Re: [Freeswitch-users] any way ring fifo members one by one?
seven ha scritto: oh, thank you Antonio. I think it would be better to collect more ideas before open a bounty. And I more interested in playing(including patching the code) with that than use the function. I was working on other stuff yesterday and just looked at the wiki: - it seems there is already a bounty for something like that; - there is a wiki page about how to implement it with Javascript, ofc you need to tailor it to your own needs; AgX ___ 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] dynamically change the presence status (half solved + probably found a bug)
I looked at FS code directly and had an improvement about the problem. Topic 1: presence can be changed and working The wiki page: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_presence is totally wrong about the presence application, indeed when i first asked on IRC some weeks ago i get confused. On my user page, http://wiki.freeswitch.org/wiki/User:Agx, i posted some working example. I tested using X-Lite and Zoiper from my home connecting to FS behind NAT in the office. Topic 2: is this a BUG? 1. phone A register and subsribe about presence of B 2. phone B register and A get a notified that B is online (IN) 3. phone B unregister and A get notified that B is offline (OUT) 4. phone B register and A *NEVER* get notified that B is online UNTIL A register/resubscribe or A reboots is FS that remove A subscribtions when B generate a presence OUT event? is this a bug? Topic 3: presence seems stateless if is not a phone 1. phone A subscribe about presence of t...@domain (not a phone) and about b...@domain (a phone) 2. i change t...@domain status to busy (using the presence tool) and A get notified 3. i start a call from b...@domain and A get notified 4. i reboot/unregister A 5. i restart/register A then A get notified about b...@domain still being on the phone but not about t...@domain been busy i looked in the code and seems the PRESENCE_PROBE stuffs looks about the internal database of registrations to have t...@domain been persistence across FS restart/reboot do i need to create a module myself that handle presence using a database? Thanks in advance, Antonio (AGX) ___ 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] any way ring fifo members one by one?
Hi, It should be easy to modify mod_fifo to include this functionality. Correct me if I'm wrong: For call back agents at least, when X calls are in the the queue, Freeswitch tries to search for up to X agents in database. This algorithm is much more optimized than Asterisk, as Asterisk will take calls one by one and try to connect them to an agent, it should then stay as it is. The simplest idea to control the call distribution algorithm would be to modify the database query in the find_consumers function (right now, the algorithm is: order by outbound_call_count). A variable could control the order by of this query, and the problem would be solved at least for call back agents. I guess sqlite3 should allow very complex queries, but I don't know if there could be performance issues. Do you think it is a possible -trivial- solution? François. On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote: seven ha scritto: oh, thank you Antonio. I think it would be better to collect more ideas before open a bounty. And I more interested in playing(including patching the code) with that than use the function. I was working on other stuff yesterday and just looked at the wiki: - it seems there is already a bounty for something like that; - there is a wiki page about how to implement it with Javascript, ofc you need to tailor it to your own needs; AgX ___ 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] Compact, fanless appliance?
EdPimentl wrote: Here is a list of the resources posted on this thread After giving it more thoughts, I got to the conclusion that I'd rather a stand-alone miniPC that can take a PCI card with a riser, instead of a really tiny box that relies on an external box to connect to a PSTN line. Mini-box offers a $120 kit that includes an Intel D945GCLF mobo and a pico-PSU. All it misses, is RAM and some mass storage. Does someone know if there's some kind of CompactFlash that can connect to an IDE port? That would save space + noise. Thank you. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23295672.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] Compact, fanless appliance?
By IDE, I'm assuming you mean the all too familiar 40 pin PATA. Yes, as a matter of fact: http://www.newegg.com/Product/ProductList.aspx?Submit=ENEN=2003240636+1421530855Configurator=Subcategory=636description=Ntk=SpeTabStoreType=srchInDesc= Do some research since they're not all created equal. I'm not sure what kind of MTBF you're looking for. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Apr 29, 2009, at 7:47 AM, Fred-145 wrote: EdPimentl wrote: Here is a list of the resources posted on this thread After giving it more thoughts, I got to the conclusion that I'd rather a stand-alone miniPC that can take a PCI card with a riser, instead of a really tiny box that relies on an external box to connect to a PSTN line. Mini-box offers a $120 kit that includes an Intel D945GCLF mobo and a pico-PSU. All it misses, is RAM and some mass storage. Does someone know if there's some kind of CompactFlash that can connect to an IDE port? That would save space + noise. Thank you. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23295672.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 PGP.sig Description: This is a digitally signed message part ___ 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] dynamically change the presence status (half solved + probably found a bug)
On Wed, Apr 29, 2009 at 2:01 AM, Antonio Gallo ga...@mctelefonia.comwrote: I looked at FS code directly and had an improvement about the problem. Topic 1: presence can be changed and working The wiki page: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_presence is totally wrong about the presence application, indeed when i first asked on IRC some weeks ago i get confused. On my user page, http://wiki.freeswitch.org/wiki/User:Agx, i posted some working example. I tested using X-Lite and Zoiper from my home connecting to FS behind NAT in the office. Topic 2: is this a BUG? 1. phone A register and subsribe about presence of B 2. phone B register and A get a notified that B is online (IN) 3. phone B unregister and A get notified that B is offline (OUT) 4. phone B register and A *NEVER* get notified that B is online UNTIL A register/resubscribe or A reboots is FS that remove A subscribtions when B generate a presence OUT event? is this a bug? It depends, not every phone does presence right and they all have different levels of support. The 2 you are using are very different from how the majority of them work (snom, polycom, linksys etc) Did you try reversing the role of A and B Topic 3: presence seems stateless if is not a phone 1. phone A subscribe about presence of t...@domain (not a phone) and about b...@domain (a phone) 2. i change t...@domain status to busy (using the presence tool) and A get notified 3. i start a call from b...@domain and A get notified 4. i reboot/unregister A 5. i restart/register A then A get notified about b...@domain still being on the phone but not about t...@domain been busy i looked in the code and seems the PRESENCE_PROBE stuffs looks about the internal database of registrations to have t...@domain been persistence across FS restart/reboot do i need to create a module myself that handle presence using a database? YES Thanks in advance, Antonio (AGX) ___ 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/ 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] dynamically change the presence status (half solved + probably found a bug)
It depends, not every phone does presence right and they all have different levels of support. The 2 you are using are very different from how the majority of them work (snom, polycom, linksys etc) Did you try reversing the role of A and B I will pickup a snom320 and a gxp 20xx and will try Topic 3: presence seems stateless if is not a phone to create a module myself that handle presence using a database? YES I looked at source code of conferece/fico (that generate presence from XML config) and dingaling (who use database). So i think its not difficult to write it. Hope to have free time :) Thank you, Antonio (AGX) ___ 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] Condition and custom variables
Hello.
Can I use custom variables in the condition field?
For example if i set my custom var ${forward_all} and continue search extensions
condition field=${forward_all} .../ not work even if the condition is met
and call does not go in the uplink.
Or, in the condition I can use only variables freeswitch like ${sip_from_user}?
Is it possible to use custom variables?
extension name=set-vars continue=true
condition
action application=set data=forward_all=123456789/
/condition
/extension
extension name=call
condition field=${forward_all} expression=^(\d{8,9,10})$ break=on-true
...
action application=bridge data=sofia/gateway/uplink/${forward_all}/
action application=hangup/
/condition
condition field=destination_number expression=^(.*)$
...
action application=bridge data=user/${dialed_extensi...@${domain_name}/
action application=hangup/
/condition
/extension
--
Alex Gusak
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Re: [Freeswitch-users] Compact, fanless appliance?
On Wed, 2009-04-29 at 04:47 -0700, Fred-145 wrote: EdPimentl wrote: Here is a list of the resources posted on this thread After giving it more thoughts, I got to the conclusion that I'd rather a stand-alone miniPC that can take a PCI card with a riser, instead of a really tiny box that relies on an external box to connect to a PSTN line. Mini-box offers a $120 kit that includes an Intel D945GCLF mobo and a pico-PSU. All it misses, is RAM and some mass storage. Does someone know if there's some kind of CompactFlash that can connect to an IDE port? That would save space + noise. http://www.google.com/products?q=ide+to+compact+flashscoring=p Regards, Cliff ___ 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] Condition and custom variables
OK let me explain this for the 100th time. :)
The dialplan isn't executed line by line. The dialplan is just a
compiled list of instructions to execute but its execution doesn't
take place until the session enters the execute state. So your set
forward_all= hasn't happened yet. BUT you can use
execute_extension or transfer to send the session back into the
dialplan to re-evaluate the the list of things to do based on certain
conditions you previously set.
/b
On Apr 29, 2009, at 9:30 AM, Alex Gusak wrote:
Hello.
Can I use custom variables in the condition field?
For example if i set my custom var ${forward_all} and continue
search extensions
condition field=${forward_all} .../ not work even if the
condition is met
and call does not go in the uplink.
Or, in the condition I can use only variables freeswitch like $
{sip_from_user}?
Is it possible to use custom variables?
extension name=set-vars continue=true
condition
action application=set data=forward_all=123456789/
/condition
/extension
extension name=call
condition field=${forward_all} expression=^(\d{8,9,10})$
break=on-true
...
action application=bridge data=sofia/gateway/uplink/$
{forward_all}/
action application=hangup/
/condition
condition field=destination_number expression=^(.*)$
...
action application=bridge data=user/${dialed_extensi...@$
{domain_name}/
action application=hangup/
/condition
/extension
--
Alex Gusak
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] Condition and custom variables
There should be several archive threads on this including:
http://n2.nabble.com/Using-Variables-in-Dialplans-tt2678222.html#a2678222
Mike
On Apr 29, 2009, at 10:30 AM, Alex Gusak wrote:
Hello.
Can I use custom variables in the condition field?
For example if i set my custom var ${forward_all} and continue
search extensions
condition field=${forward_all} .../ not work even if the
condition is met
and call does not go in the uplink.
Or, in the condition I can use only variables freeswitch like $
{sip_from_user}?
Is it possible to use custom variables?
extension name=set-vars continue=true
condition
action application=set data=forward_all=123456789/
/condition
/extension
extension name=call
condition field=${forward_all} expression=^(\d{8,9,10})$
break=on-true
...
action application=bridge data=sofia/gateway/uplink/$
{forward_all}/
action application=hangup/
/condition
condition field=destination_number expression=^(.*)$
...
action application=bridge data=user/${dialed_extensi...@$
{domain_name}/
action application=hangup/
/condition
/extension
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[Freeswitch-users] Very confusing startup error
All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitchfreeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag /context] However, none of the files in conf have a tag called /context. All files are conforming xml. I can't seem to find what's changed. Any ideas? ___ 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] Very confusing startup error
The first is an error that is unrelated to the second error. Check out freeswitch.xml.fsxml line 2423 you'll have an extra / context line there. /b On Apr 29, 2009, at 10:21 AM, Gerry Hull wrote: All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitchfreeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\.. \sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag / context] However, none of the files in conf have a tag called /context. All files are conforming xml. I can't seem to find what's changed. Any ideas? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Very confusing startup error
At least your dialplan should have a tag named context. See default dialplan ! Original Messageprocessed by David.InfoCenter Subject: [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26) From:Gerry Hull ge...@pstn2.net To: g...@exram.de All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitchfreeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag /context] However, none of the files in conf have a tag called /context. All files are conforming xml. I can't seem to find what's changed. Any ideas? ___ 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] Very confusing startup error
Thanks Guys! I could not find my problem -- but you pointed me in the correct direction. I had a mismatched tag in my public.xml in the dialpan. So, is freeswitch.xml.fsxml a logged representation of the complete config file in memory? On Wed, Apr 29, 2009 at 11:36 AM, Guido Kuth g...@exram.de wrote: At least your dialplan should have a tag named context. See default dialplan ! Original Message * processed by David.InfoCenter* Subject: [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26) From: Gerry Hull ge...@pstn2.net ge...@pstn2.net To: g...@exram.de All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitchfreeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag /context] However, none of the files in conf have a tag called /context. All files are conforming xml. I can't seem to find what's changed. Any ideas? ___ 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] HELP 3-way network access
Hi guys .. I need your help please... I am trying to setup an FS box. It has to be like a 3 way thing since i reside in one network - my FS machine resides on another network - and my provider (gateway) resides on another network. I am going to try to be specific as much as i can. I did a Quick and Dirty install. Here is my testing servers info. Hardware Intel Dual Core 2.6 GHZ Real memory 3.56 GB total, 267.71 MB used Hard Drives 1 SATA 250GB MotherBoard Biostar P4M900-M4 Motherboard - VIA P4M900, Socket 478, MicroATX Software Operating system CentOS Linux 5.2 Kernel and CPU Linux 2.6.18-92.1.22.el5 on i686 Apache2.2.3 MySQL5.0.45 SSHOpenSSH 4.3 freeswi...@internal version FreeSWITCH Version 1.0.trunk (13181M) Ok the FS testing server resides on xxx.9.10.xxx. The gateway resides on xxx.9.9.xxx. And my computer resides on 75.74.xxx.xxx (My computer has X-lite) installed. When i create the SIP profile on X-Lite in my computer and tell X-Lite to register on xxx.9.10.xxx It says discovering network ... Initializing... Registering ... And then it shows up your Your username is: 1000 (looks like it is registered). Now when i try to dial 5000 to listen at least to the IVR demo i get ... The person you are calling is unavailable please try again ... message. and shows on the top of the username ... Call failed: Request Timeout (message) And on the fs_cli console it shows this .. freeswi...@internal 2009-04-29 11:46:05 [DEBUG] sofia.c:4242 sofia_handle_sip_i_invite() IP 75.74.xxx.xxx Rejected by acl domains. Falling back to Digest auth. I am a total noob on this. I replaced my original acl.conf.xml with this... configuration name=acl.conf description=Network Lists network-lists list name=test1 default=deny node type=allow cidr=75.74.xxx.0/24/ /list /network-lists /configuration I shutdown FS and then restart FS with the -nc option. And Still the same thing. My gateway is a CANTATA switch which does not require authentication. I am trying to generate a call from my 75.74.xxx.xxx using X-Lite to a PSTN phone on the outside using the CANTATA switch on xxx.9.9.xxx through my FS box on xxx.9.10.xxx But as for now I can't even get the IVR to work for now ... Since i don't know anything about FS i would like to know what am i doing wrong.. And what files have to be either created or updated to do this. Thanks to everyone for all the help Edward ___ 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] is there something like flash operator panel for freeswitch?
ok i did some test today using the yesterday's trunk with a gxp2010 and a snom360 both with 2 LEDS monitoring each other and themselves. Configuration: gxp2010 user: 1000 led1: 1000 led2: 1001 snom360 user: 1001 led1: 1000 led2: 1001 Problem with both phones: - when a phone reboot and it subscribe it does not get notified of the current status of the subscribed phones i.e. if gxp is on the phone the snom led1 is off/unlit i.e. if snom is on the phone the gxp led2 is off/unlit Problem with GXP only: - both subscribe LED stop working after a 1 or 2 calls until the gxp re-subscribe or re-register To skip using LED on phones is there is something like flash operator panel to display telephone status? Thanks in advance, Antonio (AGX) ___ 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] HELP 3-way network access
Now you need to open up the sofia profile in sip_profile/internal.xml and apply the test1 acl instead of the domains acl. /b On Apr 29, 2009, at 11:12 AM, Edward Q. wrote: freeswi...@internal 2009-04-29 11:46:05 [DEBUG] sofia.c:4242 sofia_handle_sip_i_invite() IP 75.74.xxx.xxx Rejected by acl domains. Falling back to Digest auth. I am a total noob on this. I replaced my original acl.conf.xml with this... configuration name=acl.conf description=Network Lists network-lists list name=test1 default=deny node type=allow cidr=75.74.xxx.0/24/ /list /network-lists /configuration Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Very confusing startup error
On Wed, Apr 29, 2009 at 8:54 AM, Gerry Hull ge...@pstn2.net wrote: Thanks Guys! I could not find my problem -- but you pointed me in the correct direction. I had a mismatched tag in my public.xml in the dialpan. So, is freeswitch.xml.fsxml a logged representation of the complete config file in memory? Affirmative. :) -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] First Linux soft phone running TLS/SRTP on Linux (Zoipe Bizz)
After 6 months of discussions with Attractel, today we finally got a new version of Zoiper Bizz, which works with TLS and SRTP (previous versions only supported TLS). I have added the info, how to set it up, in the wiki http://wiki.freeswitch.org/wiki/Interop_List#Zoiper_Bizz_2.10_and_TLS.2FSRTP We've been searching for a long time to have a working secure VoIP client under Linux. So far Zoiper seems to be the only VoIP soft phone capable of managing TLS/SRTP with Freeswitch under Linux. BTW: The free version does not support encryption. The Zoiper Bizz version does, but is not for free. Best regards 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
[Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL
I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1...@ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Any help or suggestion is welcome. Thanks in advance...Guido ___ 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] Serious Problem detecting DTMF in bridged SIP call using ESL
If you subscribe to the event you will receive one on every DTMF press if FreeSWITCH gets it... if you happen to be getting them via inband you won't receive an event unless you enable the inband detection app. http://wiki.freeswitch.org/wiki/Event_list#DTMF http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf I also highly recommend you update to SVN trunk. /b On Apr 29, 2009, at 12:21 PM, Guido Kuth wrote: What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Serious Problem detecting DTMF in bridged SIP call using ESL
set the async flag on the socket app call that triggers your ESL connection On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth g...@exram.de wrote: I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1...@ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Any help or suggestion is welcome. Thanks in advance...Guido ___ 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/ 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] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL
First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to info? I think setting it to rfc2833 would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Original Messageprocessed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:34) From:Brian West br...@freeswitch.org To: g...@exram.de If you subscribe to the event you will receive one on every DTMF press if FreeSWITCH gets it... if you happen to be getting them via inband you won't receive an event unless you enable the inband detection app. http://wiki.freeswitch.org/wiki/Event_list#DTMF http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf I also highly recommend you update to SVN trunk. /b On Apr 29, 2009, at 12:21 PM, Guido Kuth wrote: What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL
Hello Anthony, sorry, but I forgot to tell you that I have an inbound ESL connection not an outbound one. So I connect to FS and then wait for Events. I know that I can set async flag in outbound socket, but is this also possible for inbound socket, and when, is it the same as in outbound socket behind the IP-Address? Thank you very much...Guido Original Messageprocessed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:47) From:Anthony Minessale anthony.miness...@gmail.com To: g...@exram.de set the async flag on the socket app call that triggers your ESL connection On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth g...@exram.de wrote: I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1...@ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Any help or suggestion is welcome. Thanks in advance...Guido ___ 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/ 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] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL
Well the best option is to NOT use inband at all if possible. And use RFC2833 which eyebeam/xlite support as do most providers out there... You do not HAVE to start_dtmf on sip channels unless they only send the DTMF inband. set the dtmf-type back to rfc2833 and restart FS. /b On Apr 29, 2009, at 12:48 PM, Guido Kuth wrote: First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to info? I think setting it to rfc2833 would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Phones become unreachable after some time
I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably 30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here. ___ 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] Re-2: Re: Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL
Thank you again Brian. The reason why I want to test with inband dtmf is that in the real environment FS will be behind a conventional ISDN PBX which will work as a gateway to the ISDN Network. So I do not know if the PBX will do something like a translation between DTMF Tones to rfc and backwars. If you have experience with this any help will be very welcomeGuido Original Message processed by David InfoCenter Subject: Re: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:59) From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Well the best option is to NOT use inband at all if possible. And use RFC2833 which eyebeam/xlite support as do most providers out there... You do not HAVE to start_dtmf on sip channels unless they only send the DTMF inband. set the dtmf-type back to rfc2833 and restart FS. /b On Apr 29, 2009, at 12:48 PM, Guido Kuth wrote: First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_ dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to info? I think setting it to rfc2833 would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Phones become unreachable after some time
Do the phones and FS have a firewall between them? If so, sounds like the pin hole in the fw is being closed. Alot only stay open for 4 mins Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of paul.degt Sent: 29 April 2009 20:15 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Phones become unreachable after some time I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably 30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here. ___ 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] Phones become unreachable after some time
They do, but all necessary ports for FS are open. If that is fw issue, are there ways to fight with it? Nik Middleton wrote: Do the phones and FS have a firewall between them? If so, sounds like the pin hole in the fw is being closed. Alot only stay open for 4 mins Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of paul.degt Sent: 29 April 2009 20:15 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Phones become unreachable after some time I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably 30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here. ___ 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] Phones become unreachable after some time
Don't know where the setting is in FS, but force them to register every 120 seconds and see if that helps Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of paul.degt Sent: 29 April 2009 20:50 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Phones become unreachable after some time They do, but all necessary ports for FS are open. If that is fw issue, are there ways to fight with it? Nik Middleton wrote: Do the phones and FS have a firewall between them? If so, sounds like the pin hole in the fw is being closed. Alot only stay open for 4 mins Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of paul.degt Sent: 29 April 2009 20:15 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Phones become unreachable after some time I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably 30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here. ___ 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] is there something like flash operator panel for freeswitch?
None that I know of, but it should be fairly simple to create FOP for FS with the event socket. 2009/4/29 Antonio Gallo ga...@mctelefonia.com ok i did some test today using the yesterday's trunk with a gxp2010 and a snom360 both with 2 LEDS monitoring each other and themselves. Configuration: gxp2010 user: 1000 led1: 1000 led2: 1001 snom360 user: 1001 led1: 1000 led2: 1001 Problem with both phones: - when a phone reboot and it subscribe it does not get notified of the current status of the subscribed phones i.e. if gxp is on the phone the snom led1 is off/unlit i.e. if snom is on the phone the gxp led2 is off/unlit Problem with GXP only: - both subscribe LED stop working after a 1 or 2 calls until the gxp re-subscribe or re-register To skip using LED on phones is there is something like flash operator panel to display telephone status? Thanks in advance, Antonio (AGX) ___ 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] is there something like flash operator panel for freeswitch?
On Wed, Apr 29, 2009 at 2:13 PM, Mikael Bjerkeland mik...@bjerkeland.comwrote: None that I know of, but it should be fairly simple to create FOP for FS with the event socket. The fairly simple part is actually doing it. The really hard part is finding the time/energy/inclination to do it... -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] is there something like flash operator panel for freeswitch?
edit autoload_configs/sofia.conf.xml in global_settings add param name=debug-presence value=2/ then you will see all the sql stmts etc and you can debug your issue On Wed, Apr 29, 2009 at 11:18 AM, Antonio Gallo ga...@mctelefonia.comwrote: ok i did some test today using the yesterday's trunk with a gxp2010 and a snom360 both with 2 LEDS monitoring each other and themselves. Configuration: gxp2010 user: 1000 led1: 1000 led2: 1001 snom360 user: 1001 led1: 1000 led2: 1001 Problem with both phones: - when a phone reboot and it subscribe it does not get notified of the current status of the subscribed phones i.e. if gxp is on the phone the snom led1 is off/unlit i.e. if snom is on the phone the gxp led2 is off/unlit Problem with GXP only: - both subscribe LED stop working after a 1 or 2 calls until the gxp re-subscribe or re-register To skip using LED on phones is there is something like flash operator panel to display telephone status? Thanks in advance, Antonio (AGX) ___ 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/ 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
