[Freeswitch-users] internal external ip addresses of freeswitch
Hi. We have a local network 192.168.1.0/24, where all the users are. Out FreeSWITCH server is connected to this network, and has ip address 192.168.1.242. Through different network card it is connected to external gateway, and has address 172.16.12.11 in this network. I set up a test client with softphone. When incoming call is deliviered to this client, call is set up normally, but client can't hang it up. It sends BYE to external address - 172.16.12.11 - which is not reachable from the client. It seems this address is coming from Contact: field in INVITE that FreeSWITCH sends: U 192.168.1.242:5060 - 192.168.1.34:37169 INVITE sip:1...@192.168.1.34:37169 SIP/2.0. Via: SIP/2.0/UDP 172.16.12.11;rport;branch=z9hG4bKrvp6jm3myyaaF. Max-Forwards: 70. From: FreeSWITCH sip:000...@172.16.12.11;tag=v817pS9c6v6Fe. To: sip:1...@192.168.1.34:37169. Call-ID: 797bd088-29cd-122d-9b93-0060979d54c5. CSeq: 121117089 INVITE. Contact: sip:mod_so...@172.16.12.11:5060. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14898. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 267. Remote-Party-ID: FreeSWITCH sip:000...@172.16.12.11;party=calling;screen=yes;privacy=off. What should I tweak in freeswitch to change this behaviour? -- Timur Irmatov, xmpp:irma...@jabber.ru ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference participant from behind NAT
Hi folks! Suddenly I found this http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.htmltopic and that explains a lot. From there I see that sofia sends refresher messages for NATed client in order to check if it still alive. It means I have problems in my client. Sorry for the mess. Cheers, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
Anyone have this issue? On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I have just started to use dingaling again, and noticed I constantly get a stun error. 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed! stun.fwdnet.net:3478 [Remote Address Error!] I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers and keep getting this error with dingaling. I have no problems with inbound sip calls, so I don't think its the actual stun server. Has anyone else seen this? I am using: FreeSWITCH Version 1.0.trunk (14952) Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
Yes, i had same problem, then i changed stun server to one of our own servers. You may try some of public stun servers listed on below link, http://www.voip-info.org/wiki/view/STUN Thank you. On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Anyone have this issue? On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I have just started to use dingaling again, and noticed I constantly get a stun error. 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed! stun.fwdnet.net:3478 [Remote Address Error!] I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers and keep getting this error with dingaling. I have no problems with inbound sip calls, so I don't think its the actual stun server. Has anyone else seen this? I am using: FreeSWITCH Version 1.0.trunk (14952) Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?
You can use the api and check that the channel is occupied with show channels? You can write a small javascript that checks if the channel is occupied by means of session.execute api. /aep -- Stopping junk mailers is good for the environment My SIP provider allows only one call (incoming or outgoing) via one SIP account. For FreeSWITCH I have configured it as public DID extension and outgoing gateway. Now I would like to transfer to another gw (or generate limit exceded) when one tries to place an outgoing call while incoming call is in progress. How tho do that? Limiting the number of outgoing calls is easy (mod_limit), but how to take into account incoming one? - Dmitry Bely ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
Michael Collins m...@freeswitch.org said: I believe the OpenZAP and 1 are coming from your conf file: openzap.conf [span zt PRI_1] name = OpenZAP number = 1 That is correct. If that information is removed, then X-Lite displays FreeSWITCH [Other: 00] Are there any variables to set to get CID, or is OpenZap supposed to be filling that in? As far as debugging with ozmod_libpri I believe the syntax is: oz libpri debug 1 all That works. Now, I have to figure out what some of these abbreviations mean. -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] configure FS: config.status: error: cannot find input file: Makefile.in
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 tculj...@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 tculjaga121 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 tculjaga433 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 tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?
what if you are running some huge traffic e.g. 2000 calls with media?
a typical application for that is an IVR system handling several different
services. I'd like to dedicate some capacity for inbound on per service
basis.
e.g.
DID 10001 limit to 500 calls
DID 10002 limit to 400 calls
DID 10003 limit to 100 calls
DID 10005 limit to 1000 calls
This will be a total of 2000 calls.
don't you think js is simply too weak for that? It should cont
calls/channels, brake counts per service/DID and update the counters on
every call hit.
in the DP you would have something like this for every DID:
include
extension name=MY_DID_NUM
condition field=destination_number expression=^MY_DID_NUMBER$
action application=set data=SERVICE_LIMIT=500/
!--
count number of active channels going towards MY_DID_NUMBER and
store it into COUNT_MY_DID_NUMBER
--
action application=transfer data=do_MY_SERVICE XML public/
/condition
/extension
/include
include
extension name=SERVICE1
condition field=destination_number expression=^do_MY_SERVICE$/
condition field=${COUNT_MY_DID_NUMBER} expression=^SERVICE_LIMIT$
!-- do your service here --
action application=playback data=I_Accept_Your_Call.wav/
action application=hangup data=NORMAL_CLEARING/
!-- do your limitation here --
anti-action application=respond data=403 Forbidden/ = put your
response here!
/condition
/extension
/include
but the question is ... how powerful a JavaScript can be? Will it be enough
to handle that load?
Tihomir.
On Fri, Oct 2, 2009 at 1:11 PM, Alberto Escudero aep.li...@it46.se wrote:
You can use the api and check that the channel is occupied with show
channels?
You can write a small javascript that checks if the channel is occupied by
means of session.execute api.
/aep
--
Stopping junk mailers is good for the environment
My SIP provider allows only one call (incoming or outgoing) via one
SIP account. For FreeSWITCH I have configured it as public DID
extension and outgoing gateway. Now I would like to transfer to
another gw (or generate limit exceded) when one tries to place an
outgoing call while incoming call is in progress. How tho do that?
Limiting the number of outgoing calls is easy (mod_limit), but how to
take into account incoming one?
- Dmitry Bely
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Re: [Freeswitch-users] Dialplan Issue
anyhow, this is how it works for me!
include
context name=public
extension name=LNP
condition field=destination_number
expression=(^30)(.*)
action application=lnp_getprefix data=in $2, out
reroutingalias/
action application=redirect data=sip:${
reroutingali...@10.4.13.11:5060/
/condition
/extension
extension name=LBS
condition field=destination_number
expression=(^300010)(.*)
action application=lbs_getpublicphone data=in
${caller_id_number}, in $2, out reroutingalias/
action application=redirect data=sip:${
reroutingali...@10.4.13.11:5060/
/condition
/extension
extension name=CPS
condition field=destination_number
expression=(^300020)(.*)
action application=cps_verifyphone data=in
${caller_id_number}, in $2, out radiusacc/
/condition
condition field=radiusacc expression=1
action application=redirect data=sip:${
caller_id_numb...@10.4.13.11:5060/
anti-action application=respond data=403
Forbidden/
/condition
/extension
extension name=ServiceLookup
condition field=destination_number expression=(^300030)(.*)
action application=lookup_service_destination data=in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${red_contact}' ##\n/
action application=log data=INFO
CallerNum = '${caller_id_number:6:16}' ##\n/
action application=log data=INFO
RADIUS auth = '${authResult}' ##\n/
action application=execute_extension data=doRedirect XML
public/
/condition
/extension
extension name=doRedirect
condition field=destination_number expression=^doRedirect$/
condition field=${authResult} expression=^0$|^60$
action application=log data=INFO
RADIUS auth OK!!!' ##\n/
action application=redirect data=${red_contact}/
anti-action application=log data=INFO
RADIUS auth NOK!! ##\n/
anti-action application=respond data=403 Forbidden/
/condition
/extension
/context
/include
On Thu, Oct 1, 2009 at 6:18 PM, Shelby Ramsey sicfsl...@gmail.com wrote:
Just to confirm ... works like a champ.
Thanks again!!!
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[Freeswitch-users] New to freeswitch and have a few questions
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In!
Hey folks, the weekly conference call is starting. Please see the agenda for instructions on dialing: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_02 Looking forward to speaking with you all! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
On Fri, Oct 2, 2009 at 4:48 AM, russell.mosem...@cune.org wrote: Michael Collins m...@freeswitch.org said: I believe the OpenZAP and 1 are coming from your conf file: openzap.conf [span zt PRI_1] name = OpenZAP number = 1 That is correct. If that information is removed, then X-Lite displays FreeSWITCH [Other: 00] do something like: name = XYZ Corp number = 8005551212 Are there any variables to set to get CID, or is OpenZap supposed to be filling that in? As far as debugging with ozmod_libpri I believe the syntax is: oz libpri debug 1 all That works. Now, I have to figure out what some of these abbreviations mean. Welcome to the wacky world of Q931. The wiki has info: http://wiki.freeswitch.org/wiki/ISDN:_Integrated_Services_Digital_Network -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Call Forward All/Busy/No-Answer
How would I configure FS to Call Forward All or Call Forward when Busy or Call Forward when No-Answer? Can this be done at the server, rather than at the phone? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
I only mentioned the OS I used as a reference for people. If they want to do the same thing on another OS, then they might not have apt-get, etc. Mike van Lammeren On Thu, Oct 1, 2009 at 3:41 PM, Hadley Rich h...@nice.net.nz wrote: On Thu, 2009-10-01 at 14:14 -0500, Anthony Minessale wrote: can we do it without advertising to use ubuntu =D We don't like encouraging our users to use bleeding edge OS for our own sanity with debugging. I understand your stance, though if we're talking about Ubuntu 8.04 LTS (Long Term Support - 5 years) it's not really bleeding edge anymore. 18 months ago when it was released it may have been a little, but the LTS releases still aren't as bleeding edge as the standard support in between releases. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
The load balancer listens to the virtual IP address, and port-forwards to one of the FreeSWITCH boxes. Each FreeSWITCH box listens for the same virtual IP address for SIP registrations and connections, which is what FreeSWITCH needs to bind to. All other traffic actually travels over their real IP address, which is what the FreeSWITCH servers would use to talk to each other. On Thu, Oct 1, 2009 at 3:44 PM, Peder pe...@networkoblivion.com wrote: Looking thru the example, it looks like each box has a real address of 21, 22 or 23 and they all have a loopback of .17, right? So even though they connections are being load balanced, each box really thinks it is .17 and each client that connects thinks it is connecting to .17, right? If that’s the case, how does a client on one box call a client on the other box? Since every box thinks it is .17 how would you bridge to another user on another box that also thinks it is .17? Or am I totally missing how it works? *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Thursday, October 01, 2009 2:14 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey can we do it without advertising to use ubuntu =D We don't like encouraging our users to use bleeding edge OS for our own sanity with debugging. Not to say you are not allowed to I just don't want to encourage it =p On Thu, Oct 1, 2009 at 1:12 PM, Even André Fiskvik grev...@me.com wrote: That's very cool Mike! I'm going to try to configure four boxes with this as well (Btw, did you use physical hardware or virtualization?) and see how it goes. I followed Daniel Aliaman's blog as well, but I can try it again with the tips you provided on FreeSWITCH config to see if I can get it working properly this time. We did the setup on CentOS, but I wouldn't think that would be any issue. Perhaps you or we could write up a complete guide about this on the wiki since this is an scenario commonly used? Also it would be great if we could outline possible issues (and even better solutions) to this kind of setup with regards to stuff like conferencing, bridging between registered users and presence. Best regards, Even André On 1. okt. 2009, at 18.45, Mike van Lammeren wrote: Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using heartbeat and ldirectord for load-balancing, fail-over and high availability! I'm probably not the first one to do it, but as near as Google and I can tell, I'm the first one to write about it. Here's how you can duplicate my setup: 1. Install Ubuntu Server 8 on four machines, either real or VM. 2. Compile and install FreeSWITCH v1.0.4 from source on two machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start 3. Configure both FreeSWITCH boxes, and make sure they are both working. 4. Follow (most of) these instructions from Daniel Aliaman's blog. They were written for Asterisk, but since a SIP connection is a SIP connection, most of the document applies to FreeSWITCH: http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf The one problem I ran into was the IP address and port to which FreeSWITCH was bound. The default is to use the primary address, which works great out-of-the-box for everything else. When a client tried to register, all it got back was an ICMP error -- Destination Unreachable, Port Unreachable. That error is returned when no sockets are listening for UDP packets. To get FreeSWITCH to listen for your Virtual IP, you need to set it in two places: 5. In /opt/freeswitch/conf/vars.xml, set bind_server_ip. 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set sip-ip. That should do it! If you have any success, please report to this list. Keep in mind that if you want to do something like conferencing between two registered clients, then you have to deal with the fact that the clients may or may not be on the same box. Mike van Lammeren On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren m...@van.lammeren.net wrote: On Mon, Sep 28, 2009 at 9:05 PM, Even André Fiskvik grev...@me.com wrote: From: Even André Fiskvik grev...@me.com To: freeswitch-users@lists.freeswitch.org Date: Mon, 28 Sep 2009 22:52:13 +0200 Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey I have been working with a similar setup myself, but for some reason I ended up ditching the UltraMonkey setup because I just couldn't get it to work right. It's been quite a while since my effort, so I don't remember what the exact issue was. I got registrations to work, but had some other sip-dialog issues. We have since then changed over to running OpenSIPs as a loadbalancer in front of multiple FreeSWITCH instances. This
[Freeswitch-users] looking for qualified and cheap TISP
Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial out / dial in. Could anyone suggest one Telephone Service provider which is capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At this moment, I want to prove it is working with the real outside world. Thanks, e ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
I am running the servers on the free version of VMware's ESX platform, but only for development purposes. We will be setting up real machines sometime in Spring 2010. On Thu, Oct 1, 2009 at 2:12 PM, Even André Fiskvik grev...@me.com wrote: That's very cool Mike! I'm going to try to configure four boxes with this as well (Btw, did you use physical hardware or virtualization?) and see how it goes. I followed Daniel Aliaman's blog as well, but I can try it again with the tips you provided on FreeSWITCH config to see if I can get it working properly this time. We did the setup on CentOS, but I wouldn't think that would be any issue. Perhaps you or we could write up a complete guide about this on the wiki since this is an scenario commonly used? Also it would be great if we could outline possible issues (and even better solutions) to this kind of setup with regards to stuff like conferencing, bridging between registered users and presence. Best regards, Even André On 1. okt. 2009, at 18.45, Mike van Lammeren wrote: Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using heartbeat and ldirectord for load-balancing, fail-over and high availability! I'm probably not the first one to do it, but as near as Google and I can tell, I'm the first one to write about it. Here's how you can duplicate my setup: 1. Install Ubuntu Server 8 on four machines, either real or VM. 2. Compile and install FreeSWITCH v1.0.4 from source on two machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start3. Configure both FreeSWITCH boxes, and make sure they are both working. 4. Follow (most of) these instructions from Daniel Aliaman's blog. They were written for Asterisk, but since a SIP connection is a SIP connection, most of the document applies to FreeSWITCH: http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdfThe one problem I ran into was the IP address and port to which FreeSWITCH was bound. The default is to use the primary address, which works great out-of-the-box for everything else. When a client tried to register, all it got back was an ICMP error -- Destination Unreachable, Port Unreachable. That error is returned when no sockets are listening for UDP packets. To get FreeSWITCH to listen for your Virtual IP, you need to set it in two places: 5. In /opt/freeswitch/conf/vars.xml, set bind_server_ip. 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set sip-ip. That should do it! If you have any success, please report to this list. Keep in mind that if you want to do something like conferencing between two registered clients, then you have to deal with the fact that the clients may or may not be on the same box. Mike van Lammeren On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren m...@van.lammeren.net wrote: On Mon, Sep 28, 2009 at 9:05 PM, Even André Fiskvik grev...@me.com wrote: From: Even André Fiskvik grev...@me.com To: freeswitch-users@lists.freeswitch.org Date: Mon, 28 Sep 2009 22:52:13 +0200 Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey I have been working with a similar setup myself, but for some reason I ended up ditching theUltraMonkey setup because I just couldn't get it to work right. It's been quite a while since my effort, so I don't remember what the exact issue was. I got registrations to work, but had some other sip-dialog issues. We have since then changed over to running OpenSIPs as a loadbalancer in front of multiple FreeSWITCH instances. This setup is still in testing, but seemlingy works fine (and if it doesn't, it's my own fault for writing a bad opensips config). After we have done some more testing I can create a wiki-page with config details. Best regards, Even André Thanks, Even, that would be great! I might have to give up on the ultramonkey solution, since I can't find anyone who has made it work. It's too bad, because it would fit well with the rest of our architecture. Mike van Lammeren ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones
Okay, I put a log up on the pastebin that shows the PUBLISH event coming from a CounterPath Bria Professional phone. For some reason, FS is getting an error and not relaying the presence status to the subscriber. Best Regards, Jerry _ From: João Mesquita [mailto:jmesqu...@freeswitch.org] Sent: Thursday, October 01, 2009 8:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.com wrote: If you have time to take a look, I could put a trace in the pastebin? Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, October 01, 2009 10:29 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: 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.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
Michael Collins m...@freeswitch.org said: do something like: name = XYZ Corp number = 8005551212 I was expecting that information to be filled with the caller name and number. It doesn't really help if someone calls from outside the business, and it looks like my business is calling me. Doesn't OpenZAP extract caller information from a PRI T1? -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] looking for qualified and cheap TISP
Hey Erwin, Can't give any personal recommendations, but on the FS site, there's several examples. Some have free or cheap in the name. Might be a good place to start, plus the means to connect is demonstrated to-boot. http://wiki.freeswitch.org/wiki/SIP_Provider_Examples Regards, Mike G. On Fri, Oct 2, 2009 at 10:50 AM, Erwin Davis davis.er...@gmail.com wrote: Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial out / dial in. Could anyone suggest one Telephone Service provider which is capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At this moment, I want to prove it is working with the real outside world. Thanks, ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] looking for qualified and cheap TISP
Hello! For dialing in, there are a number of sites that provide free DIDs, such as http://freephonelines.ca/ . For dialing out, you can get 1.5 cents per minute calling to N. America from http://les.net/ . Mike On 2009-10-02, at 11:50 AM, Erwin Davis davis.er...@gmail.com wrote: Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial out / dial in. Could anyone suggest one Telephone Service provider which is capable of connecting with FreeSWITCH and CHEAP/ even FREE if possible? At this moment, I want to prove it is working with the real outside world. Thanks, e ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones
connect to sqlite directly with sqlite3 app and try that sql stmt and see why it doesn't match anything. sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away',' 192.168.72.38',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name = 'external' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards jerry.richa...@teotech.comwrote: Okay, I put a log up on the pastebin that shows the PUBLISH event coming from a CounterPath Bria Professional phone. For some reason, FS is getting an error and not relaying the presence status to the subscriber. Best Regards, Jerry -- *From:* João Mesquita [mailto:jmesqu...@freeswitch.org] *Sent:* Thursday, October 01, 2009 8:14 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.com wrote: If you have time to take a look, I could put a trace in the pastebin? Jerry -- *From:* Jerry Richards [mailto:jerry.richa...@teotech.com] *Sent:* Thursday, October 01, 2009 10:29 AM *To:* 'freeswitch-users@lists.freeswitch.org' *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry -- *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com] *Sent:* Thursday, October 01, 2009 9:36 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: 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.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
Re: [Freeswitch-users] looking for qualified and cheap TISP
Hi! Callcentric http://wiki.freeswitch.org/wiki/Provider_Configuration:_Callcentric offers a package called IP Freedom http://www.callcentric.com/rate_plans01.php. It costs nothing and will allow you to test FS. Carlos Erwin Davis wrote: Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial out / dial in. Could anyone suggest one Telephone Service provider which is capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At this moment, I want to prove it is working with the real outside world. Thanks, ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
On Fri, Oct 2, 2009 at 10:24 AM, russell.mosem...@cune.org wrote: Michael Collins m...@freeswitch.org said: do something like: name = XYZ Corp number = 8005551212 I was expecting that information to be filled with the caller name and number. It doesn't really help if someone calls from outside the business, and it looks like my business is calling me. Doesn't OpenZAP extract caller information from a PRI T1? Can you pastebin a dialplan snippet (or put it here) so I can see what you're doing? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones
I put the sqlite3 select query in the paste bin, and prior to that, I entered the .dump command. The select command came back with a ... prompt which I don't understand. I don't know enough about sqlite3 to know what that means? Best Regards, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, October 02, 2009 10:52 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones connect to sqlite directly with sqlite3 app and try that sql stmt and see why it doesn't match anything. sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name, 'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name = 'external' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards jerry.richa...@teotech.com wrote: Okay, I put a log up on the pastebin that shows the PUBLISH event coming from a CounterPath Bria Professional phone. For some reason, FS is getting an error and not relaying the presence status to the subscriber. Best Regards, Jerry _ From: João Mesquita [mailto:jmesqu...@freeswitch.org] Sent: Thursday, October 01, 2009 8:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.com wrote: If you have time to take a look, I could put a trace in the pastebin? Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, October 01, 2009 10:29 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: 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.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch
Re: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones
You are missing the trailing ; On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards jerry.richa...@teotech.com wrote: I put the sqlite3 select query in the paste bin, and prior to that, I entered the .dump command. The select command came back with a ... prompt which I don't understand. I don't know enough about sqlite3 to know what that means? Best Regards, Jerry -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
Can you pastebin a dialplan snippet (or put it here) so I can see what
you're doing?
-MC
It is the stock FS configuration with a small change. We're still testing
things, getting them to work. This is from public.xml. It detects calls to
internal 71xx extensions and transfers them. The transfer works. Do some
additional variables need to be set here?
extension name=public_extensions
condition field=destination_number expression=^(10[01][0-9]|71\d{2})$
action application=transfer data=$1 XML default/
/condition
/extension
--
Russell Mosemann
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[Freeswitch-users] Asterisk vs Freeswitch
Is there benchmark test results on how many simultaneous calls Freeswtich can do (with RTP anchored through it) vs the Asterisk. For any hardware/CPU/Mem that anyone may have performed this performance testing. Any numbers on average how much Freeswitch scores over the Asterisk in terms of capacity will help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
On Fri, Oct 2, 2009 at 11:53 AM, Russell Mosemann russell.mosem...@cune.org
wrote:
Can you pastebin a dialplan snippet (or put it here) so I can see what
you're doing?
-MC
It is the stock FS configuration with a small change. We're still testing
things, getting them to work. This is from public.xml. It detects calls to
internal 71xx extensions and transfers them. The transfer works. Do some
additional variables need to be set here?
extension name=public_extensions
condition field=destination_number
expression=^(10[01][0-9]|71\d{2})$
action application=transfer data=$1 XML default/
/condition
/extension
cool. can you pastebin a debug log on an incoming call?
-MC
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Re: [Freeswitch-users] Asterisk vs Freeswitch
Hi, for example here: http://blogs.zdnet.com/Greenfield/?p=214 We *replaced* a cluster of *10 Asterisk* servers with a *single FreeSwitch*server, said Chris Parker, director of systems for a large publicly traded CLEC. Parker says hes getting several hundred concurrent calls on a single, dual-core box thats also doing all of the media processing, a computationally intensive task. -- Best regards, Dmitry Kadantsev http://www.kadantsev.com - Home page (MS Silverlight required) http://www.doxwox.com - Best web meeting and online collaboration tool On Fri, Oct 2, 2009 at 9:10 PM, Ujjval Karihaloo ujj...@simplesignal.comwrote: Is there benchmark test results on how many simultaneous calls Freeswtich can do (with RTP anchored through it) vs the Asterisk. For any hardware/CPU/Mem that anyone may have performed this performance testing. Any numbers on average how much Freeswitch scores over the Asterisk in terms of capacity will help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
cool. can you pastebin a debug log on an incoming call? -MC Here you go. http://pastebin.freeswitch.org/10570 One thing I notice is that in the second line, the caller number is missing. 2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel 1:1 (from to 7100) If libpri doesn't know the number, then it's probably not being sent by the Hicomm. -- Russell Mosemann ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
On Fri, Oct 2, 2009 at 2:53 PM, Russell Mosemann russell.mosem...@cune.orgwrote: cool. can you pastebin a debug log on an incoming call? -MC Here you go. http://pastebin.freeswitch.org/10570 One thing I notice is that in the second line, the caller number is missing. 2009-10-02 16:42:39.539736 [NOTICE] ozmod_libpri.c:772 -- Ring on channel 1:1 (from to 7100) If libpri doesn't know the number, then it's probably not being sent by the Hicomm. Exactly. Turn on q931 debugging and try again: oz libpri debug 1 all PB the results again and we'll check it out. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
Exactly. Turn on q931 debugging and try again: oz libpri debug 1 all PB the results again and we'll check it out. -MC Here's the next one. I'm not sure what to look for, but nothing pops out right away. http://pastebin.freeswitch.org/10571 -- Russell Mosemann ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Need Help in Getting DTMF
Dear all, I am in the process of implementing IVR server in Perl using event outbound socket. Let take the following scenario. There are three menus in the IVR. First menu will authenticate you, second menu get the option value from you,. third menu will give the you the result. You already know all the numbers that you could give. So when the call answered you are giving the value in ONE SHOT. Is it possible to get all the DTMF values in one shot in freeswitch? It should have facility to recollect DTMF values and clear the DTMF values. -- Regards, Thangappan.M ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Need Help in Getting DTMF
You can use play_and_get_digits command or the read command. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read Thanks, Vinuth. On Sat, Oct 3, 2009 at 9:54 AM, Thangappan.M thangappan...@gmail.comwrote: Dear all, I am in the process of implementing IVR server in Perl using event outbound socket. Let take the following scenario. There are three menus in the IVR. First menu will authenticate you, second menu get the option value from you,. third menu will give the you the result. You already know all the numbers that you could give. So when the call answered you are giving the value in ONE SHOT. Is it possible to get all the DTMF values in one shot in freeswitch? It should have facility to recollect DTMF values and clear the DTMF values. -- Regards, Thangappan.M ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
