[Freeswitch-users] meaning of created_time channel variable.
Dear All, What is the value of created_time channel variable? Is this epoch seconds? Thanks Regards, Velusamy. ___ 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] Can not record session. Media not enabled on channel.
Yes, I can confirm - this exact error occurs each time when I start recording
before the call is answered (just after sending ORIGINATE command) - but I
think that's completely understandable that media is not ready for an
unanswered call.
But... is there any other event that guarantees media to be ready?
mercutioviz wrote:
On Fri, Oct 23, 2009 at 12:36 AM, Maciej Aniserowicz
maciej.aniserow...@gmail.com wrote:
The dialplan is very simple:
extension name=Recording test
condition field=destination_number
expression=^11\d*$
action application=sleep data=3000 /
action application=answer/
action application=say data=en name_spelled
iterated
${destination_number}/
action application=sleep data=1000 /
action application=playback
data=local_stream://my_music/
/condition
/extension
Before debugging I have another question: I start recording in event
handler
for ChannelAnswer event. Is it possible that it's too soon to start
recording? Maybe I should start recording in some other event?
That would be an odd scenario but maybe. It would be best if you could
catch
it in the act so that we could see exactly what is happening. The other
thing you could do is deliberately start recording on the channel prior to
answering and see if you always get the error. In other words, try to make
it fail under a certain set of circumstances to see if your theory is
correct.
-MC
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View this message in context:
http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3890610.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
P.S. people from russian community report what current version of module work fine on fs trunk version. that's strange that they report it working as m_txAudioOpened is never gonna be ready in pre_answer :P... i had to comment it to make it working. anyhow, i moved everything to trunk and will do some tests on Monday. T Hello Yuriy, I tried the trunk (FreeSWITCH Version 1.0.trunk (15216M)) and i'm getting some nice coredumps... FS crashes when placing outgoing calls. coredump on outbound call: FS log and backtrace http://pastebin.freeswitch.org/10834 FS crashes on incoming calls. coredump on inbound call: FS log and backtrace http://pastebin.freeswitch.org/10835 FS crashes when i try to load mod_h323. I need 3-4 attempts to load it without a crash. coredump on mod_h323 load: FS log and backtrace: http://pastebin.freeswitch.org/10833 FS crashes on shutdown procedure if mod_h323 was loaded previously. coredump on mod_h323 load: FS log and backtrace: http://pastebin.freeswitch.org/10836 It is quite bad :) ___ 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] Qustion about INFO messages after Connect/Answer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Anthony, hm, no not really. There is no change in this behaviour in FreeSWITCH Version 1.0.trunk (15225M) There is still a caller name change in callee's display. I'm not sure who is wrong here. Either FS or Snom ... regards Helmut On 23.10.2009 17:51, Anthony Minessale wrote: should be even better in 15210 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK5YE94tZeNddg3dwRAigrAJ0chk/kiDQo2Z6yFbjpJovmPor46ACfValq u9RajDfF0rNdzkaUOVjULmk= =697e -END PGP SIGNATURE- ___ 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] mod_nibblebill and memory problem
Dear All, I'm running mod_nibblebill for my prepaid solution. I still have problem with memory. I have 4 GB RAM and runing debian squeeze 64 bit and 200 calls concurrent Last time nibblebill running with 1 min heartbeat. when i check memory by htop FS user memory 2% anf growth to 60-89% in 8-9 hr. and then FS crash. Now i change hearbeat to 0 it's mean nibble update balance when end of call. but everything are same FS start from 2% and growth to 20% in 2 days. When i unload nibblebill FS running fine. My question is when concurrent calls drop to 1-2 calls why FS (I think nibblebill) still use memory ? something wrong in nibblebill ? BG Dome C. ___ 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] Hostname
Thank you for the get idea, It works perfectly. On Fri, Oct 23, 2009 at 4:37 PM, Metik freeswitch-users-l...@metik.comwrote: Why not simply overwrite the value of the variable used throughout the script... -- xml_curl.conf -- ... param name=gateway-url value= http://localhost/index.php?xhostname=myhost; bindings=dialplan/ ... -- index.php -- ? $_REQUEST['hostname'] = $xhostname; ... - Original Message - *From:* freeswitch noob freeswitch.n...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Friday, October 23, 2009 4:42 PM *Subject:* Re: [Freeswitch-users] Hostname Yeah, I was just trying to make it easier on myself. I have scripts from a friend that parse xml_curl requests based on the hostname, I was hoping to not have to re-write them to read something else from the post that FS makes from xml_curl. But from what it sounds like I will have to. On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns ch...@cloudtel.com wrote: one real quick way would be put different GET var in each server's binding On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote: Can't you use different contexts or something else to tell them apart? On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob freeswitch.n...@gmail.com wrote: I have mod xml_curl installed and I am getting the following passed to my script. [hostname] = myhost.local [section] = dialplan I also have multiple versions of FS running on the same box. Is there a way to have each FS instance on my box have a unique hostname ? Thanks in advance. ___ 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 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 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] INFO-Messages: Send_display in pre_answer state doesnt work anymore
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Anthony, thx, it works now again. But the only way get it to work is using send_display after pre_answer. (sip|originate)_callee_id_name doesn't work with playback. regards Helmut On 23.10.2009 16:33, Anthony Minessale wrote: There was a concern with letting this happen before answer. I changed it in last revision, try it out. The problem is that the way polycom does it, it violates the RFC. Also know you can set sip_callee_id_name and sip_callee_id_number before you send a ring_ready pre_answer or answer to send the display update in that packet instead of sending a separate one. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK5bEO4tZeNddg3dwRAkycAKCg61dUy1jkmzauQUDliqYw81aA5ACgmeim hayGIQk5KlRN8edZGLom1mQ= =a5QJ -END PGP SIGNATURE- ___ 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] Downloaded tar vs latest SVN - 200 OK has more headers
No you can't remove them... And they are 100% valid so your SBC is in the wrong. /b On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote: Hi, I used the downlaoded TAR ball and my calls worked, however, when upgrading to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call - using Conferencing app).. Here are teh bad and good 200 OKI see a lot of additional headers startin gwith X:FS , can I remove them? ___ 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] How to load user account from databse ?
Hi All: I'm a newbie to FS. I'm using FS as a sbc and have about 2 user account . Does somebody can tell me how to make FS load use account information from a database such as mssql or mysql? Could you give me a sample configuration file? Thanks a lots. ___ 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] Qustion about INFO messages after Connect/Answer
Could you maybe consolidate all of your problems into 1 thread. I am getting dizzy. You have 2 on the same subject and you say it works on one and does not on the other. Last week we tested all of this with latest trunk and there is no longer any problems of any sort with the display related stuff. On Mon, Oct 26, 2009 at 6:00 AM, Helmut Kuper helmut.ku...@ewetel.dewrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Anthony, hm, no not really. There is no change in this behaviour in FreeSWITCH Version 1.0.trunk (15225M) There is still a caller name change in callee's display. I'm not sure who is wrong here. Either FS or Snom ... regards Helmut On 23.10.2009 17:51, Anthony Minessale wrote: should be even better in 15210 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK5YE94tZeNddg3dwRAigrAJ0chk/kiDQo2Z6yFbjpJovmPor46ACfValq u9RajDfF0rNdzkaUOVjULmk= =697e -END PGP SIGNATURE- ___ 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Resend: Issues with SIP + TCP?
I originally sent this last Friday but I've been unable to confirm it ever made it to the list. Hello everyone, I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior... Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line. This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM). TCP: http://pastebin.freeswitch.org/10825 UDP: http://pastebin.freeswitch.org/10826 dialplan (UDP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62/ /condition /extension dialplan (TCP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62;transport=tcp/ /condition /extension Any thoughts? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 load user account from databse ?
Lei, I am still learning myself, but I think I can help enough and others can chime in where I am wrong. If you have FS up and running you will need to install xml_curl ( http://wiki.freeswitch.org/wiki/Mod_xml_curl). Then you will need to edit the config for it in your conf/autoload_configs/xml_curl_conf.xml and add the following to the file. binding name=externalDir param name=gateway-url value=http://my.webserver.com/index.php; bindings=directory/ /binding Then for every registered user call, a request will be posted to that script location. The posts vary but look something like: [hostname] = test.local [section] = directory [tag_name] = domain [key_name] = name [key_value] = company1.test.local [action] = sip_auth [sip_profile] = internal [sip_user_agent] = eyeBeam release 1100v stamp 47073 [sip_auth_username] = 100 [sip_auth_realm] = company1.test.local [sip_auth_nonce] = 37bd639a-d181-4df3-b53d-ab4172ca3be9 [sip_auth_uri] = sip:company1.test.local;transport=udp [sip_contact_user] = 100 [sip_contact_host] = 10.43.43.2 [sip_to_user] = 100 [sip_to_host] = company1.test.local [sip_from_user] = 100 [sip_from_host] = company1.test.local [sip_request_host] = company1.test.local [sip_auth_qop] = auth [sip_auth_cnonce] = 207aae18a2af959346f87a2a3c2c7f8a [sip_auth_nc] = 0001 [sip_auth_response] = 770a1519789ba7606b3dcc6c4b7a99c5 [sip_auth_method] = REGISTER [key] = id [user] = 100 [domain] = company1.test.local [ip] = 10.43.43.2 Then your script can handle the connection to the DB and the verification of the user information. My example uses PHP but any server side language that can be posted to can be used in the above scenario. On Sun, Oct 25, 2009 at 8:58 AM, Lei Tang lei.tl...@gmail.com wrote: Hi All: I'm a newbie to FS. I'm using FS as a sbc and have about 2 user account . Does somebody can tell me how to make FS load use account information from a database such as mssql or mysql? Could you give me a sample configuration file? Thanks a lots. ___ 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] SIP UPDATE Method
It was never there before and it caused extreme havoc once we added it so we took it away again. On Sun, Oct 25, 2009 at 12:55 AM, DJB djbin...@yahoo.com wrote: I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Am I missing something here? Thank you. ___ 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has more headers
Brian, This bothers me a bit. Of course they are valid. Anything prefixed with X- should be ignored by the remote end unless they are specifically looking for it. However: 1) We all know that just because the spec says they are valid doesn't make it so with every vendor. 2) As long as these headers are being attached people are going to keep coming to the list with issues because of them. 3) Attaching (seemingly arbitrary) headers like this breaks interop agreements that we (the users) have with carriers and other vendors. Every header we add is another reason why $CARRIER will refuse to support me because my INVITE doesn't match what I had during interop testing. 4) Why do we /need/ these headers in the first place? What is using them? If we don't really, really, really need them they shouldn't be there. 5) Depending on the reason for them being there, this should still be configurable. I just don't see why not. Obviously we could always just patch the code (Sonus log statements :) but that probably shouldn't be necessary here. I'm not trying to be difficult; I'm just looking for a better explanation and real discussion. On Mon, Oct 26, 2009 at 10:35 AM, Brian West br...@freeswitch.org wrote: No you can't remove them... And they are 100% valid so your SBC is in the wrong. /b On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote: Hi, I used the downlaoded TAR ball and my calls worked, however, when upgrading to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call - using Conferencing app).. Here are teh bad and good 200 OKI see a lot of additional headers startin gwith X:FS , can I remove them? ___ 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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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
[Freeswitch-users] Mod_socket: custom caller sip domain in originate command
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
origination_caller_id_number=1...@uat.pbx.mblagov.starpoundtech.net - FS
adds its IP address so the
result From address would be 1...@uat.pbx.mblagov.starpoundtech.net@
172.26.200.250
2) specify sip_from_uri varuable like follows:
bgapi originate {origination_caller_id_number=1000,
sip_from_uri=1...@uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
park()
FS doesn't even dial the called endpoint.
Thanks,
Artem
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Re: [Freeswitch-users] Setting up Conference with Moderator
Thx a lot Rob, reading the wiki your way or using IVR seems correct..
===
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
==
I will try it out.
Again thx a lot for your help. Will keep everyone posted.
From: freeswitch-users-boun...@lists.freeswitch.org
[freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rob Forman
[rob4manh...@gmail.com]
Sent: Friday, October 23, 2009 12:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I just re-tested with the pin in my dial plan:
action application=conference data=conference 123...@default
+flags{}+1234 /
And it doesn't challenge me for the pin. I just drop right in. I
figured this is how it was intended, since the wiki says the pin is
set initially and only challenged in later attempts [by future callers]:
The first time a conference name (confname) is used, it will be
created on demand, and the pin will be set to what ever is specified
at that time: the pin in the data string if specified, or if not, the
pin setting in the conference profile, and if that is also
unspecified, then there is no pin protection. Any later attempt to
join the conference must specify the same pin number, if one existed
when it was created.
The wiki also says that the wait-mod might be used in conjunction
with an IVR where the moderators are authenticated with an extra pass-
code, which is what I did. I guess that's why I didn't understand
the point of the +pin.
I'm sure there's a scenario where its used and useful, the wiki just
doesn't explain it.
Rob
On Oct 23, 2009, at 12:43 PM, Brian West wrote:
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
123...@default+flags{} /
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Re: [Freeswitch-users] Resend: Issues with SIP + TCP?
i cant seem to reproduce it. originate sofia/internal/1...@conference.freeswitch.org;transport=tcp 9998 I get a working call and trace. Could you possibly have a dns error? I know it's an ip but it may still fail if it has no dns. try sofia loglevel all 9 and look for other errors. On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: I originally sent this last Friday but I've been unable to confirm it ever made it to the list. Hello everyone, I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior... Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line. This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM). TCP: http://pastebin.freeswitch.org/10825 UDP: http://pastebin.freeswitch.org/10826 dialplan (UDP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62/ /condition /extension dialplan (TCP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62;transport=tcp/ /condition /extension Any thoughts? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 -- 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Reroute a Call Based on the Disconnect Cause
Hello VoIP Geeks!
I am looking for a way to reroute calls on specific disconnect reasons. My
application is in Python. I will provide my solves so far, which for some
reason I cannot use. Why I need rerouting? Well, I am using the LCR module
which provides multiple routes per destination, like this:
routes=
[lcr_carrier=fs-rocks.com,lcr_rate=0.15000]sofia/default/1...@194.135.34.200
:5062|
[lcr_carrier=fs-rocks.com,lcr_rate=0.16000]sofia/default/1...@194.135.34.200
:5064
Solve 1: Python solve
Split the routes string to multiple routes. Bridge the call. Check if the
call failed and bridge again.
ivr_reroute_on=(\
CALL_REJECTED,\
CRASH,\
INVALID_NUMBER_FORMAT,\
NORMAL_TEMPORARY_FAILURE,\
NO_ROUTE_DESTINATION,\
NO_USER_RESPONSE,\
NUMBER_CHANGED,\
RECOVERY_ON_TIMER_EXPIRE,\
SERVICE_NOT_IMPLEMENTED,\
)
...
session.execute(bridge, route)
hangup_cause = str(session.getVariable(originate_disposition))
if hangup_cause in ivr_reroute_on:
#bridge again
...
else:
...
#do not bridge again
This is tested and works. However my manager does not likes it. I am not
commenting on the side effects of this code (i.e. if you have some
timers/schedules/scheduled events etc).
Solve 2: Freeswitch Source Code solve
Modify the source code of switch_ivr_originate.c:
if (to
!oglobals.continue_on_timeout) {
goto
outer_for;
}
after line 2523:
+if
(switch_channel_cause2str(*cause) == SWITCH_CAUSE_CALL_REJECTED ) {
+ goto
outer_for;
+}
(In the best case I will add a new channel variable to Freeswitch to control
all the disconnect reasons from here) However my manager again does not
likes this. He says that I am not good enough in C/C++ to use this solution.
Solve 3: continue_on_fail
I have tried using the channel variable continue_on_fail which I believe
works only in the dialplan with no luck. I was unable to get it running both
in the dialplan and in Python. By the way I saw that continue_on_fail is
indeed implemented elsewhere, while continue_on_timeout is exactly in
switch_ivr_originate.c as expected.
Solve 4:
So can you please people recommend me any other solution? Any function or
variable to do the trick?
Thank you for your time and help!
Delian Tashev
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Re: [Freeswitch-users] Mod_socket: custom caller sip domain in originate command
On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov shiya...@gmail.com wrote:
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
origination_caller_id_number=1...@uat.pbx.mblagov.starpoundtech.net - FS
adds its IP address so the
result From address would be 1...@uat.pbx.mblagov.starpoundtech.net@
172.26.200.250
2) specify sip_from_uri varuable like follows:
bgapi originate {origination_caller_id_number=1000,
sip_from_uri=1...@uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
park()
FS doesn't even dial the called endpoint.
I use
sip_invite_domain
on bridge. I suppose it works with originate too.
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[Freeswitch-users] Estimating Call Capacity
With the following spec for CPU and Memory can someone help me guesstimating how many simultaneous calls and Calls/sec a FS server can handle - Used as a Conferencing Server. cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping: 1 cpu MHz : 2800.386 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips: 5604.12 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping: 1 cpu MHz : 2800.386 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips: 5600.22 # cat /proc/meminfo MemTotal: 1035080 kB MemFree:517972 kB Buffers: 85812 kB Cached: 271264 kB SwapCached: 0 kB Active: 224292 kB Inactive: 223008 kB HighTotal: 130816 kB HighFree:29484 kB LowTotal: 904264 kB LowFree:488488 kB SwapTotal: 2031608 kB SwapFree: 2031520 kB Dirty: 80 kB Writeback: 0 kB AnonPages: 90172 kB Mapped: 39880 kB Slab:60060 kB PageTables: 3232 kB NFS_Unstable:0 kB Bounce: 0 kB CommitLimit: 2549148 kB Committed_AS: 345780 kB VmallocTotal: 114680 kB VmallocUsed: 3584 kB VmallocChunk: 110888 kB HugePages_Total: 0 HugePages_Free: 0 HugePages_Rsvd: 0 Hugepagesize: 4096 kB ___ 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] Downloaded tar vs latest SVN - 200 OK has more headers
The headers are used to pass the callee-id info back to the other side so you have the id of who you called. The standards have failed us in this case as everything does it differently to the point that there is no standard thus we have invented our own way to carry this across from one FreeSWITCH box to another, but of course we can never make anybody happy. =/ On Mon, Oct 26, 2009 at 10:15 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Brian, This bothers me a bit. Of course they are valid. Anything prefixed with X- should be ignored by the remote end unless they are specifically looking for it. However: 1) We all know that just because the spec says they are valid doesn't make it so with every vendor. 2) As long as these headers are being attached people are going to keep coming to the list with issues because of them. 3) Attaching (seemingly arbitrary) headers like this breaks interop agreements that we (the users) have with carriers and other vendors. Every header we add is another reason why $CARRIER will refuse to support me because my INVITE doesn't match what I had during interop testing. 4) Why do we /need/ these headers in the first place? What is using them? If we don't really, really, really need them they shouldn't be there. 5) Depending on the reason for them being there, this should still be configurable. I just don't see why not. Obviously we could always just patch the code (Sonus log statements :) but that probably shouldn't be necessary here. I'm not trying to be difficult; I'm just looking for a better explanation and real discussion. On Mon, Oct 26, 2009 at 10:35 AM, Brian West br...@freeswitch.org wrote: No you can't remove them... And they are 100% valid so your SBC is in the wrong. /b On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote: Hi, I used the downlaoded TAR ball and my calls worked, however, when upgrading to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call - using Conferencing app).. Here are teh bad and good 200 OKI see a lot of additional headers startin gwith X:FS , can I remove them? ___ 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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 -- 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Resend: Issues with SIP + TCP?
Tony, It seemed strange to me too (I'm using TCP in other places). I'll take another look at this with your suggestions for debugging. Thanks! On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale anthony.miness...@gmail.com wrote: i cant seem to reproduce it. originate sofia/internal/1...@conference.freeswitch.org;transport=tcp 9998 I get a working call and trace. Could you possibly have a dns error? I know it's an ip but it may still fail if it has no dns. try sofia loglevel all 9 and look for other errors. On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: I originally sent this last Friday but I've been unable to confirm it ever made it to the list. Hello everyone, I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior... Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line. This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM). TCP: http://pastebin.freeswitch.org/10825 UDP: http://pastebin.freeswitch.org/10826 dialplan (UDP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62/ /condition /extension dialplan (TCP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62;transport=tcp/ /condition /extension Any thoughts? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] Qustion about INFO messages after Connect/Answer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Anthony, sorry for making you dizzy ... bun in fact in my point of view I have two different problems. 1. One concerns the way using send_display in pre_answer mode. Simply to send error texts to caller's display. This works again with latest trunk. 2. The other one (this thread) concerns the (in my eyes newly introduced) two INFO messages which FS sends to callee after callee picked up his phone. The first INFO switches callee's display to a name set in originaton_callee_id_name (a) and immediately after that the second INFO switches it back to callee's real name (b). You can see this in display only if (a) and (b) are not the same. I used Snom 370 phones with FW 8.2.16 as caller and callee. I did an internal call (sip to sip). Maybe this is the same code problem, but on my level they are two different problems, so sorry for confusing you. I hope this clears things up. On 26.10.2009 15:41, Anthony Minessale wrote: Could you maybe consolidate all of your problems into 1 thread. I am getting dizzy. You have 2 on the same subject and you say it works on one and does not on the other. Last week we tested all of this with latest trunk and there is no longer any problems of any sort with the display related stuff. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK5cVz4tZeNddg3dwRAt2+AJsEQuRTEhE74+XvciTXOC0+kr0tbwCfRzUf fgVIZwAV/IthjWwvXzRO3TA= =n7Sr -END PGP SIGNATURE- ___ 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] Mod_socket: custom caller sip domain in originate command
Tested- it works!
Thanks a lot!!
On Mon, Oct 26, 2009 at 6:32 PM, mayamatakeshi mayamatake...@gmail.comwrote:
On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov shiya...@gmail.comwrote:
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
origination_caller_id_number=1...@uat.pbx.mblagov.starpoundtech.net - FS
adds its IP address so the
result From address would be 1...@uat.pbx.mblagov.starpoundtech.net@
172.26.200.250
2) specify sip_from_uri varuable like follows:
bgapi originate {origination_caller_id_number=1000,
sip_from_uri=1...@uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250
park()
FS doesn't even dial the called endpoint.
I use
sip_invite_domain
on bridge. I suppose it works with originate too.
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Re: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has more headers
Anthony, So if I'm understanding you correctly, if you are always using FreeSWITCH as an edge to other systems you should be able to safely disable these headers? On Mon, Oct 26, 2009 at 11:41 AM, Anthony Minessale anthony.miness...@gmail.com wrote: The headers are used to pass the callee-id info back to the other side so you have the id of who you called. The standards have failed us in this case as everything does it differently to the point that there is no standard thus we have invented our own way to carry this across from one FreeSWITCH box to another, but of course we can never make anybody happy. =/ -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] Qustion about INFO messages after Connect/Answer
depending on your dialplan every time you bridge to a channel it changes the display to match who you are talking to. if you tried to set it with the variable then you call someone that is going to cause this. Take away the display app and/or any special variables and let it naturally work. On Mon, Oct 26, 2009 at 10:51 AM, Helmut Kuper helmut.ku...@ewetel.dewrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Anthony, sorry for making you dizzy ... bun in fact in my point of view I have two different problems. 1. One concerns the way using send_display in pre_answer mode. Simply to send error texts to caller's display. This works again with latest trunk. 2. The other one (this thread) concerns the (in my eyes newly introduced) two INFO messages which FS sends to callee after callee picked up his phone. The first INFO switches callee's display to a name set in originaton_callee_id_name (a) and immediately after that the second INFO switches it back to callee's real name (b). You can see this in display only if (a) and (b) are not the same. I used Snom 370 phones with FW 8.2.16 as caller and callee. I did an internal call (sip to sip). Maybe this is the same code problem, but on my level they are two different problems, so sorry for confusing you. I hope this clears things up. On 26.10.2009 15:41, Anthony Minessale wrote: Could you maybe consolidate all of your problems into 1 thread. I am getting dizzy. You have 2 on the same subject and you say it works on one and does not on the other. Last week we tested all of this with latest trunk and there is no longer any problems of any sort with the display related stuff. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK5cVz4tZeNddg3dwRAt2+AJsEQuRTEhE74+XvciTXOC0+kr0tbwCfRzUf fgVIZwAV/IthjWwvXzRO3TA= =n7Sr -END PGP SIGNATURE- ___ 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has more headers
On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale anthony.miness...@gmail.com wrote: The headers are used to pass the callee-id info back to the other side so you have the id of who you called. The standards have failed us in this case as everything does it differently to the point that there is no standard thus we have invented our own way to carry this across from one FreeSWITCH box to another, but of course we can never make anybody happy. =/ I agree with you, X headers should be ignored by the equipment normally. Anyhow Kristian has a point here; there will be a lot of complains because of broken SIP stack on many vendor equipments So, can you consider some customizable a config option for such headers? T. ___ 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] Downloaded tar vs latest SVN - 200 OK has more headers
Thus perpetuating the wild-west of sip where you can't do anything according to spec because you have to worry about stupid things not keeping up. Sounds like the education system where I live too. I'll see what I can do. It's always the other end that ppl pay for that drive the free stuff to change its code. On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga tculj...@gmail.comwrote: On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale anthony.miness...@gmail.com wrote: The headers are used to pass the callee-id info back to the other side so you have the id of who you called. The standards have failed us in this case as everything does it differently to the point that there is no standard thus we have invented our own way to carry this across from one FreeSWITCH box to another, but of course we can never make anybody happy. =/ I agree with you, X headers should be ignored by the equipment normally. Anyhow Kristian has a point here; there will be a lot of complains because of broken SIP stack on many vendor equipments So, can you consider some customizable a config option for such headers? T. ___ 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Resend: Issues with SIP + TCP?
I'm also having problems with this. When running FS compiled about 10 days ago it works fine (don't remember exact revision), but when using latest SVN it doesn't work anymore. I seems like it's trying to use UDP when it should use TCP. My setup is this: Avaya Communication Manager PBX - Talks TLS to Avaya SIP SES Server - Talks TCP to FreeSwitch. The replies from FS seems to be sent using UDP instead of TCP, and when I keep the config and revert to the 10 day old version it starts working again, so there is definately something wrong. I'll try to do some more testing, and get back with some SIP-traces as well. /Peter -Ursprungligt meddelande- Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Kristian Kielhofner Skickat: den 26 oktober 2009 16:47 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? Tony, It seemed strange to me too (I'm using TCP in other places). I'll take another look at this with your suggestions for debugging. Thanks! On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale anthony.miness...@gmail.com wrote: i cant seem to reproduce it. originate sofia/internal/1...@conference.freeswitch.org;transport=tcp 9998 I get a working call and trace. Could you possibly have a dns error? I know it's an ip but it may still fail if it has no dns. try sofia loglevel all 9 and look for other errors. On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: I originally sent this last Friday but I've been unable to confirm it ever made it to the list. Hello everyone, I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior... Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line. This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM). TCP: http://pastebin.freeswitch.org/10825 UDP: http://pastebin.freeswitch.org/10826 dialplan (UDP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62/ /condition /extension dialplan (TCP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62;transport=tcp/ /condition /extension Any thoughts? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 !DSPAM:4ae5c6c832935743011996!
Re: [Freeswitch-users] Resend: Issues with SIP + TCP?
Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad! However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it. /Peter -Ursprungligt meddelande- Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Kristian Kielhofner Skickat: den 26 oktober 2009 16:47 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? Tony, It seemed strange to me too (I'm using TCP in other places). I'll take another look at this with your suggestions for debugging. Thanks! On Mon, Oct 26, 2009 at 11:25 AM, Anthony Minessale anthony.miness...@gmail.com wrote: i cant seem to reproduce it. originate sofia/internal/1...@conference.freeswitch.org;transport=tcp 9998 I get a working call and trace. Could you possibly have a dns error? I know it's an ip but it may still fail if it has no dns. try sofia loglevel all 9 and look for other errors. On Mon, Oct 26, 2009 at 9:56 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: I originally sent this last Friday but I've been unable to confirm it ever made it to the list. Hello everyone, I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior... Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line. This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM). TCP: http://pastebin.freeswitch.org/10825 UDP: http://pastebin.freeswitch.org/10826 dialplan (UDP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62/ /condition /extension dialplan (TCP): extension name=smhpbx condition field=destination_number expression=^(7887)$ action application=set data=call_timeout=60/ action application=set data=effective_caller_id_name=Voalte Test/ action application=set data=effective_caller_id_number=19412848354/ action application=bridge data=sofia/avaya/7...@10.70.0.62;transport=tcp/ /condition /extension Any thoughts? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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 !DSPAM:4ae5c6c832935743011996! ___ 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] Estimating Call Capacity
You'll have to do your own load testing. Nobody can really tell you exactly how many you'll get. /b On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote: With the following spec for CPU and Memory can someone help me guesstimating how many simultaneous calls and Calls/sec a FS server can handle - Used as a Conferencing Server. ___ 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] Resend: Issues with SIP + TCP?
Finding the exact rev that broke it would be helpful. /b On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote: Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad! However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it. /Peter ___ 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] Resend: Issues with SIP + TCP?
Yes, I know :) However, now I think this is related to the new headers introduced, it's probably not a TCP issue. Everything seems to work just fine until the 200 OK is sent, the Avaya PBX doesn't seem to accept that reply anymore. The only differences I've found between a working revision, and a non-working is this; In the working 200-OK transaction method UPDATE is listed in the Allow-header, and there is a header called X-Actually-Support: UPDATE. In the non-working one I don't have these, and instead I have these headers; X-FS-Display-Name: 9099 X-FS-Display-Number: 9099 X-FS-Support: update_display P-Asserted-Identity: 9099 9099 So I guess this could be related to the other thread going on right now Downloaded tar vs latest SVN - 200 OK has more headers, and not a TCP issue. It's probably the Avaya doing something wrong (not the first time), but still it seems these changes affect more systems than mine. I'm just using it as a lab setup for now, so if anyone want me to test something, I can do it immediately. /Peter -Ursprungligt meddelande- Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 26 oktober 2009 18:23 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? Finding the exact rev that broke it would be helpful. /b On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote: Hmm... I remembered incorrectly about my setup :) The Avaya PBX talks TLS to the Avaya SES Server, and then UDP to FS, not TCP - sorry, my bad! However, something that has changed the last 10 days seems to affect my setup so it doesn't work anymore. I'll do some more SIP tracing, and get back when I know more about it. /Peter ___ 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 !DSPAM:4ae5dcba32932131620271! ___ 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] Resend: Issues with SIP + TCP?
Bet your hardware just barfs on those like others have... I mean really I HATE SIP. This is stupid. /b On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote: In the non-working one I don't have these, and instead I have these headers; X-FS-Display-Name: 9099 X-FS-Display-Number: 9099 X-FS-Support: update_display P-Asserted-Identity: 9099 9099 ___ 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] Setup advice on small LAN
Get a dedicated DSL line. They aren't that expensive... I have four of them at my house! /b On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote: Is there a write-up anywhere that might help me with this problem, or lacking that, can anyone offer advice? ___ 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] Setup advice on small LAN
Get a dedicated DSL. That'll work better than any sort of traffic prioritization or shaping (I've tried). Depending on your average channel use and codec, you could probably go with the smallest package and be fine. Rob On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote: I am currently running FreeSwitch very successfully, thanks to the help from many on this list. I am new to Linux so it was a challenge. FS runs on a small LAN with about 5 other computers. The connection to the internet is DSL with 3M down and 768kb down via Covad. The ITSP is Flowroute. If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure. Somehow I need to isolate the FS box from the rest of the LAN, or give its traffic precedence. Covad’s suggestion was to place the FS box in the DMZ. If I have to, I’ll get another DSL line and isolate it that way. Is there a write-up anywhere that might help me with this problem, or lacking that, can anyone offer advice? 581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering flowroute 582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected 583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing expire time to 2243 by request of proxy sip:sip.flowroute.com ... 1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302 Socket up listening on 127.0.0.1:8021 1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected 1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing expire time to 1607 by request of proxy sip:sip.flowroute.c 1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing expire time to 505 by request of proxy sip:sip.flowroute.co 1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing expire time to 210 by request of proxy sip:sip.flowroute.co 1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing expire time to 52 by request of proxy sip:sip.flowroute.com 1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing expire time to 33 by request of proxy sip:sip.flowroute.com 1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing expire time to 22 by request of proxy sip:sip.flowroute.com 1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing expire time to 10 by request of proxy sip:sip.flowroute.com 1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing expire time to 3 by request of proxy sip:sip.flowroute.com 1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering flowroute 1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing expire time to 2 by request of proxy sip:sip.flowroute.com 1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering flowroute 1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing expire time to 1 by request of proxy sip:sip.flowroute.com 1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering flowroute 1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering flowroute 1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing expire time to 60 by request of proxy sip:sip.flowroute.com 1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute Registration Failed with status Operation has no matching challenge [904]. failure #1 Thanks Lars __ Information from ESET NOD32 Antivirus, version of virus signature database 4545 (20091026) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.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 ___ 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] Setup advice on small LAN
I agree, it might be a dirty solution, but its so much easier than trying to get QoS running on a DSL line, or learning how to run a traffic classifier... On Mon, Oct 26, 2009 at 5:46 PM, Brian West br...@freeswitch.org wrote: Get a dedicated DSL line. They aren't that expensive... I have four of them at my house! /b On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote: Is there a write-up anywhere that might help me with this problem, or lacking that, can anyone offer advice? ___ 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] Resend: Issues with SIP + TCP?
try r15230 add the profile param param name=pass-callee-id value=false/ On Mon, Oct 26, 2009 at 12:46 PM, Brian West br...@freeswitch.org wrote: Bet your hardware just barfs on those like others have... I mean really I HATE SIP. This is stupid. /b On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote: In the non-working one I don't have these, and instead I have these headers; X-FS-Display-Name: 9099 X-FS-Display-Number: 9099 X-FS-Support: update_display P-Asserted-Identity: 9099 9099 ___ 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Downloaded tar vs latest SVN - 200 OK has more headers
This is ridiculous but here it is try r15230 add the profile param param name=pass-callee-id value=false/ On Mon, Oct 26, 2009 at 11:16 AM, Anthony Minessale anthony.miness...@gmail.com wrote: Thus perpetuating the wild-west of sip where you can't do anything according to spec because you have to worry about stupid things not keeping up. Sounds like the education system where I live too. I'll see what I can do. It's always the other end that ppl pay for that drive the free stuff to change its code. On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga tculj...@gmail.comwrote: On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale anthony.miness...@gmail.com wrote: The headers are used to pass the callee-id info back to the other side so you have the id of who you called. The standards have failed us in this case as everything does it differently to the point that there is no standard thus we have invented our own way to carry this across from one FreeSWITCH box to another, but of course we can never make anybody happy. =/ I agree with you, X headers should be ignored by the equipment normally. Anyhow Kristian has a point here; there will be a lot of complains because of broken SIP stack on many vendor equipments So, can you consider some customizable a config option for such headers? T. ___ 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 -- 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Resend: Issues with SIP + TCP?
I understand your frustration :) We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though. /Peter -Ursprungligt meddelande- Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 26 oktober 2009 18:46 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? Bet your hardware just barfs on those like others have... I mean really I HATE SIP. This is stupid. /b On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote: In the non-working one I don't have these, and instead I have these headers; X-FS-Display-Name: 9099 X-FS-Display-Number: 9099 X-FS-Support: update_display P-Asserted-Identity: 9099 9099 ___ 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 !DSPAM:4ae5e23832938073513968! ___ 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] Downloaded tar vs latest SVN - 200 OK has more headers
New sofia profile param as follows: !-- set this param to false if your gateway for some reason hates X- headers that is is supposed to ignore-- !--param name=pass-callee-id value=false/-- On Oct 26, 2009, at 12:16 PM, Anthony Minessale wrote: Thus perpetuating the wild-west of sip where you can't do anything according to spec because you have to worry about stupid things not keeping up. Sounds like the education system where I live too. I'll see what I can do. It's always the other end that ppl pay for that drive the free stuff to change its code. On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga tculj...@gmail.com wrote: On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale anthony.miness...@gmail.com wrote: The headers are used to pass the callee-id info back to the other side so you have the id of who you called. The standards have failed us in this case as everything does it differently to the point that there is no standard thus we have invented our own way to carry this across from one FreeSWITCH box to another, but of course we can never make anybody happy. =/ I agree with you, X headers should be ignored by the equipment normally. Anyhow Kristian has a point here; there will be a lot of complains because of broken SIP stack on many vendor equipments So, can you consider some customizable a config option for such headers? T. ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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] Resend: Issues with SIP + TCP?
At some point we'll have to NO NO NO fix your broken crap. :P The reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with me... NO! /b On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote: I understand your frustration :) We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though. /Peter ___ 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] Setup advice on small LAN
On 26/10/09 10:37 -0700, Lars Zeb wrote: If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure. QoS is a two way street. You could spend a lot of time getting your egress traffic properly prioritized, but if your ISP does not do the same prioritization on your ingress traffic (toward you), you'll still have problems during downloads. If your friendly neighborhood ISP will work with you on prioritization, that's another matter. -- Dan White ___ 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] SIP UPDATE Method
what exactly are you expecting to use it for? We never really supported it anyway. On Mon, Oct 26, 2009 at 12:19 PM, DJB djbin...@yahoo.com wrote: I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it? Regards, Dorn B. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, October 26, 2009 8:08:59 AM *Subject:* Re: [Freeswitch-users] SIP UPDATE Method It was never there before and it caused extreme havoc once we added it so we took it away again. On Sun, Oct 25, 2009 at 12:55 AM, DJB djbin...@yahoo.com wrote: I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Am I missing something here? Thank you. ___ 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 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP UPDATE Method
We were trying to see whether we can adjust call duration on Session timers. It was a question from the application developers. I am not sure what they are trying to do exactly. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, October 26, 2009 11:45:31 AM Subject: Re: [Freeswitch-users] SIP UPDATE Method what exactly are you expecting to use it for? We never really supported it anyway. On Mon, Oct 26, 2009 at 12:19 PM, DJB djbin...@yahoo.com wrote: I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it? Regards, Dorn B. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, October 26, 2009 8:08:59 AM Subject: Re: [Freeswitch-users] SIP UPDATE Method It was never there before and it caused extreme havoc once we added it so we took it away again. On Sun, Oct 25, 2009 at 12:55 AM, DJB djbin...@yahoo.com wrote: I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Am I missing something here? Thank you. ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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
Re: [Freeswitch-users] Setup advice on small LAN
Lars, If your ISP is a COVAD partner, ask them to reprovision the line for VOA (voice optimized access). They will provide you with two seperate VLANs (one for best effort data and the other for real-time/voice traffic). If they are unable to do so or do not understand your request, feel free to email my off list and I can help you. The cost difference should be minimal unless you move to second line (a/k/a naked, unbundled, or dedicated) DSL. -metik - Original Message - From: Lars Zeb To: freeswitch-users@lists.freeswitch.org Sent: Monday, October 26, 2009 1:37 PM Subject: [Freeswitch-users] Setup advice on small LAN I am currently running FreeSwitch very successfully, thanks to the help from many on this list. I am new to Linux so it was a challenge. FS runs on a small LAN with about 5 other computers. The connection to the internet is DSL with 3M down and 768kb down via Covad. The ITSP is Flowroute. If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure. Somehow I need to isolate the FS box from the rest of the LAN, or give its traffic precedence. Covad's suggestion was to place the FS box in the DMZ. If I have to, I'll get another DSL line and isolate it that way. Is there a write-up anywhere that might help me with this problem, or lacking that, can anyone offer advice? 581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering flowroute 582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected 583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing expire time to 2243 by request of proxy sip:sip.flowroute.com ... 1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302 Socket up listening on 127.0.0.1:8021 1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected 1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing expire time to 1607 by request of proxy sip:sip.flowroute.c 1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing expire time to 505 by request of proxy sip:sip.flowroute.co 1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing expire time to 210 by request of proxy sip:sip.flowroute.co 1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing expire time to 52 by request of proxy sip:sip.flowroute.com 1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing expire time to 33 by request of proxy sip:sip.flowroute.com 1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing expire time to 22 by request of proxy sip:sip.flowroute.com 1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing expire time to 10 by request of proxy sip:sip.flowroute.com 1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing expire time to 3 by request of proxy sip:sip.flowroute.com 1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering flowroute 1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing expire time to 2 by request of proxy sip:sip.flowroute.com 1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering flowroute 1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing expire time to 1 by request of proxy sip:sip.flowroute.com 1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering flowroute 1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering flowroute 1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing expire time to 60 by request of proxy sip:sip.flowroute.com 1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute Registration Failed with status Operation has no matching challenge [904]. failure #1 Thanks Lars __ Information from ESET NOD32 Antivirus, version of virus signature database 4545 (20091026) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.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 ___ 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] SIP UPDATE Method
I think session timers will use invite if there is no update. the session-timeout profile param should control that but you have to double the number you actually want because it sends the new invite at the halfway point. On Mon, Oct 26, 2009 at 2:29 PM, DJB djbin...@yahoo.com wrote: We were trying to see whether we can adjust call duration on Session timers. It was a question from the application developers. I am not sure what they are trying to do exactly. Thank you. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, October 26, 2009 11:45:31 AM *Subject:* Re: [Freeswitch-users] SIP UPDATE Method what exactly are you expecting to use it for? We never really supported it anyway. On Mon, Oct 26, 2009 at 12:19 PM, DJB djbin...@yahoo.com wrote: I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it? Regards, Dorn B. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, October 26, 2009 8:08:59 AM *Subject:* Re: [Freeswitch-users] SIP UPDATE Method It was never there before and it caused extreme havoc once we added it so we took it away again. On Sun, Oct 25, 2009 at 12:55 AM, DJB djbin...@yahoo.com wrote: I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Am I missing something here? Thank you. ___ 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 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 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP UPDATE Method
but i think the minimum value you can set is 120 On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale anthony.miness...@gmail.com wrote: I think session timers will use invite if there is no update. the session-timeout profile param should control that but you have to double the number you actually want because it sends the new invite at the halfway point. On Mon, Oct 26, 2009 at 2:29 PM, DJB djbin...@yahoo.com wrote: We were trying to see whether we can adjust call duration on Session timers. It was a question from the application developers. I am not sure what they are trying to do exactly. Thank you. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, October 26, 2009 11:45:31 AM *Subject:* Re: [Freeswitch-users] SIP UPDATE Method what exactly are you expecting to use it for? We never really supported it anyway. On Mon, Oct 26, 2009 at 12:19 PM, DJB djbin...@yahoo.com wrote: I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it? Regards, Dorn B. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, October 26, 2009 8:08:59 AM *Subject:* Re: [Freeswitch-users] SIP UPDATE Method It was never there before and it caused extreme havoc once we added it so we took it away again. On Sun, Oct 25, 2009 at 12:55 AM, DJB djbin...@yahoo.com wrote: I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Am I missing something here? Thank you. ___ 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 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 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 -- 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
Re: [Freeswitch-users] Estimating Call Capacity
Are there any benchmarking test results available publicly? From: freeswitch-users-boun...@lists.freeswitch.org [freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West [br...@freeswitch.org] Sent: Monday, October 26, 2009 11:18 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity You'll have to do your own load testing. Nobody can really tell you exactly how many you'll get. /b On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote: With the following spec for CPU and Memory can someone help me guesstimating how many simultaneous calls and Calls/sec a FS server can handle - Used as a Conferencing Server. ___ 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] SIP UPDATE Method
in r15233 i put it back to the way it originally was but I may have to remove that if it causes more problems. We tried to handle update for display updating which was not working but the handler for it was still in place which may have broken some automatic behavior regarding update so I added it back to how it was originally to determine that. On Mon, Oct 26, 2009 at 2:50 PM, Anthony Minessale anthony.miness...@gmail.com wrote: but i think the minimum value you can set is 120 On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale anthony.miness...@gmail.com wrote: I think session timers will use invite if there is no update. the session-timeout profile param should control that but you have to double the number you actually want because it sends the new invite at the halfway point. On Mon, Oct 26, 2009 at 2:29 PM, DJB djbin...@yahoo.com wrote: We were trying to see whether we can adjust call duration on Session timers. It was a question from the application developers. I am not sure what they are trying to do exactly. Thank you. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, October 26, 2009 11:45:31 AM *Subject:* Re: [Freeswitch-users] SIP UPDATE Method what exactly are you expecting to use it for? We never really supported it anyway. On Mon, Oct 26, 2009 at 12:19 PM, DJB djbin...@yahoo.com wrote: I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it? Regards, Dorn B. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, October 26, 2009 8:08:59 AM *Subject:* Re: [Freeswitch-users] SIP UPDATE Method It was never there before and it caused extreme havoc once we added it so we took it away again. On Sun, Oct 25, 2009 at 12:55 AM, DJB djbin...@yahoo.com wrote: I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Am I missing something here? Thank you. ___ 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 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 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 -- Anthony Minessale II
Re: [Freeswitch-users] Resend: Issues with SIP + TCP?
On Mon, Oct 26, 2009 at 11:40 AM, Brian West br...@freeswitch.org wrote: At some point we'll have to NO NO NO fix your broken crap. :P The reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with me... NO! /b I was wondering... does anyone make a SIP certification program kinda like a pen-tester except to find all the ways your SIP setup is broken? Just curious. -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] Estimating Call Capacity
I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ 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] Resend: Issues with SIP + TCP?
You have SIPit, which was the SIP Backoff till Pillsbury got their panties in a wad. /b On Oct 26, 2009, at 3:03 PM, Michael Collins wrote: I was wondering... does anyone make a SIP certification program kinda like a pen-tester except to find all the ways your SIP setup is broken? Just curious. -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] How to load user account from databse ?
On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang lei.tl...@gmail.com wrote: Hi All: I'm a newbie to FS. I'm using FS as a sbc and have about 2 user account . Does somebody can tell me how to make FS load use account information from a database such as mssql or mysql? Could you give me a sample configuration file? Thanks a lots. Lei, The feature that you want is mod_xml_curl - it allows you to pull config information from a web server, and that web server will do the db lookup. For more information check out these resources: Wiki docs: http://wiki.freeswitch.org/wiki/Mod_xml_curl Example: http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/trixter/xml-curl We don't have a soup-to-nuts how-to on mod_xml_curl because it can be done in so many different ways. Someone could write a book on implementing xml_curl techniques. Your best bet is to figure out how you want to store your user information, then decide what's the best way to pull that information from the database, then setup a web server to handle the request/return process. Between the docs and the examples you should be able to get up and running. -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] Estimating Call Capacity
Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ 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] Downloaded tar vs latest SVN - 200 OK has more headers
It is ridiculous but thank you very much! On Mon, Oct 26, 2009 at 2:26 PM, Anthony Minessale anthony.miness...@gmail.com wrote: This is ridiculous but here it is try r15230 add the profile param param name=pass-callee-id value=false/ -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] Resend: Issues with SIP + TCP?
Woof! On Mon, 26 Oct 2009 16:03:59 -0400, Michael Collins m...@freeswitch.org wrote: I was wondering... does anyone make a SIP certification program kinda like a pen-tester except to find all the ways your SIP setup is broken? Just curious. Here is a start: http://interop.sipxecs.org/ It's best for testing phone implementations, but it can handle other UA's as well. --Woof! ___ 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] Estimating Call Capacity
On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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] problems registering on www.freeswitch.org
Hi, I tried to register on freeswitch.org using username sjmudd. It said the username already existed so I requested having my password resent. Nothing recieved. I've also tried to register a new user simon.mudd with the same results, no mail received. I also see the following when trying to register which looks worrying: http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png I'm pretty sure that I'm not filtering out the emails but can't be sure. Could whoever maintains the web page contact me off list to help me determine where the problem is? Thanks, Simon Mudd ___ 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] Can not record session. Media not enabled on channel.
On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz maciej.aniserow...@gmail.com wrote: Yes, I can confirm - this exact error occurs each time when I start recording before the call is answered (just after sending ORIGINATE command) - but I think that's completely understandable that media is not ready for an unanswered call. But... is there any other event that guarantees media to be ready? Update to latest SVN and try again. -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] Inbound DTMF Not Recognized By IVR
On Fri, Oct 23, 2009 at 10:31 AM, Jerry Richards jerry.richa...@teotech.com wrote: I installed FS on a machine with a Sangoma A101D (PRI) card and if I make an inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN phone. If I call IVR from an internal phone, then it does recognize the DTMF digits. I have mostly default configurations for everything. Best Regards, JErry I'm assuming that the DTMFs are coming in-band? If so make sure that you issue the start_dtmf dialplan app. -MC http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf ___ 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] problems registering on www.freeswitch.org
On Mon, Oct 26, 2009 at 2:23 PM, Simon J Mudd sjm...@pobox.com wrote: Hi, I tried to register on freeswitch.org using username sjmudd. It said the username already existed so I requested having my password resent. Nothing recieved. I've also tried to register a new user simon.mudd with the same results, no mail received. I also see the following when trying to register which looks worrying: http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png I'm pretty sure that I'm not filtering out the emails but can't be sure. Could whoever maintains the web page contact me off list to help me determine where the problem is? Thanks, Simon Mudd Thanks for letting us know. I'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] Downloaded tar vs latest SVN - 200 OK has more headers
On Mon, Oct 26, 2009 at 7:26 PM, Anthony Minessale anthony.miness...@gmail.com wrote: This is ridiculous but here it is try r15230 add the profile param param name=pass-callee-id value=false/ sorry for that but, this will save you a lot of e-mail explaining why calls are not going through... thanks man! T. ___ 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] Setup advice on small LAN
If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure. In any case, regardless if you are using a dedicated or mixed dsl line you should flag your voice traffic properly. signaling AF41, RTP EF... your voice traffic must never be flagged as pure date when sending it through open internet! T. ___ 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 load user account from databse ?
Thanks for your response Michael, Both the resources you've referred to don't explicitly say much about databases. Could you elaborate just a bit on how on information would retrieved from a database (MySQL) and presented to mod_xml_curl. Thanks again. 2009/10/26 Michael Collins m...@freeswitch.org On Sun, Oct 25, 2009 at 6:58 AM, Lei Tang lei.tl...@gmail.com wrote: Hi All: I'm a newbie to FS. I'm using FS as a sbc and have about 2 user account . Does somebody can tell me how to make FS load use account information from a database such as mssql or mysql? Could you give me a sample configuration file? Thanks a lots. Lei, The feature that you want is mod_xml_curl - it allows you to pull config information from a web server, and that web server will do the db lookup. For more information check out these resources: Wiki docs: http://wiki.freeswitch.org/wiki/Mod_xml_curl Example: http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/trixter/xml-curl We don't have a soup-to-nuts how-to on mod_xml_curl because it can be done in so many different ways. Someone could write a book on implementing xml_curl techniques. Your best bet is to figure out how you want to store your user information, then decide what's the best way to pull that information from the database, then setup a web server to handle the request/return process. Between the docs and the examples you should be able to get up and running. -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 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] Setup advice on small LAN
Unfortunately, most North American providers ignore (and in most cases reclassify) it before it reaches their border routers and it will be treated as best effort. Typically, the problems are introduced within the first mile and its simply a matter of getting the packet safely pass the edge unless your ISP is grossly oversubscribed (in terms of servicable traffic limited by their particular hardware and bandwidth). If it is too costly to move to second line or dedicated DSL, he should be able to improve audio quality by acquiring a broadband router that has minimal QoS capabilities and adequate CPU (since the majority of them use software based queuing and packet fragmentation). The only caveat is that the degree of success can vary between firmware versions. Some of the consumer (gaming) or small business (VPN) grade routers work well (Linksys, DLINK, etc.). -metik From: Tihomir Culjaga To: freeswitch-users@lists.freeswitch.org Sent: Monday, October 26, 2009 5:34 PM Subject: Re: [Freeswitch-users] Setup advice on small LAN If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure. In any case, regardless if you are using a dedicated or mixed dsl line you should flag your voice traffic properly. signaling AF41, RTP EF... your voice traffic must never be flagged as pure date when sending it through open internet! T. -- ___ 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] Estimating Call Capacity
Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, October 26, 2009 4:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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] Estimating Call Capacity
i wonder if we can at least get a taco-bell steak burrito for that if we can't win the s-prize On Mon, Oct 26, 2009 at 5:01 PM, Eliot Gable egable+freeswi...@gmail.comegable%2bfreeswi...@gmail.com wrote: Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, October 26, 2009 4:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2http://wiki.voiceworks.pl/display/%7Epawel/FreeSwitch+performance+on+SUN+x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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 -- 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
Re: [Freeswitch-users] Estimating Call Capacity
On Mon, Oct 26, 2009 at 4:03 PM, Anthony Minessale anthony.miness...@gmail.com wrote: i wonder if we can at least get a taco-bell steak burrito for that if we can't win the s-prize Or at least a chalupa. -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] mod_nibblebill and memory problem
Why don't you get us more information for debugging? We could use some vg output, maybe? JM On Mon, Oct 26, 2009 at 9:24 AM, Dome Charoenyost d...@tel.co.th wrote: Dear All, I'm running mod_nibblebill for my prepaid solution. I still have problem with memory. I have 4 GB RAM and runing debian squeeze 64 bit and 200 calls concurrent Last time nibblebill running with 1 min heartbeat. when i check memory by htop FS user memory 2% anf growth to 60-89% in 8-9 hr. and then FS crash. Now i change hearbeat to 0 it's mean nibble update balance when end of call. but everything are same FS start from 2% and growth to 20% in 2 days. When i unload nibblebill FS running fine. My question is when concurrent calls drop to 1-2 calls why FS (I think nibblebill) still use memory ? something wrong in nibblebill ? BG Dome C. ___ 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] Estimating Call Capacity
yo quiero taco bell /b On Oct 26, 2009, at 7:04 PM, Michael Collins wrote: Or at least a chalupa. -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] Problem with inbound call answered but no sound
I have tried to update (make current) twice since 15183. All inbound calls are picked up but the caller hears nothing but a couple of clicks. The most recent version I've tried is 15241. Any ideas on what may be causing this? http://pastebin.freeswitch.org/10843 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars ___ 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] Problem with inbound call answered but no sound
you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip correctly? /b On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote: I have tried to update (make current) twice since 15183. All inbound calls are picked up but the caller hears nothing but a couple of clicks. The most recent version I’ve tried is 15241. Any ideas on what may be causing this? http://pastebin.freeswitch.org/10843 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars ___ 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] Problem with inbound call answered but no sound
FYI, it generally makes debugging easier if you do this: sofia profile external siptrace on sofia profile internal siptrace on That way you can see the actual signaling and it is usually more clear what is going on. In most cases, you will probably be able to figure it out yourself just looking at the signaling. On Mon, Oct 26, 2009 at 10:29 PM, Lars Zeb larc...@yahoo.com wrote: I have tried to update (make current) twice since 15183. All inbound calls are picked up but the caller hears nothing but a couple of clicks. The most recent version I’ve tried is 15241. Any ideas on what may be causing this? http://pastebin.freeswitch.org/10843 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars ___ 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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ 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] Problem with inbound call answered but no sound
I haven't changed anything since v15183, where it worked OK. In conf/sip_profiles/external: param name=ext-rtp-ip value=auto-nat/ param name=ext-sip-ip value=auto-nat/ And Nameexternal Domain Name N/A DBName sofia_reg_external Pres Hosts DialplanXML Context public Challenge Realm auto_to RTP-IP 192.168.10.29 Ext-RTP-IP 192.168.10.29 SIP-IP 192.168.10.29 Ext-SIP-IP 192.168.10.29 URL sip:mod_so...@192.168.10.29:5090 BIND-URLsip:mod_so...@192.168.10.29:5090 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN1 FAILED-CALLS-IN 1 CALLS-OUT 0 FAILED-CALLS-OUT0 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, October 26, 2009 7:46 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem with inbound call answered but no sound you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip correctly? /b On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote: I have tried to update (make current) twice since 15183. All inbound calls are picked up but the caller hears nothing but a couple of clicks. The most recent version I've tried is 15241. Any ideas on what may be causing this? http://pastebin.freeswitch.org/10843 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars ___ 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] Playing Background music as well as a file
Hi all, I've done experimenting with the uuid_displace and mux. What mux does is playing a file when the conversation is also happening. But I've a different requirement. I need to play a background music to a UUID, that will get played continuously and also I need to play some other voice message to that uuid. Is it possible in freeswitch? If so please guide me on how to do that! ___ 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
