[Freeswitch-users] Problem with subscription expire
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, it seems exired subsciptions are never cleared in FS. A look into sofia_presence.c confirms explains this /* negative in exptime means keep bumping up sub time to avoid a snafu where every device has it's own rules about subscriptions that somehow barely resemble the RFC not that I blame them because the RFC MAY be amibiguous and SHOULD be deleted. So to avoid the problem we keep resetting the expiration date of the subscription so it never expires. Eybeam completely ignores this option and most other subscription-state: directives from rfc3265 and still expires. Polycom is happy to keep upping the subscription expiry back to the original time on each new notify. The rest ... who knows...? */ For some reasons subscriptions created by Snom phones are filling up the sip_subscriptions table over time. This leads to some kind of DOS by FS against the subscribing phone ... The subscribtions are differentiate by call-id. This can be explained by RFC 3842 chapter 3.6 where expired subscriptions must be renewed with a NEW call-id. Because there is no hint about unsubscribing the old subscription I guess the clean up process has to be done by FS. Any way to get FS to do this job? Since there is no creation date or expire value which represents the expire as a timestamp I have no way to clean up the table manually via sql and cronjob - except cleaning the whole table ... A further (but background) question is, why do the subscriptions expire in snom phones at all ... regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKxG3q4tZeNddg3dwRArNEAJ9fjHLox1tt038ze0liUG0ki+wrfgCgsz09 pO+XUioXrBKJ/ozUOy1ZqeA= =nZaf -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] Listening to a connected call [barge in]
In ES outbound, I need to do the following, * A calls 2000(FS ES outbound extension) * In the script, It'll answer the call, play some files and get the reply from A(as voice). * Simultaneously(when doing the above), the script has to call B. * When B attends the call, B has to listen to the live conversation between 2000 A. How should I do.? I've tried this with async mode and by listening to the events. But I couldn't do it. Help me to do this.. Regards, Nagalenoj -- View this message in context: http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p25696889.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Listening to a connected call [barge in]
Hi, Use the eavesdrop command. Just supply it with the call UUID and the extension of B. Wiki has more details. Thanks, Vinuth. On Thu, Oct 1, 2009 at 7:20 PM, Nagalenoj nagale...@gmail.com wrote: In ES outbound, I need to do the following, * A calls 2000(FS ES outbound extension) * In the script, It'll answer the call, play some files and get the reply from A(as voice). * Simultaneously(when doing the above), the script has to call B. * When B attends the call, B has to listen to the live conversation between 2000 A. How should I do.? I've tried this with async mode and by listening to the events. But I couldn't do it. Help me to do this.. Regards, Nagalenoj -- View this message in context: http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p25696889.html Sent from the Freeswitch-users mailing list archive at Nabble.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
[Freeswitch-users] Dialplan Issue
Hello:
I asked this on IRC yesterday and I think I confused everyone involved. So
I apologize in advance here for reposting the question and if I wasted
anyone's time.
So here is the issue I'm having. I'm trying to use FS as a redirect server
(specifically to serve up LNP queries via 302 redirects). But I'm having an
issue where based on the string in the dialplan FS will respond with a 500
internal error message instead of a 300 redirect.
The call flow should be this:
-- remote party sends an Invite to my FS instance
-- FS should respond with a 302
The following works as expected (FS will send a 302 when it receives an
Invite):
action application=redirect data=sip:${destination_number}@
${network_addr};rn=${rn};npdi=yes/
However if I do this (which is the way the response should look) FS will
respond with a 500 internal server error:
action application=redirect
data=sip:${destination_number};rn=${rn};npdi=...@${network_addr}/
So the issue is the placement of the user params if they are before the
@ FS will send a 500 internal server error ... if they are after the @ FS
will send a 302. Unfortunately placing the user params after the @ doesn't
quite conform to the way other devices expect to receive the 302 for this
application.
Any help would be greatly appreciated.
Shelby
PS ... hats off to the author of mod_memcache ... that is extremely useful!
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Re: [Freeswitch-users] Dialplan Issue
On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:
action application=redirect data=sip:${destination_number};rn=${rn};npdi=...@${network_addr}
/
This will produce an INVALID sip uri... You can not feed this to sofia
it'll get PISSED.
Its missing the host portion.
So the issue is the placement of the user params if they are
before the @ FS will send a 500 internal server error ... if they
are after the @ FS will send a 302. Unfortunately placing the user
params after the @ doesn't quite conform to the way other devices
expect to receive the 302 for this application.
Any help would be greatly appreciated.
Shelby
PS ... hats off to the author of mod_memcache ... that is extremely
useful!
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Re: [Freeswitch-users] REGISTER fails with 407 after minutes of success register
Thanks for posting the logs... But I'm not going to spend the time to download it.. unzip it and look at it... I would rather just click a link with the logs in plain text and read them in my browser. I'll do it now but next time lets not add steps to the process that are not needed. This goes for Jira too don't upload zip files of text logs that just makes it harder for us to quickly help you. This isn't going to help me much know why Sofia/FreeSWITCH isn't working. sofia profile xxx siptrace on press F8 sofia loglevel all 9 Then post that please. /b On Oct 1, 2009, at 9:35 AM, Fernando Testa wrote: In the link below you have the entire SIP trace from system startup until start receiving this annoying 407 Proxy Auth Required, preventing FS to register successfully on the Ericsson Pabx. You can notice multiple registrations from named ericsson_1050 to ericsson_1064 that starts failing after ~50 minutes after the boot. Issuing a 'sofia external profile restart' solves the registration problems. Brian, thanks for reply, but I really didn't get your point. Thank you, I apreciate any help. http://dl.getdropbox.com/u/410277/sip.log.gz ___ 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] Dialplan Issue
you left too fast.
1) I told you if you put around the sip uri it will work.
2) I told you I added a patch in tree to add one for you if it's not
supplied so update to trunk.
On Thu, Oct 1, 2009 at 9:27 AM, Shelby Ramsey sicfsl...@gmail.com wrote:
Hello:
I asked this on IRC yesterday and I think I confused everyone involved. So
I apologize in advance here for reposting the question and if I wasted
anyone's time.
So here is the issue I'm having. I'm trying to use FS as a redirect server
(specifically to serve up LNP queries via 302 redirects). But I'm having an
issue where based on the string in the dialplan FS will respond with a 500
internal error message instead of a 300 redirect.
The call flow should be this:
-- remote party sends an Invite to my FS instance
-- FS should respond with a 302
The following works as expected (FS will send a 302 when it receives an
Invite):
action application=redirect data=sip:${destination_number}@
${network_addr};rn=${rn};npdi=yes/
However if I do this (which is the way the response should look) FS will
respond with a 500 internal server error:
action application=redirect
data=sip:${destination_number};rn=${rn};npdi=...@${network_addr}/
So the issue is the placement of the user params if they are before
the @ FS will send a 500 internal server error ... if they are after the @
FS will send a 302. Unfortunately placing the user params after the @
doesn't quite conform to the way other devices expect to receive the 302 for
this application.
Any help would be greatly appreciated.
Shelby
PS ... hats off to the author of mod_memcache ... that is extremely useful!
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--
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
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Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To Subscribing Phones
By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dialplan Issue
Tony, Once again ... you are the man! I'll try this right now. SDR ___ 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] Dialplan Issue
Brian,
Thanks for the info. I guess I'll go read section 19.1 of RFC3261 again. I
do think the above has a valid host portion (I don't think the port is
required).
I'm not so sure that putting params in the user portion of the uri is valid
(from the RFC it states sip:user:passw...@host:port;uri-parameters?headers).
The issue is that in the real world this is done all the time SIP is
fantastic :)
Shelby
On Thu, Oct 1, 2009 at 9:42 AM, Brian West br...@freeswitch.org wrote:
On Oct 1, 2009, at 9:27 AM, Shelby Ramsey wrote:
action application=redirect data=
sip:${destination_number};rn=${rn};npdi=...@${network_addr}/
This will produce an INVALID sip uri... You can not feed this to sofia
it'll get PISSED.
Its missing the host portion.
So the issue is the placement of the user params if they are before
the @ FS will send a 500 internal server error ... if they are after the @
FS will send a 302. Unfortunately placing the user params after the @
doesn't quite conform to the way other devices expect to receive the 302 for
this application.
Any help would be greatly appreciated.
Shelby
PS ... hats off to the author of mod_memcache ... that is extremely useful!
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Re: [Freeswitch-users] Dialplan Issue
I wouldn't go that far! :P You might be able to get away with it on the patch tony wrote but not sure. /b On Oct 1, 2009, at 10:59 AM, Shelby Ramsey wrote: SIP is fantastic :) ___ 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] Dialplan Issue
Just to confirm ... works like a champ. Thanks again!!! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To Subscribing Phones
which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.comwrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using heartbeat and ldirectord for load-balancing, fail-over and high availability! I'm probably not the first one to do it, but as near as Google and I can tell, I'm the first one to write about it. Here's how you can duplicate my setup: 1. Install Ubuntu Server 8 on four machines, either real or VM. 2. Compile and install FreeSWITCH v1.0.4 from source on two machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start3. Configure both FreeSWITCH boxes, and make sure they are both working. 4. Follow (most of) these instructions from Daniel Aliaman's blog. They were written for Asterisk, but since a SIP connection is a SIP connection, most of the document applies to FreeSWITCH: http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdfThe one problem I ran into was the IP address and port to which FreeSWITCH was bound. The default is to use the primary address, which works great out-of-the-box for everything else. When a client tried to register, all it got back was an ICMP error -- Destination Unreachable, Port Unreachable. That error is returned when no sockets are listening for UDP packets. To get FreeSWITCH to listen for your Virtual IP, you need to set it in two places: 5. In /opt/freeswitch/conf/vars.xml, set bind_server_ip. 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set sip-ip. That should do it! If you have any success, please report to this list. Keep in mind that if you want to do something like conferencing between two registered clients, then you have to deal with the fact that the clients may or may not be on the same box. Mike van Lammeren On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren m...@van.lammeren.netwrote: On Mon, Sep 28, 2009 at 9:05 PM, Even André Fiskvik grev...@me.com wrote: From: Even André Fiskvik grev...@me.com To: freeswitch-users@lists.freeswitch.org Date: Mon, 28 Sep 2009 22:52:13 +0200 Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey I have been working with a similar setup myself, but for some reason I ended up ditching theUltraMonkey setup because I just couldn't get it to work right. It's been quite a while since my effort, so I don't remember what the exact issue was. I got registrations to work, but had some other sip-dialog issues. We have since then changed over to running OpenSIPs as a loadbalancer in front of multiple FreeSWITCH instances. This setup is still in testing, but seemlingy works fine (and if it doesn't, it's my own fault for writing a bad opensips config). After we have done some more testing I can create a wiki-page with config details. Best regards, Even André Thanks, Even, that would be great! I might have to give up on the ultramonkey solution, since I can't find anyone who has made it work. It's too bad, because it would fit well with the rest of our architecture. Mike van Lammeren ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?
My SIP provider allows only one call (incoming or outgoing) via one SIP account. For FreeSWITCH I have configured it as public DID extension and outgoing gateway. Now I would like to transfer to another gw (or generate limit exceded) when one tries to place an outgoing call while incoming call is in progress. How tho do that? Limiting the number of outgoing calls is easy (mod_limit), but how to take into account incoming one? - Dmitry Bely ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones
I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
That's very cool Mike! I'm going to try to configure four boxes with this as well (Btw, did you use physical hardware or virtualization?) and see how it goes. I followed Daniel Aliaman's blog as well, but I can try it again with the tips you provided on FreeSWITCH config to see if I can get it working properly this time. We did the setup on CentOS, but I wouldn't think that would be any issue. Perhaps you or we could write up a complete guide about this on the wiki since this is an scenario commonly used? Also it would be great if we could outline possible issues (and even better solutions) to this kind of setup with regards to stuff like conferencing, bridging between registered users and presence. Best regards, Even André On 1. okt. 2009, at 18.45, Mike van Lammeren wrote: Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using heartbeat and ldirectord for load-balancing, fail- over and high availability! I'm probably not the first one to do it, but as near as Google and I can tell, I'm the first one to write about it. Here's how you can duplicate my setup: 1. Install Ubuntu Server 8 on four machines, either real or VM. 2. Compile and install FreeSWITCH v1.0.4 from source on two machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start 3. Configure both FreeSWITCH boxes, and make sure they are both working. 4. Follow (most of) these instructions from Daniel Aliaman's blog. They were written for Asterisk, but since a SIP connection is a SIP connection, most of the document applies to FreeSWITCH: http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf The one problem I ran into was the IP address and port to which FreeSWITCH was bound. The default is to use the primary address, which works great out-of-the-box for everything else. When a client tried to register, all it got back was an ICMP error -- Destination Unreachable, Port Unreachable. That error is returned when no sockets are listening for UDP packets. To get FreeSWITCH to listen for your Virtual IP, you need to set it in two places: 5. In /opt/freeswitch/conf/vars.xml, set bind_server_ip. 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set sip-ip. That should do it! If you have any success, please report to this list. Keep in mind that if you want to do something like conferencing between two registered clients, then you have to deal with the fact that the clients may or may not be on the same box. Mike van Lammeren On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren m...@van.lammeren.net wrote: On Mon, Sep 28, 2009 at 9:05 PM, Even André Fiskvik grev...@me.com wrote: From: Even André Fiskvik grev...@me.com To: freeswitch-users@lists.freeswitch.org Date: Mon, 28 Sep 2009 22:52:13 +0200 Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey I have been working with a similar setup myself, but for some reason I ended up ditching the UltraMonkey setup because I just couldn't get it to work right. It's been quite a while since my effort, so I don't remember what the exact issue was. I got registrations to work, but had some other sip-dialog issues. We have since then changed over to running OpenSIPs as a loadbalancer in front of multiple FreeSWITCH instances. This setup is still in testing, but seemlingy works fine (and if it doesn't, it's my own fault for writing a bad opensips config). After we have done some more testing I can create a wiki-page with config details. Best regards, Even André Thanks, Even, that would be great! I might have to give up on the ultramonkey solution, since I can't find anyone who has made it work. It's too bad, because it would fit well with the rest of our architecture. Mike van Lammeren ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones
If you have time to take a look, I could put a trace in the pastebin? Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, October 01, 2009 10:29 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
can we do it without advertising to use ubuntu =D We don't like encouraging our users to use bleeding edge OS for our own sanity with debugging. Not to say you are not allowed to I just don't want to encourage it =p On Thu, Oct 1, 2009 at 1:12 PM, Even André Fiskvik grev...@me.com wrote: That's very cool Mike! I'm going to try to configure four boxes with this as well (Btw, did you use physical hardware or virtualization?) and see how it goes. I followed Daniel Aliaman's blog as well, but I can try it again with the tips you provided on FreeSWITCH config to see if I can get it working properly this time. We did the setup on CentOS, but I wouldn't think that would be any issue. Perhaps you or we could write up a complete guide about this on the wiki since this is an scenario commonly used? Also it would be great if we could outline possible issues (and even better solutions) to this kind of setup with regards to stuff like conferencing, bridging between registered users and presence. Best regards, Even André On 1. okt. 2009, at 18.45, Mike van Lammeren wrote: Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using heartbeat and ldirectord for load-balancing, fail-over and high availability! I'm probably not the first one to do it, but as near as Google and I can tell, I'm the first one to write about it. Here's how you can duplicate my setup: 1. Install Ubuntu Server 8 on four machines, either real or VM. 2. Compile and install FreeSWITCH v1.0.4 from source on two machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start3. Configure both FreeSWITCH boxes, and make sure they are both working. 4. Follow (most of) these instructions from Daniel Aliaman's blog. They were written for Asterisk, but since a SIP connection is a SIP connection, most of the document applies to FreeSWITCH: http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdfThe one problem I ran into was the IP address and port to which FreeSWITCH was bound. The default is to use the primary address, which works great out-of-the-box for everything else. When a client tried to register, all it got back was an ICMP error -- Destination Unreachable, Port Unreachable. That error is returned when no sockets are listening for UDP packets. To get FreeSWITCH to listen for your Virtual IP, you need to set it in two places: 5. In /opt/freeswitch/conf/vars.xml, set bind_server_ip. 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set sip-ip. That should do it! If you have any success, please report to this list. Keep in mind that if you want to do something like conferencing between two registered clients, then you have to deal with the fact that the clients may or may not be on the same box. Mike van Lammeren On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren m...@van.lammeren.net wrote: On Mon, Sep 28, 2009 at 9:05 PM, Even André Fiskvik grev...@me.com wrote: From: Even André Fiskvik grev...@me.com To: freeswitch-users@lists.freeswitch.org Date: Mon, 28 Sep 2009 22:52:13 +0200 Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey I have been working with a similar setup myself, but for some reason I ended up ditching theUltraMonkey setup because I just couldn't get it to work right. It's been quite a while since my effort, so I don't remember what the exact issue was. I got registrations to work, but had some other sip-dialog issues. We have since then changed over to running OpenSIPs as a loadbalancer in front of multiple FreeSWITCH instances. This setup is still in testing, but seemlingy works fine (and if it doesn't, it's my own fault for writing a bad opensips config). After we have done some more testing I can create a wiki-page with config details. Best regards, Even André Thanks, Even, that would be great! I might have to give up on the ultramonkey solution, since I can't find anyone who has made it work. It's too bad, because it would fit well with the rest of our architecture. Mike van Lammeren ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- 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
Re: [Freeswitch-users] Connecting FS to Hicom 300
You might want to try the ozmod_pri instead of ozmod_isdn until the new
revision of ozmod_isdn is published into the source tree.
On Thu, Oct 1, 2009 at 9:37 AM, russell.mosem...@cune.org wrote:
We have connected FS to a Siemens Hicomm 300. As you might guess, it's
not working right. Here is what we are working with.
Dell 1750 (dual socket, dual core Xeon 2.8GHz)
Debian 5
FS (15029), OpenZAP (without libpri)
TE110P T1 card (Zaptel driver)
Handles 71xx extensions
Siemens Hicom 300
TMDN64P T1 card
Handles 74xx extensions
We are pretty much using the stock FS configuration, yet, because we're
trying to get this to work. I have configured OpenZAP and the associated
files like the examples on the wiki (see below) to work with a PRI T1.
There are 23 B channels and 1 D channel. The Zaptel side looks fine.
OpenZAP is able to open the channels when FS boots. So far, so good.
When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta
from CounterPath on an office PC), X-Lite rings. The call can be
answered, and the conversation sounds fine. That means the routing,
registration and authorization are working on the network between X-Lite
and FS. It also means that FS is able to communicate with the Hicom over
the T1. Great.
When the caller presses the transfer button on the 74xx phone, the Hicom
sends a message over the D channel, and the call is disconnected
(watching with fs_cli). As best I can interpret the bytes in the message,
the Hicom sends a disconnect message when 74xx presses the transfer key.
In order to call 74xx, I created dialplan/default/02_hicom.xml. The
contents are
include
extension name=hicom
condition field=destination_number expression=^(74\d{2})$
action application=bridge data=openzap/1/a/$1/
/condition
/extension
/include
If a call is made from 71xx to 74xx, the Hicom forwards the call to the
switchboard with 7100-7445 connection not possible (or whatever
extensions) in the switchboard display.
1. Are these issues related to the way I have configured FS?
The Hicom is maintained by the local phone company. I do not have access
to view or configure the T1 card on the Hicom. According to the phone
guy, there isn't anything else that needs to be configured on the Hicom.
He believes that if 74xx can call 71xx, then 71xx should be able to call
74xx.
I suspect that something more needs to be done on the Hicom in order to
accept calls from FS and bridge/transfer them to a local extension on the
Hicom. It's as if the Hicom doesn't know how or is not permitted to route
incoming calls on the T1 to local extensions. I have no way to know,
though. I'm hoping someone else has connected FS to a Hicom 300 and can
provide configuration details. If I could tell the phone guy something
like, You need to look at this, that would help him out.
2. Should I receive CID/ANI from the Hicom?
X-Lite displays OpenZAP as the call and 1 as Other when the call
comes in, which is the information for the endpoint. Is there something I
need to do in the FS configuration to capture CID/ANI information from
the Hicom and make it available (or is it not being provided by the Hicom)?
3. When dialing from the Rolmphone is there a way for FS to send the
called name back to the Hicom for it to appear in the display?
When dialing 74xx to 74xx, of course, it shows the called number and name
in the display. We also have a HiPath 4000 connected to the Hicom 300.
When dialing an extension on the HiPath from the Hicom, the HiPath ships
the called name back to the Hicom for display on the phone. It would be
nice to do that from FS.
Let me know if you need additional information. Thanks for any pointers
or insight as to how things work.
--
Russell Mosemann
openzap.conf
[span zt PRI_1]
name = OpenZAP
number = 1
trunk_type = t1
b-channel = 1-23
d-channel = 24
zt.conf
[defaults]
codec_ms = 20
wink_ms = 150
flash_ms = 750
echo_cancel_level = 64
rxgain = 0.0
txgain = 0.0
openzap.conf
configuration name=openzap.conf description=OpenZAP Configuration
settings
param name=debug value=0/
!--param name=hold-music value=$${moh_uri}/--
!--param name=enable-analog-option value=call-swap/--
!--param name=enable-analog-option value=3-way/--
/settings
pri_spans
span name=PRI_1
param name=q921loglevel value=alert/
param name=q931loglevel value=alert/
param name=mode value=user/
param name=dialect value=national/
param name=dialplan value=XML/
param name=context value=public/
/span
/pri_spans
/configuration
zaptel.conf
# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER)
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24
# Global data
loadzone= us
defaultzone = us
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!
Re: [Freeswitch-users] Problem with gateway registration
Any ideas about this? The SIP provider is offering H323, but I'm not quite sure about that, is mod_opal working right? Thanks! Nicolas On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner nico...@medularis.comwrote: Anthony, thanks. Below are my config files for the two gateways from the sip trace. Both files are located in conf/directory/default. - redvoiss.xml (the one that works) include user id=gateway_redvoiss gateways gateway name=redvoiss-pp param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=pxextmy.redvoiss.net/ param name=realm value=pxextmy.redvoiss.net/ param name=proxy value=pxextmy.redvoiss.net/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2010/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - orange.xml (the one that doesn't work) include user id=gateway_orange gateways gateway name=orange param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=216.72.10.39/ param name=realm value=216.72.10.39/ param name=proxy value=216.72.10.39/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2011/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - If I remove the register=true param for the non-working gateway, I don't get the registration error on the cli, but then all call attempts get rejected with a 401 Unauthorized, and I get a hangup cause of NORMAL_UNSPECIFIED. Best, Nicolas On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale anthony.miness...@gmail.com wrote: 900 level errors are sofia internal errors so probably something is wrong with your gateway config xml. if you want to send it with any critical info replaced with XXX maybe we can see the issue for you. On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.comwrote: Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing sofia profile external restart reloadxml on the cli, captured with tshark -i eth0 -o rtp.heuristic_rtp: TRUE -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38' Thanks! Nicolas ___ 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:
Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
On Thu, 2009-10-01 at 14:14 -0500, Anthony Minessale wrote: can we do it without advertising to use ubuntu =D We don't like encouraging our users to use bleeding edge OS for our own sanity with debugging. I understand your stance, though if we're talking about Ubuntu 8.04 LTS (Long Term Support - 5 years) it's not really bleeding edge anymore. 18 months ago when it was released it may have been a little, but the LTS releases still aren't as bleeding edge as the standard support in between releases. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
Looking thru the example, it looks like each box has a real address of 21, 22 or 23 and they all have a loopback of .17, right? So even though they connections are being load balanced, each box really thinks it is .17 and each client that connects thinks it is connecting to .17, right? If thats the case, how does a client on one box call a client on the other box? Since every box thinks it is .17 how would you bridge to another user on another box that also thinks it is .17? Or am I totally missing how it works? From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, October 01, 2009 2:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey can we do it without advertising to use ubuntu =D We don't like encouraging our users to use bleeding edge OS for our own sanity with debugging. Not to say you are not allowed to I just don't want to encourage it =p On Thu, Oct 1, 2009 at 1:12 PM, Even André Fiskvik grev...@me.com wrote: That's very cool Mike! I'm going to try to configure four boxes with this as well (Btw, did you use physical hardware or virtualization?) and see how it goes. I followed Daniel Aliaman's blog as well, but I can try it again with the tips you provided on FreeSWITCH config to see if I can get it working properly this time. We did the setup on CentOS, but I wouldn't think that would be any issue. Perhaps you or we could write up a complete guide about this on the wiki since this is an scenario commonly used? Also it would be great if we could outline possible issues (and even better solutions) to this kind of setup with regards to stuff like conferencing, bridging between registered users and presence. Best regards, Even André On 1. okt. 2009, at 18.45, Mike van Lammeren wrote: Guess what? I have two FreeSWITCH servers working behind UltraMonkey, using heartbeat and ldirectord for load-balancing, fail-over and high availability! I'm probably not the first one to do it, but as near as Google and I can tell, I'm the first one to write about it. Here's how you can duplicate my setup: 1. Install Ubuntu Server 8 on four machines, either real or VM. 2. Compile and install FreeSWITCH v1.0.4 from source on two machines, following these instructions: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start 3. Configure both FreeSWITCH boxes, and make sure they are both working. 4. Follow (most of) these instructions from Daniel Aliaman's blog. They were written for Asterisk, but since a SIP connection is a SIP connection, most of the document applies to FreeSWITCH: http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf The one problem I ran into was the IP address and port to which FreeSWITCH was bound. The default is to use the primary address, which works great out-of-the-box for everything else. When a client tried to register, all it got back was an ICMP error -- Destination Unreachable, Port Unreachable. That error is returned when no sockets are listening for UDP packets. To get FreeSWITCH to listen for your Virtual IP, you need to set it in two places: 5. In /opt/freeswitch/conf/vars.xml, set bind_server_ip. 6. In /opt/freeswitch/conf/sip_profiles/internal.xml, set sip-ip. That should do it! If you have any success, please report to this list. Keep in mind that if you want to do something like conferencing between two registered clients, then you have to deal with the fact that the clients may or may not be on the same box. Mike van Lammeren On Tue, Sep 29, 2009 at 10:20 AM, Mike van Lammeren m...@van.lammeren.net wrote: On Mon, Sep 28, 2009 at 9:05 PM, Even André Fiskvik grev...@me.com wrote: From: Even André Fiskvik grev...@me.com To: freeswitch-users@lists.freeswitch.org Date: Mon, 28 Sep 2009 22:52:13 +0200 Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey I have been working with a similar setup myself, but for some reason I ended up ditching the UltraMonkey setup because I just couldn't get it to work right. It's been quite a while since my effort, so I don't remember what the exact issue was. I got registrations to work, but had some other sip-dialog issues. We have since then changed over to running OpenSIPs as a loadbalancer in front of multiple FreeSWITCH instances. This setup is still in testing, but seemlingy works fine (and if it doesn't, it's my own fault for writing a bad opensips config). After we have done some more testing I can create a wiki-page with config details. Best regards, Even André Thanks, Even, that would be great! I might have to give up on the ultramonkey solution, since I can't find anyone who has made it work. It's too bad, because it would fit well with the rest of our architecture. Mike van Lammeren
Re: [Freeswitch-users] conference participant from behind NAT
I am still experiencing problem with lost media in conference on a client behind NAT. This is what I've done - disabled VAD on a NATed client and asked my friend to produce lots of animal sounds in order to keep channel busy. But at the end of minute sounds of wild nature disapeared again. We reproduced that without security with tcp SIP transport and got the same result. Then I started to dig into SIP trace and this is what I found. This client (behind NAT) recieve subsequent INVITE message from FS which seem to destroy dialog and causes client app to close media stream after a session being established normally. I performed the same call from box with public ip and saw no subsequent INVITE's from FS. How come FS sends an INVITE message to already connected client? Is it OK? Should client handle this normally? Below is client's SIP trace: INVITE sip:1.conference...@74.208.167.44:5081;transport=TLS SIP/2.0 ... User-Agent: DoxWox SIP user agent .. SIP/2.0 100 Trying .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. SIP/2.0 407 Proxy Authentication Required .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. ACK sip:1.conference...@74.208.167.44:5081;transport=TLS SIP/2.0 .. SIP/2.0 100 Trying .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. SIP/2.0 183 Session Progress .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. SIP/2.0 200 OK .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. ACK sip:1.conference...@74.208.167.44:5081;transport=tls SIP/2.0 .. *Finally, this message cause media stream closing* INVITE sip:1...@87.184.52.45:64183;transport=tls SIP/2.0 Via: SIP/2.0/TLS 74.208.167.44:5081;branch=z9hG4bK8269NDyXQNjyH Max-Forwards: 70 From: sip:1.conference...@74.208.167.44sip%3a1.conference...@74.208.167.44;tag=vQH234QtN2U8Q To: sip:1...@74.208.167.44 sip%3a1...@74.208.167.44;tag=3a231ba86c894ceca81d5021b68d3b6c Call-ID: 37edc38329f64fe98c36cc0a6ddcbd9d CSeq: 121093810 INVITE Contact: sip:1.conference...@74.208.167.44:5081;transport=tls User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 340 v=0 o=FreeSWITCH 1254396503 1254396504 IN IP4 74.208.167.44 s=FreeSWITCH c=IN IP4 74.208.167.44 t=0 0 m=audio 27726 RTP/SAVP 103 101 a=rtpmap:103 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:J6x2UgRVUY8GfbwjCuyyttrtnXnwwuWA9Pt+o3VW ___ 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] register problem
Can someone point out what is wrong here. Thanks.
Siptrace at http://carmickle.com/fs.txt
include
gateway name=voiptalk.org
param name=username value=81100068/
param name=password value=my_pass/
param name=from-user value=81100068/
param name=from-domain value=voiptalk.org/
param name=proxy value=voiptalk.org/
param name=expire-seconds value=600/
param name=register value=true/
param name=retry-seconds value=30/
param name=extension value=1000/
param name=contact-params value=domain_name=$${domain}/
param name=context value=public/
/gateway
/include
--FC
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[Freeswitch-users] bgapi jobid to uuid
Hi, I decided to go with a linked list for current channels and maintain that through state changes. So, basically it works like this: 1. Originate a call using bgapi (we get a jobid in response) 2. Receive an event with the jobid and a uuid 3. Lookup the linked list for the jobid, set the uuid 4. Receive hangup etc (with uuid) remove from linked list The problem is that sometimes I receive the hangup with the uuid before I'm told what the match between the jobid and the uuid are. Any ideas? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Connecting FS to Hicom 300
Anthony Minessale anthony.miness...@gmail.com said: You might want to try the ozmod_pri instead of ozmod_isdn until the new revision of ozmod_isdn is published into the source tree. libpri took care of the problem with the transfer. Now, someone can call into FS from the Hicomm and then transfer the call to another extension on the Hicomm. A call from FS to the Hicomm still transfers to the switchboard. I'm not seeing any CID/ANI on the X-Lite. It shows up as OpenZAP and 1. Is there something like ngrep for the D channel of a PRI? It would be nice to see what data is being sent between FS and the Hicom. -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference participant from behind NAT
And here is a short piece of log from the server side: ... nua(): refersh session after 62 seconds (in [55..65])... send INVITE ... rcv OK... send ACK... rcv BYE... I see now that sdp for natted client has additional lines in OK response compared to client with public ip. Session-Expires: 120;refresher=uas Min-SE: 120 How come that they differs? And how do I resolve this situation? Should client handle these refresher messages normally? Best regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] bgapi jobid to uuid
if you pick your own job-uuid you can set it in the originate too
send
job-uuid: 1234
in your bgapi event
and
{job_uuid=1234}sofia/internal/f...@bar.com
in your dial string.
*shrug* i am not sure exactly what the goal is so maybe it's not a useful
suggestion...
On Thu, Oct 1, 2009 at 4:36 PM, Matt Riddell li...@venturevoip.com wrote:
Hi,
I decided to go with a linked list for current channels and maintain
that through state changes.
So, basically it works like this:
1. Originate a call using bgapi (we get a jobid in response)
2. Receive an event with the jobid and a uuid
3. Lookup the linked list for the jobid, set the uuid
4. Receive hangup etc (with uuid) remove from linked list
The problem is that sometimes I receive the hangup with the uuid before
I'm told what the match between the jobid and the uuid are.
Any ideas?
--
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Matt Riddell
Director
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googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
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Re: [Freeswitch-users] Connecting FS to Hicom 300
there was a feature to generate a pcap from the debug logs but i forgot who posted it. On Thu, Oct 1, 2009 at 4:19 PM, russell.mosem...@cune.org wrote: Anthony Minessale anthony.miness...@gmail.com said: You might want to try the ozmod_pri instead of ozmod_isdn until the new revision of ozmod_isdn is published into the source tree. libpri took care of the problem with the transfer. Now, someone can call into FS from the Hicomm and then transfer the call to another extension on the Hicomm. A call from FS to the Hicomm still transfers to the switchboard. I'm not seeing any CID/ANI on the X-Lite. It shows up as OpenZAP and 1. Is there something like ngrep for the D channel of a PRI? It would be nice to see what data is being sent between FS and the Hicom. -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- 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] bgapi jobid to uuid
On 2/10/09 10:50 AM, Anthony Minessale wrote:
if you pick your own job-uuid you can set it in the originate too
send
job-uuid: 1234
in your bgapi event
and
{job_uuid=1234}sofia/internal/f...@bar.com mailto:f...@bar.com
in your dial string.
*shrug* i am not sure exactly what the goal is so maybe it's not a
useful suggestion...
Does the job-uuid get used as the actual call uuid? The problem is that
the uuid for the call changes once it's answered (or hungup etc) and
that the mapping doesn't come till after occasionally.
I'm getting the job uuid no problem, but it's the replacement of the
job-uuid with a call-uuid that is the issue.
If this doesn't make too much sense I can post an example.
--
Cheers,
Matt Riddell
Director
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Re: [Freeswitch-users] bgapi jobid to uuid
if you make your own uuids you could set them in the originate string with
{origination_uuuid=foo}
where foo is a real uuid.
if you have no other way to make them you can ask FS for one with the
create_uuid api call.
On Thu, Oct 1, 2009 at 5:02 PM, Matt Riddell li...@venturevoip.com wrote:
On 2/10/09 10:50 AM, Anthony Minessale wrote:
if you pick your own job-uuid you can set it in the originate too
send
job-uuid: 1234
in your bgapi event
and
{job_uuid=1234}sofia/internal/f...@bar.com mailto:f...@bar.com
in your dial string.
*shrug* i am not sure exactly what the goal is so maybe it's not a
useful suggestion...
Does the job-uuid get used as the actual call uuid? The problem is that
the uuid for the call changes once it's answered (or hungup etc) and
that the mapping doesn't come till after occasionally.
I'm getting the job uuid no problem, but it's the replacement of the
job-uuid with a call-uuid that is the issue.
If this doesn't make too much sense I can post an example.
--
Cheers,
Matt Riddell
Director
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Re: [Freeswitch-users] Connecting FS to Hicom 300
On Thu, Oct 1, 2009 at 2:19 PM, russell.mosem...@cune.org wrote: Anthony Minessale anthony.miness...@gmail.com said: You might want to try the ozmod_pri instead of ozmod_isdn until the new revision of ozmod_isdn is published into the source tree. libpri took care of the problem with the transfer. Now, someone can call into FS from the Hicomm and then transfer the call to another extension on the Hicomm. A call from FS to the Hicomm still transfers to the switchboard. I'm not seeing any CID/ANI on the X-Lite. It shows up as OpenZAP and 1. Is there something like ngrep for the D channel of a PRI? It would be nice to see what data is being sent between FS and the Hicom. I believe the OpenZAP and 1 are coming from your conf file: openzap.conf [span zt PRI_1] name = OpenZAP number = 1 As far as debugging with ozmod_libpri I believe the syntax is: oz libpri debug 1 all It will do a traditional libpri-style debug, just like pri debug span 1 in Asterisk. -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] bgapi jobid to uuid
On Thu, Oct 1, 2009 at 2:36 PM, Matt Riddell li...@venturevoip.com wrote: Hi, I decided to go with a linked list for current channels and maintain that through state changes. So, basically it works like this: 1. Originate a call using bgapi (we get a jobid in response) 2. Receive an event with the jobid and a uuid 3. Lookup the linked list for the jobid, set the uuid 4. Receive hangup etc (with uuid) remove from linked list The problem is that sometimes I receive the hangup with the uuid before I'm told what the match between the jobid and the uuid are. Any ideas? If I may ask, what's the application? Are you working on Vicidial-ish stuff for FreeSWITCH? Also, you might want to call the FreeSWITCH conference tomorrow. We have some QA time with the FS devs and this kind of thing might work better in realtime rather than email threads. -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] bgapi jobid to uuid
On 2/10/09 11:36 AM, Anthony Minessale wrote:
if you make your own uuids you could set them in the originate string with
{origination_uuuid=foo}
where foo is a real uuid.
if you have no other way to make them you can ask FS for one with the
create_uuid api call.
Awesome, thanks - will give it a whirl over the weekend.
--
Cheers,
Matt Riddell
Director
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Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones
Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.comwrote: If you have time to take a look, I could put a trace in the pastebin? Jerry -- *From:* Jerry Richards [mailto:jerry.richa...@teotech.com] *Sent:* Thursday, October 01, 2009 10:29 AM *To:* 'freeswitch-users@lists.freeswitch.org' *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry -- *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com] *Sent:* Thursday, October 01, 2009 9:36 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org 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] Minimum audio length for uuid_record
Thanks, I think I found the thread you were referring to ([Freeswitch-users] session record does not for very short calls), which doesn't seem to be a solution for my situation. However, I did find that using session:recordFile() didn't delete the file if it was really short. And following that thread lead me to an interesting channel variable that could be useful to us, record_ms, but having trouble getting it to reflect the audio length. I can see the variable when printing data from the info application, but it's always 0. My snippet of code is very simple: if session:ready() then session:recordFile(C:/Temp/recording.wav, 30, 600, 6); local record_length = session:getVariable(record_ms); freeswitch.consoleLog(INFO, Recorded a .. record_length .. ms file.\n); end Another side question, the silence secs parameter (in this example, 6), is that 6s silence hits during the entire recording session or 6s of consecutive silence? From a few tests, it seems to be the former, but just wanted to verify, something that would make a good addition to the wiki. Dan On Wed, Sep 30, 2009 at 11:31 PM, Jason White ja...@jasonjgw.net wrote: Dan Le dule.maill...@gmail.com wrote: We're running into a problem with the minimum file size when recording using uuid_record. It seems if the audio is too short it deletes the audio file. Is there a way to override that? Yes. It was discussed on the list recently. I suggest searching the list archives. Someone may have documented it on the wiki by now also. ___ 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
