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Conference/intercom/paging
by Jon Young 21 Sep '10

21 Sep '10

How efficient is the conference module for FS? I'm trying to understand capacities, like how many call legs in a single conference? If all but one are muted (in the conference bridge) is it still a heavy burden if 25, 50 or 100 users are in the conference? Does it mix only active talkers or all? Is there a design document or something that will help me understand conferencing better? Thanks, Jon

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Re: [Freeswitch-users] G723-Wrong ptime
by Anthony Minessale 21 Sep '10

21 Sep '10

Turn off scrooge mode Define it as g723_1@30i in the config On Sep 18, 2010 8:27 AM, "gamalsa" <gamalsa(a)hotmail.com> wrote: I don't get any message from FS, i just hear a distorted voice when using g.723. and from the debug output i see that freeswitch compares 120 ms p-time to 20 ms p-time on the other end (and chooses 120ms), while the other end is using p-time 30 ms. It is strange because g.723 uses p-time 30 ms by default. Is there any way to force FS to use g723 with p-time 30 ms with this specific provider? This problem doesn't exist when using g729 codec. I'm using the last svn of freeswitch. thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/G723-Wrong-ptime-tp5543130p55… Sent from the freeswitch-users mailing list archive at Nabble.com. ________________________________...

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Usage of optional argument -bf ?
by Patrick Lists 21 Sep '10

21 Sep '10

Hi all, In the output of freeswitch -h I noticed there is a (for me) new argument -bf that FreeSWITCH can be started with. Is -bf supposed to work in combination with -nc where it will block until it has fully started and then goes into the background? When I start FreeSWITCH (git from yesterday) as follows: /opt/freeswitch/bin/freeswitch -u freeswitch -g freeswitch -core -bf -nc then FreeSWITCH does start, goes into the background but I can not connect to it with fs_cli on the same host. When I omit -bf from the line above then all is well and I can connect with fs_cli. I'm interested in the blocking part because when you do # service freeswitch start right now it will immediately return [ OK ] but I prefer to get an ok after FreeSWITCH actually has fully started. Did I misunderstood the -bf arg? Thanks! Patrick

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ITSP inbound ok but cannot complete outbound call
by Mario 21 Sep '10

21 Sep '10

For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions! >From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <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/uid1/1${default_areacode}$1(a)sip.voipyourlife.com"/> </condition> </extension>

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Routing DID to a PSTN Number
by Aloysius Lloyd 21 Sep '10

21 Sep '10

Hello, I have the following Dial plan for Routing DID to a PSTN Number and it is working. <extension name="voipms_6471231234"> <condition field="destination_number" expression="^6471231234$"> <action application="answer"/> <action application="bridge" data="sofia/gateway/voipms/14161231234"/> <action application="hangup"/> </condition> </extension> Is there anyway to route the call without answer ( <action application="answer"/>) Any help is appreciated. Thanks Lloyd

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Freeswitch - mod_skypopen fail
by Shamun toha md 21 Sep '10

21 Sep '10

Alsa driver installation in CentOS 5.4 getting fail: [root@localhost alsa-driver-1.0.20]# ./configure --with-redhat=yes --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer --with-card-options=all checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking for ranlib... ranlib checking for a BSD-compatible install... /usr/bin/install -c checking how to run the C preprocessor... gcc -E checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for an ANSI C-conforming const... yes checking for inline... inline checking whether time.h and sys/time.h may both be included... yes checking whether gcc needs -traditional... no checking for current directory... /usr/local/src/alsa-driver-1.0.20 checking cross compile... checking for directory with kernel source... ./configure: line 4870: cd: /lib/modules/2.6.18-164.el5/source: No such file or directory /lib/modules/2.6.18-164.el5/source checking for directory with kernel build... /lib/modules/2.6.18-164.el5/build checking for kernel linux/version.h... no The file /include/linux/version.h does not exist. Please install the package with full kernel sources for your distribution or use --with-kernel=dir option to specify another directory with kernel sources (default is /lib/modules/2.6.18-164.el5/source). [root@localhost alsa-driver-1.0.20]# [root@localhost alsa-driver-1.0.20]# ./configure --with-redhat=yes --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer --with-card-options=all --with-kernel=/lib/modules/2.6.18-164.el5/source checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking for ranlib... ranlib checking for a BSD-compatible install... /usr/bin/install -c checking how to run the C preprocessor... gcc -E checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for an ANSI C-conforming const... yes checking for inline... inline checking whether time.h and sys/time.h may both be included... yes checking whether gcc needs -traditional... no checking for current directory... /usr/local/src/alsa-driver-1.0.20 checking cross compile... checking for directory with kernel source... ./configure: line 4870: cd: /lib/modules/2.6.18-164.el5/source: No such file or directory /lib/modules/2.6.18-164.el5/source checking for directory with kernel build... checking for kernel linux/version.h... no The file /include/linux/version.h does not exist. Please install the package with full kernel sources for your distribution or use --with-kernel=dir option to specify another directory with kernel sources (default is /lib/modules/2.6.18-164.el5/source).

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FS processing 302
by Chris Veazey 21 Sep '10

21 Sep '10

Freeswitch via the external profile(port 5080) is sending an Invite and receiving a 302 response. Should Freeswitch by default be able to process the maddr of a 302 redirect? SIP/2.0 302 Moved temporarily Via:SIP/2.0/UDP 10.11.0.80:5080;branch=z9hG4bKZ1H8r8c2rmHpc;rport From:"9995551000000"<sip:superadmin@sip.blinkmind.net;transport=udp>;tag=ZXj rggj4HymDe To:<sip:9995552000000@sip.blinkmind.net>;tag=1812200698-1281407077651 Call-ID:43a1c9c6-1ec9-122e-b19d-0018512d505c CSeq:367474 INVITE Contact:<sip:9995552000000@sip.blinkmind.net:5060;transport=udp;maddr=10.11. 0.81>;q=0.5;ton=PUBLIC;ct=NIL;cat=OTHER Content-Length:0 Thanks Chris

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Changing IP addresses
by Jon Young 21 Sep '10

21 Sep '10

Is there anything that would prevent changing IP addresses or subnets on a FS install? Is there a doc or wiki that points to any or all changes necessary? Regards, Jon

2 1

recording of voice of non registered users.
by sharad 21 Sep '10

21 Sep '10

Hi Is tt possible to record the voice of two IP phones who are registered with some other IPPBX & their voice is P2P. Means can FS record the rtp of non registered users ? Would really appreciate if someone could reply this. Regards Sharad

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Voice mail folders
by Tim St. Pierre 21 Sep '10

21 Sep '10

Hello, I have a lot of former Asterisk users really miffed about the lack of folders in Freeswitch voice mail. I noticed in the database table voicemail_msgs there is a folder column that is always inbox. Is this used for anything? How hard would it be to implement some sort of mechanism that would let you tag a voice mail message as being part of a specific folder, then search those folders later on for playback? Thanks! -Tim

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