.com"> <condition field="destination_number" expression="^(9\d{8})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/asterisk/$1"/> </condition> </extension> ... more stuff not shown ... </include> Which routes out to my existing asterisk installation. I now want to record all calls. Looking at http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session it seems I need to add a section like into each entry prior to the application="bridge". <action application="set" data="gateway_name=XXXXX"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(C) 2009 my data here."/> <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/> <action application="set" data="RECORD_ARTIST=My name"/> <action application="set" data="RECORD_COMMENT=My comment"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${gateway_name}_${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> My question is how do I avoid repeatedly using these same 8-9 lines and "merge" this into some macro which given the right parameters allows me to define various extensions using different gateways but all configured "the same way"? Thanks for any pointers to how this should be done. Simon

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anyway to not to retry
by Juan Backson 18 Oct '09

18 Oct '09

Hi, Is there anyway to configure freeswitch so that it won't retry the SIP message when 200 OK is not received? thanks, jb

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Detecting Forwarding Device Caller ID (Off Topic)
by Max Bridgewater 18 Oct '09

18 Oct '09

Hey Guys, Pleas help me answer the following question. Sorry for being a bit off-topic. I have a GSM-VoiP modem with just 4 SIM Cards. Is there a known mechanism that allows me to use this modem for more than 4 (albeit not at the same time) GSM numbers? Let's say the phone numbers of the SIM cards inserted in the modem are call real phone numbers. My goal is to have some 10 additional virtual phone numbers served by this very same modem. One way i thought this could be done is by forwarding the virtual phone numbers the 4 real phone numbers. Then, when somebody calls one of the virtual numbers, the Telco will forward it to the real phone number and it will end on Freeswitch box. My only problem is now that on the Freeswitch box I have no idea what virtual phone number was actually dialed. Any idea how to recover this phone number? I looked at the GSM specification and couldn't find a way. PS. Purchasing DIDs won't solve my underlying problem. Thanks, Max.

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Problem connecting to Spidermonkey ODBC
by Adam Wilt 18 Oct '09

18 Oct '09

Hi, I know there are a lot of answered questions about this already, but I think I read all of the answers and I'm still stuck. * I'm using FreeSWITCH Version 1.0.trunk (14961). * I installed UnixODBC * My odbc.ini and odbcinst.ini files are in /usr/local/etc/ * I have symlinks to /usr/local/etc from /usr/etc and from /usr/local/freeswitch/etc/ * Im running freeswitch as root, and I copied odbc.ini to /root/.odbc.ini * I have mod_spidermonkey_odbc uncommented in spidermonkey.conf.xml * When I run "isql dbname" from the command prompt it works fine. But when I try to run my script, I get: 2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-10-17 23:56:21.848930 [ERR] mod_spidermonkey_odbc.c:245 Database is not connected! Please help! Thanks.

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Dingaling / Jingle DTMF support?
by Mark Campbell-Smith 18 Oct '09

18 Oct '09

Hi! I was wondering if the Dingaling implementation in FS supports DTMF? This is now supported in the jingle specs ( http://www.jabberforum.org/showthread.php?t=2709 ), even though Google Talk client does not currently support DTMF. If not, are there plans to implement this? Thanks!

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Dingaling: using a hostname instead of stun for rtp
by Mark Campbell-Smith 18 Oct '09

18 Oct '09

Hi! I have a hostname set in vars.conf.xml for the parameters external_rtp_ip and the external_sip_ip instead of the usual stun. I found that stun was timing out and was causing some problems. And as I have a hostname, it makes sense to use that instead of relying on stun. However, when I use Dingaling, I see the following error message in the dl_debug tracing: <iq to="<FS_EMAIL>@gmail.com/gtalk77BBCB94" id="344" type="error" from="<email1>@gmail.com/Talk.v10488E8134A"> <ses:session type="candidates" id="3697758817" initiator=""<email1>@gmail.com/Talk.v10488D9134A" xmlns:ses="http://www.google.com/session"> <ses:candidate name="rtp" address="host:<hostname>" port="30710" username="nVv7FqwNyWbiMuTt" password="nVv7BqwQyVbiMuTt" preference="1.0" protocol="udp" type="local" network="0" generation="0"></ses:candidate> </ses:session> <error type="modify"> <sta:bad-request xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:bad-request> <sta:text xml:lang="en" xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas">candidate has address of zero</sta:text> </error> </iq> Is Gtalk expecting an address in this field '<ses:candidate name="rtp" address="host:<hostname>"' instead of a hostname? Or should FS be sending the hostname only and not host:hostname? Thanks!

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SIP CANCEL error
by Rogelio Perez 17 Oct '09

17 Oct '09

Hi guys, I'm having a signaling issue on FS when calling from an endpoint to a GW. To reproduce the error I just call and hungup the call right away. FS gets the CANCEL and replies back with 481 Call/Transaction Does Not Exist, so the call continues and eventually connects, even when the A side of the call is closed. I have verified that the Call-ID of the CANCEL matches that from the INVITE, so I see no reasons for FS to reject it. SIP+FS logs in pastebin 10738. Any ideas? Thanks, Rogelio

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