October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

problem with compiling freeswith
by Woody Dickson 06 Oct '09

06 Oct '09

Hi, Is this just me who is having this problem? I can't compile the latest freeswitch source code and here is the error: checking for gcc option to accept ANSI C... none needed checking for style of include used by make... GNU checking dependency style of gcc... gcc3 checking whether gcc and cc understand -c and -o together... yes ./configure: line 3377: syntax error near unexpected token `echo' ./configure: line 3377: `echo "$as_me:$LINENO: checking for a BSD-compatible install" >&5' configure: error: /bin/sh './configure.gnu' failed for libs/tiff-3.8.2 [root@localhost freeswitch-snapshot]# Does anyone know why? woody

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Detecting a fax
by Mark Campbell-Smith 06 Oct '09

06 Oct '09

Hi I was hoping someone could help me to setup the fax detection / tone detection application. I want to be able to transfer an incoming fax to a specific extension. In my default.xml file, I have the following (extracted): <extension name="1000"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="fax_detect"/> <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML features" /> I can't get the fax to be detected and transferred. Is there any way this can be done? Thanks!

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Re: [Freeswitch-users] UPDATED: Basic compile question.
by David Clark 05 Oct '09

05 Oct '09

Ok I found spandsp.h. It is a case of the project file being out of date. No surprise. ptlib.h is still not found. ------------------------------------------------------------------------------------------------------------------------------------------------------------ Ok using windows xp x64 here. I download from the trunk as expected. I fire up VS 2005 and I open the VS 2005 solution. Yes it does say unsupported. But I get two missing header files: freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error C1083: Cannot open include file: 'ptlib.h': No such file or directory .\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file: 'spandsp.h': No such file or directory Even if the project file is wrong or out of date I should be able to find the include files some place in the fileset. I can't find either file in the freeswitch directory or below it. Any idea what is up? Thanks, David Clark

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Connecting FS to Hicom 300
by Russell.Mosemann＠cune.org 05 Oct '09

05 Oct '09

We have connected FS to a Siemens Hicomm 300. As you might guess, it's not working right. Here is what we are working with. Dell 1750 (dual socket, dual core Xeon 2.8GHz) Debian 5 FS (15029), OpenZAP (without libpri) TE110P T1 card (Zaptel driver) Handles 71xx extensions Siemens Hicom 300 TMDN64P T1 card Handles 74xx extensions We are pretty much using the stock FS configuration, yet, because we're trying to get this to work. I have configured OpenZAP and the associated files like the examples on the wiki (see below) to work with a PRI T1. There are 23 B channels and 1 D channel. The Zaptel side looks fine. OpenZAP is able to open the channels when FS boots. So far, so good. When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta from CounterPath on an office PC), X-Lite rings. The call can be answered, and the conversation sounds fine. That means the routing, registration and authorization are working on the network between X-Lite and FS. It also means that FS is able to communicate with the Hicom over the T1. Great. When the caller presses the transfer button on the 74xx phone, the Hicom sends a message over the D channel, and the call is disconnected (watching with fs_cli). As best I can interpret the bytes in the message, the Hicom sends a disconnect message when 74xx presses the transfer key. In order to call 74xx, I created dialplan/default/02_hicom.xml. The contents are <include> <extension name="hicom"> <condition field="destination_number" expression="^(74\d{2})$"> <action application="bridge" data="openzap/1/a/$1"/> </condition> </extension> </include> If a call is made from 71xx to 74xx, the Hicom forwards the call to the switchboard with "7100->7445 connection not possible" (or whatever extensions) in the switchboard display. 1. Are these issues related to the way I have configured FS? The Hicom is maintained by the local phone company. I do not have access to view or configure the T1 card on the Hicom. According to the phone guy, there isn't anything else that needs to be configured on the Hicom. He believes that if 74xx can call 71xx, then 71xx should be able to call 74xx. I suspect that something more needs to be done on the Hicom in order to accept calls from FS and bridge/transfer them to a local extension on the Hicom. It's as if the Hicom doesn't know how or is not permitted to route incoming calls on the T1 to local extensions. I have no way to know, though. I'm hoping someone else has connected FS to a Hicom 300 and can provide configuration details. If I could tell the phone guy something like, "You need to look at <this>," that would help him out. 2. Should I receive CID/ANI from the Hicom? X-Lite displays "OpenZAP" as the call and "1" as Other when the call comes in, which is the information for the endpoint. Is there something I need to do in the FS configuration to capture CID/ANI information from the Hicom and make it available (or is it not being provided by the Hicom)? 3. When dialing from the Rolmphone is there a way for FS to send the called name back to the Hicom for it to appear in the display? When dialing 74xx to 74xx, of course, it shows the called number and name in the display. We also have a HiPath 4000 connected to the Hicom 300. When dialing an extension on the HiPath from the Hicom, the HiPath ships the called name back to the Hicom for display on the phone. It would be nice to do that from FS. Let me know if you need additional information. Thanks for any pointers or insight as to how things work. -- Russell Mosemann openzap.conf [span zt PRI_1] name => OpenZAP number => 1 trunk_type => t1 b-channel => 1-23 d-channel => 24 zt.conf [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 openzap.conf <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/>    </settings> <pri_spans> <span name="PRI_1"> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="national"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> </pri_spans> </configuration> zaptel.conf # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 # Global data loadzone = us defaultzone = us ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events!

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UUID of the newly originated call?
by Nagalenoj 05 Oct '09

05 Oct '09

Dear friends, I am trying with ESL outbound socket. I'm trying to make a call when I receive ANSWER event. Now, I would want to do something like, * Receive the events only for this uuid - I have done by registering all events and filtering only for this uuid($uuid). * If it is CHANNEL_ANSWER, originate a new call. Now, How can I get the uuid of the new call and receive events for this new call.? I want to receive the events for both uuids. -- View this message in context: http://www.nabble.com/UUID-of-the-newly-originated-call--tp25751228p2575122… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To Subscribing Phones
by Jerry Richards 05 Oct '09

05 Oct '09

By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the "manage-presence" parameter to "true" in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry

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New to freeswitch and have a few questions
by Orien Love 05 Oct '09

05 Oct '09

Hello Everybody, I am new to freeswitch, so forgive me if I ask stupid questions. I am planning a test setup consisting of: 1 - Pfsense router with the freeswitch package installed. 1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones. 1 - LINKSYS SPA3000 to connect to my existing land line and phones. 2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs The first question I have, Are the IP601 phones supported? The wiki lists 320, 431, 501, 550, 650 but not the 601. Second, is there a place that helps a person new to the IP phone world learn what is needed to set up a PBX using freeswitch at a small office? Finally is my test setup a good one? is there something I am missing or that I need to get the learning process started, I have found in the past, with a little information and a test system, I can learn what I am doing by breaking and fixing the test bed. Thanks for your time Orien

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Gateways, Limits, & Routes
by Kevin Green 05 Oct '09

05 Oct '09

It seems many people are looking for ways to control gateways, resiliency of termination, and limit on connections easily in FS. Here are some of the thoughts I had, and I would like to hear what others think of this. In tradition phone hardware you would define lines, put them into a group, and then assign a route to go through that group. For resiliency you could group multiple routes together into a route list, if the first route failed, was all busy, or unavailable it would start to try the second route. If you go out a secondary route you can also play a warning tone to indicate this might be going out a more expensive connection. For example you may have an IP link between two boxes, then fail back to TDM if the IP links go down, the TDM would be more expensive so you would want to warn business users so they don't spend hours on the phone. Inbound and outbound calls could both go on the lines so when a call comes in or goes out a route it takes up one slot. My thought is, why don't we create something similar to this that will allow us to handle a lot of these cases without complex dialplans. We could create routes that are assigned to gateways and limit the number of incoming, outgoing, and total connections that can be on that route. We could also specify what cases we would consider as failure to move onto the next route if we are in a route list. Route lists would then be similar to using multiple limits, and failover, but in this case it would simply be a list. The following is an example of how I think this could be prgrammed. <routes> <route id="route1" gateway="provider1" max-inbound="5" max-outbound="5" max-total="5" countinue_on_fail="NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/> <route id="route2" gateway="provider2" max-inbound="0" max-outbound="5" max-total="5" countinue_on_fail="NORMAL_TEMPORARY_FAILURE"/> <route id="route3" gateway="provider3" max-inbound="5" max-outbound="5" max-total="10" countinue_on_fail="NO_ROUTE_DESTINATION"/> </routes> <route_lists> <route_list id="list1"> <route id="route1"/> <route id="route2" warning_tone="expensive.wav"/> <route id="route3" warning_tone="really_expensive.wav"/> </route_list> </route_litsts> Then in your dialplan you would simply put: <extension name="outbound_rout"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="route/route2/$1"/> </condition> </extension> OR <extension name="outbound_route"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="route_list/list1/$1"/> </condition> </extension> The module would need to track not only outbound calls, but also inbound calls that come in through the specified gateways. This would help track cases where you have say only two channels that can be used for both incoming and outgoing calls. I'm not 100% sure if this is all feasable or if it would be that much of an improvement compaired to what is already there so I put it out to all of you for feedback. Regards, Kevin Green JohnnyVoIP 350 Legget Drive Kanata, ON, Canada K2K 2W7 Phone: 613 271 5993 Ext 1203 Fax: 613 271 9810

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configure FS: config.status: error: cannot find input file: Makefile.in
by Tihomir Culjaga 05 Oct '09

05 Oct '09

hello, i just got the last trunk and tried to compile it on one of my development machines... Well configure fails on tiff-3.8.2 where it is unable to find Makefile.in ... Can someone advice? checking if g++ static flag -static works... yes checking if g++ supports -c -o file.o... yes checking if g++ supports -c -o file.o... (cached) yes checking whether the g++ linker (/usr/bin/ld) supports shared libraries... yes checking dynamic linker characteristics... GNU/Linux ld.so checking how to hardcode library paths into programs... immediate checking for OpenGL Utility library... no checking for GLUT library... no configure: creating ./config.status config.status: error: cannot find input file: Makefile.in tculjaga@subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l total 2224 -rw-r--r-- 1 tculjaga tculjaga 23741 2009-10-02 13:19 acinclude.m4 -rw-r--r-- 1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4 -rwxr-xr-x 1 tculjaga tculjaga 121 2009-10-02 13:19 autogen.sh -rw-r--r-- 1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:28 config -rw-r--r-- 1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log -rwxr-xr-x 1 tculjaga tculjaga 73065 2009-10-02 14:00 config.status -rwxr-xr-x 1 tculjaga tculjaga 740145 2009-10-02 13:28 configure -rw-r--r-- 1 tculjaga tculjaga 20492 2009-10-02 13:19 configure.ac -rwxr-xr-x 1 tculjaga tculjaga 56 2009-10-02 13:19 configure.gnu -rwxr-xr-x 1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno drwxr-xr-x 16 tculjaga tculjaga 4096 2009-10-02 13:19 contrib -rw-r--r-- 1 tculjaga tculjaga 1146 2009-10-02 13:19 COPYRIGHT -rw-r--r-- 1 tculjaga tculjaga 1570 2009-10-02 13:19 HOWTO-RELEASE drwxr-xr-x 5 tculjaga tculjaga 4096 2009-10-02 13:19 html drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:28 libtiff drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 m4 -rw-r--r-- 1 tculjaga tculjaga 1908 2009-10-02 13:19 Makefile.am -rw-r--r-- 1 tculjaga tculjaga 1724 2009-10-02 13:19 Makefile.vc drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 man -rw-r--r-- 1 tculjaga tculjaga 6270 2009-10-02 13:19 nmake.opt drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 port -rw-r--r-- 1 tculjaga tculjaga 2363 2009-10-02 13:19 README -rw-r--r-- 1 tculjaga tculjaga 9 2009-10-02 13:19 RELEASE-DATE -rw-r--r-- 1 tculjaga tculjaga 5893 2009-10-02 13:19 SConstruct drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 test -rw-r--r-- 1 tculjaga tculjaga 433 2009-10-02 13:19 TODO drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 tools -rw-r--r-- 1 tculjaga tculjaga 6 2009-10-02 13:19 VERSION tculjaga@subZero:~/freeswitch-trunk/libs/tiff-3.8.2$ tculjaga@subZero:~/freeswitch-trunk/libs/tiff-3.8.2$

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Fail over
by srinivasula reddy 05 Oct '09

05 Oct '09

Hi, can any know how to implement fail over with freeswitch, please help me Regards -- Srinivasula Reddy K

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