January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Outbound call problem
by Joseph L. Casale 11 Jan '10

11 Jan '10

Likely an issue with my SPA3102, but when I route a call to its FXO port, I can almost faintly hear the operator if its misdialed, but otherwise the connection is loaded with feedback and static. Anyone have a suggestion on where to start looking? Inbound from that FXO port is flawless. Thanks, jlc

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lua+sqlite example?
by Steve Klein 11 Jan '10

11 Jan '10

Greetings. We are attempting to add sqlite access to an IVR application we are prototyping. We are using lua for the scripts. Is there an example anywhere of a lua + sqlite script? Do we need to install luasql? Any help/pointers greatly appreciated. --Steve Klein

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Performance issue on 1.0.4 vs 1.0.5
by itunali 11 Jan '10

11 Jan '10

Hi, I did performance tests to measure that freeswitch limits. The test just dial echo extension 9996 at default context and wait 6 sec then hangup. I used sipp test tool and set all variables/environment as described on (http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations) I reached 80 cps on 1.0.4 (install from freeswitch-drivers ppa) but not get similar rate on SVN trunk build (1.5 cps) and get many retransmission SIP packets. I build SVN trunk code with default ./bootstrap.sh && ./configure && make && make install process. My server specs; Ubuntu 9.10 Karmic Linux 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64 GNU/Linux Is there any .deb packets for 1.0.5? Regards, Ibrahim -- View this message in context: http://old.nabble.com/Performance-issue-on-1.0.4-vs-1.0.5-tp27112490p271124… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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gtalk and g723 codec
by reza mokhtarpour 11 Jan '10

11 Jan '10

Hi there I am begginer with FS , I use FS plus Gtalk everything is OK with PCMU codec but whenever i replace it whit G723 codec i got "This codec is only usable in passthrough mode!" error. SIP GW > FreeSwitch > Gtalk these are my configuration files : ------- public.xml ---------------- <extension name="Gtalk"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^1234$">  <action application="set" data="bypass_media=true"/> <action application="bridge" data="dingaling/gmail.com/xxxxx@gmail.com"/> </condition> </extension> ------------ dingaling.conf.xml ------------ <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="1"/> <param name="codec-prefs" value="G723"/> </settings> <X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/> </configuration> ---------- sip_profiles/external.xml ---------------  <param name="inbound-codec-prefs" value="G723"/> <param name="outbound-codec-prefs" value="G723"/> I was googling for a while but I got nothing. Thanks in advance.

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fifo funk?
by Gabriel Gunderson 11 Jan '10

11 Jan '10

Anyone else notice that adding and deleting members of a queue works better then reparsing with the exact same member info? It seems to keep better track of where to send the next call. Not a big deal, just wondering if there is something going on that I don't know about. All in all we've really like fifos :) Thanks FS devs! Gabe

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http://jira.freeswitch.org/browse/FSCORE-262
by Jerry Richards 11 Jan '10

11 Jan '10

Is there a plan to fix this JIRA issue: http://jira.freeswitch.org/browse/FSCORE-262 This is causing a problem in sharing presence data between FS and another gateway. Thanks, Jerry

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Codecs and things
by Ahmed Naji 11 Jan '10

11 Jan '10

Hello people, Can someone please clear the following ambiguities with codecs: 1. Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy Media mode, or does FS need to be running in bypass-media ? the Wiki is not clear in this regard 2. When an A-leg has negotiated a pass-through media codec, can the B-leg be transcoded into a non-pass-through codec, and vice-versa ? think A-leg incoming with a G.729 codec, and target for B-leg needs to be setup with a GSM-codec, say 3. Where in the developer's set of documentation are codecs discussed ? I would like to start porting some code of mine for G.729a/b/ab form a ti DSP platform to FS. FS lacking full G.729 support is proving quite a hindrance, and there is no clear direction from the dev community as to when the same will be available. Incidentally, any news on this effort ? where are we with code, and what's an ETA for a Beta ? 4. On the same lines as (3) above, there is a codec dev template in the source tree. Again, where can I find documentation relating to this ? the template has hardly any docs at all. Best regards and warm wishes for a Merry Christmas and a great New Year to one and all. Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS

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Avoiding restart of transfer_ringback file during successive bridge actions.
by mayamatakeshi 11 Jan '10

11 Jan '10

Hi, I'm implementing ACD functionality using a dialplan. Basically, I prepare a list of members in a group and try each one of them in successive bridge actions. I cannot use a single bridge action with all members separated with pipes because I am required to check some conditions like if the member is already in another call. So what I do is to add a transfer action after the bridge so that the call reenters the dialplan with the remaining of the list. Up to this point things are fine. However, if the call is answered and transferred to the group, I set transfer_ringback so that I can play a file while the bridge is happening. In case transfer_ringback is set to something like "local_stream://moh", then we hear a small glitch every time a bridge action happens as the bridge doesn't take into account that the ringback was already set up by a previous operation and sets up the playfile operation again. But this is also fine. The problem is in case transfer_ringback is set to an actual file and not local_stream. In this case, every bridge action will cause restart of the playback from the beginning. So, is it possible to ask bridge to do not setup playback if it is already running referencing the same file? regards, takeshi

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TONIGHT Join 5-6P Mon 11th - 1st Evening Meeting test IRC & VOIP online freeswitch at BerkeleyTIP-Global - for forwarding
by giovanni_re 11 Jan '10

11 Jan '10

You're invited to the first test of the Global freeswitch bimonthly evening meetings at BerkeleyTIP-Global. :) Join in tonight, Monday Jan 11, 5-6P Pacific, 8-9P Eastern, = Tues Jan 12 1A-2A UTC. http://sites.google.com/site/berkeleytip/schedule On #berkeleytip on irc.freenode.net, & on voip - whatever is working - try btip server first. http://sites.google.com/site/berkeleytip/remote-attendance This will be an online only meeting - no in person meeting at UCB. Hot topics: Community Leadership Summit review of interesting sessions, Spring 2010 efforts for UCB & all UC's & all college activities, Upcoming KDE conference end of next week, for 1 week, in Los Angeles. What do _you_ want to discuss? == Some people have asked for an evening meeting, because: a) they can't make weekend meetings, b) they want more BTIP-Global. ;) So, this will be a test, everyone invited, to see if we can make this work. == BerkeleyTIP-Global is the Global All Free SW HW & Culture meeting online via VOIP. http://sites.google.com/site/berkeleytip/ Join the global mailing list, say "hi", & what you're interested in. :) http://groups.google.com/group/BerkTIPGlobal For Forwarding: You are invited to forward this announcement wherever it might be appreciated.

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RE Help with Portech <-> Freeswitch (Max Bridgewater)
by Steven Brown 11 Jan '10

11 Jan '10

Hi, Two ways I've done it (not to say these are the 'correct' ways but they do work for me after a lot of trial and error) are either to get the Portech to register on FS as a regular endpoint then bridge to it as below <extension name="gsm"> <condition field="destination_number" expression="^9(.*)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/internal/1005^$1@${domain}"> where the Portech registers as 1005 and I prefix the number I wish to dial with 9 or alternatively just bridge directly to it <action application="bridge" data="sofia/default/$1@192.168.1.3:5060"/> Where 192.168.1.3 is my Portech In both cases you need to configure the Portech Lan to Mobile Table as below Item URL Call Num 0 192.168.1.2 # where 192.168.1.2 is my FS box This should get things going for one GSM channel, to get multiple channels running you will have to tweak it differently depending on the age of the Portech as I've discovered different firmware versions seem to behave very differently in this respect. Hope this helps Steve > Message: 1 > Date: Sat, 9 Jan 2010 12:00:54 -0500 > From: Max Bridgewater <max.bridgewater(a)gmail.com> > Subject: [Freeswitch-users] Help with Portech <-> Freeswitch > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <c55d46951001090900p36df42e3ga91e61812d5b8d03(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Guys, > > It appears quite a few people in the list are using Portech. Can you please > help me connect Freeswitch to it for termination puposes? > > Here is what I've done so far but without success. > > In Freeswitch I created a profile and stored it in under > /usr/local/freeswitch/conf/sip_profiles/external/. Here is the content: > > <include> > <gateway name="portech"> > <param name="username" value="noauthentication" /> > <param name="password" value="noauthentication" /> > <param name="proxy" value="74.24.22.59" /> > <param name="realm" value="74.24.22.59" /> > <param name="register" value="false" /> > </gateway> > </include> > > Then, in the /usr/local/freeswitch/confi/dialplan/default.xml, I added the > following extension: > > <extension name="portech"> > <condition field="destination_number" expression="^2801$"> > <action application="bridge" > data="sofia/gateway/portech/5147237479"/> > </condition> > </extension> > > In Portech MV374, what I did is simply adding one entry in the Mobile/Lan to > mobile table that consists of URL: 74.24.22.59 and call Num: #. > > Now, when I connect to Freeswitch with Xlite and try to dial extension 2801 > it simply says User Busy. i don't even see that attempts are being made to > connect to the Portech gateway. > > Any idea? > Thanks in advance. > > Max.

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