Re: [Freeswitch-users] IAX configurations
I have loaded mod_iax now that error didn't come. But, When I call I have received following message in the console. [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: FACILITY_REJECTED What configuration I missed? How to use sip to connect the Asterisk? Please give solutions above questions... __ Velusamy On Mon, Jul 27, 2009 at 11:11 AM, Michael Jerris m...@jerris.com wrote: mod_iax isn't loaded. I suggest using sip anyways. Mike On Jul 27, 2009, at 1:23 AM, velusamy velu wrote: Dear All, I have tried to call a Asterisk extension from FreeSWITCH. I have done the following configurations, * I have enabled mod_iax module in modules.conf.xml file. * Next I have configure following extension in dialplan. extension name=voipjet condition field=destination_number expression=^(222)$ action application=bridge data=iax/222:2...@192.168.6.94/$1/ /condition /extension * Next I have configured a 222 user in sip.conf file at Asterisk machine. * I wrote dialplan for that extension in extension.conf file. When I tried to call 222 from FreeSWITCH, I have received following error in Console. [ERR] switch_core_session.c:255 switch_core_session_outgoing_channel() Could not locate channel type iax What would be the problem? Is there any configuration I missed? Please help me . Regards, K.Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] Multiple gateways: Priority the first bridge with prefix
Hi FS Users, I just want to try multiple gateways. It works actually like this... action application= bridge data=sofia/sip/5...@222.333.444.555 | sofia/sip/6...@111.222.333./ But I test call like 513 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674279p24674279.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] Multiple gateways: Priority the first bridge with prefix
Hi FS Users, I just want to try multiple gateways. It works actually like this... action application= bridge data=sofia/sip/5...@222.333.444.555 | sofia/sip/6...@111.222.333./ But I test call like 513 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.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] Multiple gateways: Priority the first bridge with prefix
Hi FS Users, I just want to try multiple gateways. It works actually like this... action application= bridge data=sofia/sip/5...@222.333.444.555 | sofia/sip/6...@111.222.333./ But I test call like 513 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674381p24674381.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] Multiple gateways: Priority the first bridge with prefix
in my implementation, i would use 2 separate conditions that looks like this: condition field=destination_number data=(51\d*)\ action application=bridge data=sofia/sip/$...@222.333.444.555/ condition field=destination_number data=(63\d*)\ action application=bridge data=sofia/sip/$...@111.222.333.333/ On Mon, Jul 27, 2009 at 2:42 PM, Edmar Cruz darklio...@yahoo.com wrote: Hi FS Users, I just want to try multiple gateways. It works actually like this... action application= bridge data=sofia/sip/5...@222.333.444.555 | sofia/sip/6...@111.222.333./ But I test call like 513 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.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] Dial plan contexts
Has anything changed in the handling of dial plan contexts recently? As of rev. 14363, the context setting in the Sofia profile seems to be overriding the context setting in the user's definition in the directory. As per the default configuration, I have user-context set to public in my internal profile, my user has its context set to default, but calls made from the phone registered to that user ID end up in public context when they reach the dial plan. Either something has changed or there's something wierd in my configuration that I haven't tracked down. I haven't made any changes to any of the profiles or users recently, though, and it was working under an older revision. ___ 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] Limit is not working when originate a call
Anthony Minessale wrote:
limit is for inbound calls
you cannot call it after you already made the call.
The correct approach would be to not make the call at all.
you could maybe use the limit FSAPI interface with apiExecute to check
if the limit was exceeded and
then not bother to place the call to begin with.
otherwise it's sort of like putting a prisoner in the electric chair
then giving him his trial.
Can you tell me how to do that?
I set the limit as:
action application=limit_hash data=${destination_number}
${destination_number} 1 /
Now, how do I know what is the current limit of ${destination_number}
Can you give me a JS (or lua) example?
Thanks
Eli
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Re: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix
Not working just the same both of them are running Nandy Dagondon wrote: in my implementation, i would use 2 separate conditions that looks like this: condition field=destination_number data=(51\d*)\ action application=bridge data=sofia/sip/$...@222.333.444.555/ condition field=destination_number data=(63\d*)\ action application=bridge data=sofia/sip/$...@111.222.333.333/ On Mon, Jul 27, 2009 at 2:42 PM, Edmar Cruz darklio...@yahoo.com wrote: Hi FS Users, I just want to try multiple gateways. It works actually like this... action application= bridge data=sofia/sip/5...@222.333.444.555 | sofia/sip/6...@111.222.333./ But I test call like 513 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.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 -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674260p24675073.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] Multiple gateways: Priority the first bridge with prefix
Edmar Cruz darklio...@yahoo.com wrote: Not working just the same both of them are running Do you have them as separate extensions in the dial plan? ___ 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] Multiple gateways: Priority the first bridge with prefix
ed, i mean you use separate extension names: extension name=prefix-51 condition field=destination_number data=(51\d*)\ action application=bridge data=sofia/sip1/$...@222.333.444.555/ /extension extension name=prefix-63 condition field=destination_number data=(63\d*)\ action application=bridge data=sofia/sip2/$...@111.222.333.333/ /extension btw, you should also use separate gateway names sip1 and sip2. so differentiate them in the bridge application. On Mon, Jul 27, 2009 at 4:16 PM, Jason White ja...@jasonjgw.net wrote: Edmar Cruz darklio...@yahoo.com wrote: Not working just the same both of them are running Do you have them as separate extensions in the dial plan? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Question: Capturing VoiceData received from Sagnoma E1 card
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
any ideas?
regards
Helmut
On 21.07.2009 16:01, Helmut Kuper wrote:
Hello,
For outgoing calls I'm hunting the cause for missing some 100ms of voice
data send from remote right after pickup the remote phone (e.g. initial
Hello? sound like o? or even nothing)
On FreeSwitch server I captured the VoIP data to the called VoIP-Phone
on the sofia interface. Using wireshark it also shows that the voice
data from remote is missed. Using Mobil phones or ISDN phones calling
the same remote party there is never a bit missed.
This problem occurs rare - once or twice per day and per local voip
phone, but it's quite anoying.
So is there a way to capture the correspondig ISDN voice data FS
receives before it is transmitted via RTP or just droped? I want to c
whether FS drops the early RTP packets or whether FS never got the data
from ISDN.
Sofia Profile is using
param name=inbound-late-negotiation value=true/
The dialplan portion is:
extension name=outgoing-pstn
condition field=destination_number
expression=^94([0-9]+)$ break=never
action application=privacy data=full/
action application=set
data=effective_caller_id_name=anonymous/
action application=set
data=effective_caller_id_number=anonymous/
/condition
condition field=destination_number
expression=^([0-9]+)$
action application=set
data=ignore_early_media=true/
action application=set
data=absolute_codec_string=PCMA/
action application=set
data=continue_on_fail=true/
action application=bridge data=openzap/1/a/$1/
action application=transfer
data=${destination_number} XML et_internal_error/
/condition
Any ideas to refine my debugging?
regard
helmut
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Re: [Freeswitch-users] core dump
That backtrace is not useful because gdb was unable to locate the freeswitch binary. Since you are running it from /opt/freeswitch/bin directory, don't use 'bin/freeswitch' to point gdb to the binary: gdb freeswitch corefile On Mon, Jul 27, 2009 at 1:40 AM, Baskar yudha2...@gmail.com wrote: Hi, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9851 -- Thanks with Regards, N.Baskar ___ 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 -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] core dump
*Hi Rupa, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9854http://pastebin.freeswitch.org/9851 Can some one assist me what is error in freeswitch to hit core dump. -- Thanks with Regards, N.Baskar * ___ 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] core dump
Did you read his response to you? Please generate a usable backtrace as rupa explained and post a bug to jira freeswitch.org On Jul 27, 2009, at 9:00 AM, Baskar yudha2...@gmail.com wrote: Hi Rupa, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9854 Can some one assist me what is error in freeswitch to hit core dump. -- Thanks with Regards, N.Baskar ___ 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] core dump
Again, I like I have to say weekly, Please do not report bugs on the mailing list. http://jira.freeswitch.org Also, please completely re-checkout and rebuild latest trunk and erase your prior freeswitch install before filing the jira. We only accept bug reports of this nature confirmed on a fresh build of latest SVN. On Mon, Jul 27, 2009 at 8:17 AM, Michael Jerris m...@jerris.com wrote: Did you read his response to you? Please generate a usable backtrace as rupa explained and post a bug to jira freeswitch.org On Jul 27, 2009, at 9:00 AM, Baskar yudha2...@gmail.com wrote: *Hi Rupa, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9851 http://pastebin.freeswitch.org/9854 Can some one assist me what is error in freeswitch to hit core dump. -- Thanks with Regards, N.Baskar * ___ 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://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org 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] Dial plan contexts
Jason, You need to set the context to on the profile and the user_context variable on the user to default. There is no such thing as a user- context param on the profile. There is a user_context variable on the user. /b On Jul 27, 2009, at 2:07 AM, Jason White wrote: As per the default configuration, I have user-context set to public in my internal profile, my user has its context set to default, but calls made from the phone registered to that user ID end up in public context when they reach the dial plan. ___ 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] linksys spa962+spa932 blf, hold and intercept of ringing extensions
Hello. I am trying to configure the linksys spa-932 (at http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932 stated that he works with freeswitch). I said Server-Type option set to RFC3265_4235 added to the unit 1 key 1 string: fnc = blf + sd; sub = 1...@pbx0.test.lan; nme = test. The button blinks orange. if call on 1000 (spa962). This subscription runs spa932 and starts to show the status of the phone 1002. Thanks. signature.asc Description: OpenPGP digital 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] SIP trunk (PBX - FreeSwitch) routing problem.
I'm currently trying to connect FreeSwitch to a PBX (Alcatel-Lucent), thanks to a SIP trunk. SIP trunks are available and working on the PBX thanks to a recent update. My problem is that I can't call phones linked to the PBX. When I try to call 300, I've got this message in freeswitch console : 2009-07-27 17:38:48.514105 [NOTICE] switch_channel.c:602 New Channel sofia/internal/[EMAIL PROTECTED] [93d5a10e-7ac3-11de-b456-e5e56113066d] 2009-07-27 17:38:48.516907 [INFO] mod_dialplan_xml.c:252 Processing jgonzalez jgonzalez-300 in context default 2009-07-27 17:38:48.521084 [NOTICE] switch_channel.c:602 New Channel sofia/external/[EMAIL PROTECTED] [93d69816-7ac3-11de-b456-e5e56113066d] 2009-07-27 17:38:48.636073 [NOTICE] sofia.c:3775 Hangup sofia/external/[EMAIL PROTECTED] [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-07-27 17:38:48.636073 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-07-27 17:38:48.637788 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/internal/[EMAIL PROTECTED]) Ended 2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/[EMAIL PROTECTED] [CS_DESTROY] 2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/external/[EMAIL PROTECTED]) Ended 2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/[EMAIL PROTECTED] [CS_DESTROY] I've defined, in sip_profiles/external, a gateway to the PBX this way : include gateway name=pbxlyon param name=username value=pbxlyon/ param name=realm value=[PBX IP address]/ param name=password value=pbxlyon/ param name=register value=false/ param name=register-transport value=udp/ param name=retry_seconds value=30/ /gateway /include And in the dialplan default.xml : extension name=pbxlyon condition field=destination_number expression=300 action application=set data=hangup_after_bridge=true/ action application=set data=continue_on_fail=true/ action application=bridge data=sofia/gateway/pbxlyon/300/ action application=hangup/ /condition /extension (for the moment, I'm trying only with the number 300 which is a correct number of the phone system). As you can see, I'm far from being an expert of FreeSwitch, SIP or even VoIP in general. I'm learning. I hope you can help me. Regards, Julien Gonzalez. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem.
I have to guess that you put this at the bottom of the default.xml? /b On Jul 27, 2009, at 10:58 AM, julien wrote: And in the dialplan default.xml : extension name=pbxlyon condition field=destination_number expression=300 action application=set data=hangup_after_bridge=true/ action application=set data=continue_on_fail=true/ action application=bridge data=sofia/gateway/pbxlyon/300/ action application=hangup/ /condition /extension ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem.
No totally at the bottom. Before : X-PRE-PROCESS cmd=include data=default/*.xml/ Brian West a écrit : I have to guess that you put this at the bottom of the default.xml? /b On Jul 27, 2009, at 10:58 AM, julien wrote: And in the dialplan default.xml : extension name=pbxlyon condition field=destination_number expression=300 action application=set data=hangup_after_bridge=true/ action application=set data=continue_on_fail=true/ action application=bridge data=sofia/gateway/pbxlyon/300/ action application=hangup/ /condition /extension ___ 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] Operation has no matching challenge
Hello, I am getting some SIP registration problems with FreeSWITCH 1.0.4pre10 and my ITSP (Vitelity). The error in the logs is such: 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out Registration Failed with status Operation has no matching challenge [904]. failure #56 While these errors are happening the gateway state (via sofia status) is FAIL_WAIT. With the ever-increasing back-off wait (60, 90, 120, 150, ..., seconds) the registration never resumes. Now one might suspect that there is something wrong with the configuration/ authorization but this problem is intermittent: a simple sofia profile external restart restores the registration and all is well (state turns to REGED) and of course the initial registration succeeds just fine, too. Quite an annoying problem as you never quite know when your gateway is registered when you pick up the receiver of your phone giving the impression of unreliable service! I suspect this to be the same problem, but with a different error message, that has been reported before[1][2]. Thoughts? Anything I should try? Thanks, - Jesse [1] http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011269.html [2] http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html ___ 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] Operation has no matching challenge
Can you get a sofia loglevel all 9 and a sip trace? /b On Jul 27, 2009, at 11:25 AM, Jesse Peterson wrote: Hello, I am getting some SIP registration problems with FreeSWITCH 1.0.4pre10 and my ITSP (Vitelity). The error in the logs is such: 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out Registration Failed with status Operation has no matching challenge [904]. failure #56 While these errors are happening the gateway state (via sofia status) is FAIL_WAIT. With the ever-increasing back-off wait (60, 90, 120, 150, ..., seconds) the registration never resumes. Now one might suspect that there is something wrong with the configuration/ authorization but this problem is intermittent: a simple sofia profile external restart restores the registration and all is well (state turns to REGED) and of course the initial registration succeeds just fine, too. Quite an annoying problem as you never quite know when your gateway is registered when you pick up the receiver of your phone giving the impression of unreliable service! I suspect this to be the same problem, but with a different error message, that has been reported before[1][2]. Thoughts? Anything I should try? Thanks, - Jesse ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] freeswitch and siremis v0.9.3
Hello, recently released siremis v0.9.3 adds support for communication with freeswitch via event socket. siremis is an open source web management interface targeting the SIP routing engines kamailio (openser) and sip-router.org. freeswitch fits perfectly in the picture since it completes the routing engines with rich media services. The new release includes php code to communicate with freeswitch via tcp/event socket and a panel to send commands/display response. Code is grouped like a library, new features being straightforward to develop. For some commands, the output is pretty formatted - screenshot: http://www.asipto.com/gallery/v/siremis/siremis_20.jpg.html?g2_imageViewsIndex=1 More is planned for the future (e.g., display active calls of a certain user, click to end an active call). Feedback and contributions are welcome, visit: http://siremis.asipto.com Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/ ___ 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] Which method Can I use in IVR
See comments inline... On Sun, Jul 26, 2009 at 10:07 PM, Thangappan.M thangappan...@gmail.comwrote: Dear all, I am learning how to implement a IVR in Freeswitch.In our organization we are using Perl scripting language for doing this.So In freeswitch also I need to use Perl. Tony, Brian, and I all like Perl. :) So far I heard two methods for executing IVR. One is in dial plan using perl application.( In perl I create IVR menu and play the voice files) Another one is using event socket.In dial plan I specified socket application and write a Perl script which is listening that particular port and get the session Id. Yes, you can call a script from the dialplan using syntax like this: action application=perl data=/path/to/myivr.pl/ OR You can call an outbound socket connection like this: action application=socket data=127.0.0.1:8084 async full/ Have I understood correctly?.If it is correct means tell which method can I use?. Other make me understand well. You're on the right track. As to which method to use, that depends on your circumstances. How much does it need to scale? Do you want the IVR brain to reside physically on a different server than the FS server? Think about those things. I have seen downloaded perl IVR menu from freeswitch site.In that they called some internal functions like playandGetDigits,StreamFile,ready ...etc. These functions is been called by using $session variable.Where these functions are defined.? When you call a Perl script from the dialplan the script automatically has access to a variable called $session. Check this for more information: http://wiki.freeswitch.org/wiki/Mod_perl#Programming_with_mod_perl Of course, when using the outbound event socket you will not have this magic $session variable. Your best bet to learn more about the socket interface is to look at the sample scripts in src/libs/esl/perl/. (server.pl, server2.pl, and server3.pl) If you are building an IVR with Perl and the event socket be sure to check out src/libs/esl/perl/ESL/IVR.pm which is a small Perl module with some simple abstractions to make IVR programming a bit more convenient. I recommend that you try and create a simple IVR using each method and get a feel for how each one works. -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] Which method Can I use in IVR
Michael, For scale reasons is the best choice event socket? thanks, On Mon, Jul 27, 2009 at 2:00 PM, Michael Collinsm...@freeswitch.org wrote: See comments inline... On Sun, Jul 26, 2009 at 10:07 PM, Thangappan.M thangappan...@gmail.com wrote: Dear all, I am learning how to implement a IVR in Freeswitch.In our organization we are using Perl scripting language for doing this.So In freeswitch also I need to use Perl. Tony, Brian, and I all like Perl. :) So far I heard two methods for executing IVR. One is in dial plan using perl application.( In perl I create IVR menu and play the voice files) Another one is using event socket.In dial plan I specified socket application and write a Perl script which is listening that particular port and get the session Id. Yes, you can call a script from the dialplan using syntax like this: action application=perl data=/path/to/myivr.pl/ OR You can call an outbound socket connection like this: action application=socket data=127.0.0.1:8084 async full/ Have I understood correctly?.If it is correct means tell which method can I use?. Other make me understand well. You're on the right track. As to which method to use, that depends on your circumstances. How much does it need to scale? Do you want the IVR brain to reside physically on a different server than the FS server? Think about those things. I have seen downloaded perl IVR menu from freeswitch site.In that they called some internal functions like playandGetDigits,StreamFile,ready ...etc. These functions is been called by using $session variable.Where these functions are defined.? When you call a Perl script from the dialplan the script automatically has access to a variable called $session. Check this for more information: http://wiki.freeswitch.org/wiki/Mod_perl#Programming_with_mod_perl Of course, when using the outbound event socket you will not have this magic $session variable. Your best bet to learn more about the socket interface is to look at the sample scripts in src/libs/esl/perl/. (server.pl, server2.pl, and server3.pl) If you are building an IVR with Perl and the event socket be sure to check out src/libs/esl/perl/ESL/IVR.pm which is a small Perl module with some simple abstractions to make IVR programming a bit more convenient. I recommend that you try and create a simple IVR using each method and get a feel for how each one works. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] ClueCon 2009 - Last Chance For Early Bird Special!
ClueCon is next week! We're all gearing up for a great event. Here is some important information for those who've not already paid: Today is the last day to receive the $499 early bird special. After today, the price will go up to $699. If you have registered at the ClueCon website but you have not yet paid then please call 877.742.CLUE immediately! We want to make sure that you get the early bird rate. All registrations after today (Monday July 27) will be $699. Thank you for your support of ClueCon 2009! We are looking forward to seeing everyone in person in Chicago. -Michael Collins http://www.cluecon.com 877.742.CLUE ___ 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] Which method Can I use in IVR
On Mon, Jul 27, 2009 at 10:35 AM, roberto miles.c...@gmail.com wrote: Michael, For scale reasons is the best choice event socket? Yes. You can have the IVR stuff running on a separate server altogether. It also gives you great flexibility in designing a setup where you can have a db backend and/or a backup IVR server. The socket method requires a little more effort up front but it pays off in power and flexibility. -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] Slashdot: How-to-Help-With-a-University-ICT-Strategy
http://ask.slashdot.org/story/09/07/27/1652247/How-to-Help-With-a-University-ICT-Strategy An anonymous reader writes I have been asked to contribute to my university's revised ICT (Information and Communication Technology) strategy and I am curious what fellow Slashdot members consider to be the main advice in this context. What are the major mistakes that organizations like universities make? Given the complexity of the different participants in a university, how does one have a coherent strategy that fulfills the needs of such a wide audience? How does one promote open source in a managerial culture? How does one deal with the curse of the virtual learning environment? http://ask.slashdot.org/comments.pl?sid=1316571cid=28842157 ___ 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] IAX configurations
http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk Goes into some detail with connecting to Asterisk via SIP ___ 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] linksys spa962+spa932 blf, hold and intercept of ringing extensions
Hello, I authored that wiki article. The following key will work: fnc=blf+sd+cp;sub=1...@$proxy You need to make sure that presence is not off in the profile. Also cp in the key will enable you to do the intercept of ringing call to watched extension. For further help please join #freeswitch IRC channel. Regards, Ognjen fnc=blf+sd+cp;sub=4...@$proxy On Mon, Jul 27, 2009 at 4:42 PM, Vladimir Elizarov xengelpubl...@gmail.comwrote: Hello. I am trying to configure the linksys spa-932 (at http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932 stated that he works with freeswitch). I said Server-Type option set to RFC3265_4235 added to the unit 1 key 1 string: fnc = blf + sd; sub = 1...@pbx0.test.lan; nme = test. The button blinks orange. if call on 1000 (spa962). This subscription runs spa932 and starts to show the status of the phone 1002. Thanks. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] Distortion on approx first 200ms of G722 prompts on DECT based CPE
Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point - There may be a short break in the RTP between the separate files being played out by FS that makes up any menu. During this time the DECT handset's AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP). Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels. Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit. Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending 'silence' between prompts ? Would be interesting to validate the above 'guess'. Best Regards Keith ___ 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] Operation has no matching challenge
Just to keep those interested informed this thread is being tracked as: http://jira.freeswitch.org/browse/SFSIP-169 On Jul 27, 2009, at 9:30 AM, Brian West wrote: Can you get a sofia loglevel all 9 and a sip trace? /b On Jul 27, 2009, at 11:25 AM, Jesse Peterson wrote: Hello, I am getting some SIP registration problems with FreeSWITCH 1.0.4pre10 and my ITSP (Vitelity). The error in the logs is such: 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out Registration Failed with status Operation has no matching challenge [904]. failure #56 Thanks, - Jesse ___ 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] Dial plan contexts
From the profile: param name=context value=public/ From the user's entry in the directory: variable name=user_context value=default/ but under rev. 14363 when the phone registered to that user makes a call, the dial plan is searched in public context. I hope this helps to clarify. I tried resetting my configuration using Git to a known good state, but with no change to the above behaviour. I'm going to rebuild with the latest from svn soon. ___ 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] Distortion on approx first 200ms of G722 prompts on DECT based CPE
you can set the global var send_silence_when_idle=true in vars.xml On Mon, Jul 27, 2009 at 4:23 PM, Keith Laaks kei...@voxtelecom.co.zawrote: Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point – There may be a short break in the RTP between the separate files being played out by FS that makes up any menu. During this time the DECT handset’s AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP). Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels. Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit. Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending ‘silence’ between prompts ? Would be interesting to validate the above ‘guess’. Best Regards Keith ___ 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] Dial plan contexts
With apologies to all, it was something that sneaked into my configuration that I'm still tracking down. ___ 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] Multiple gateways: Priority the first bridge with prefix
Yes, I actually just want to not be able to communicate with the other bridges. I have already this extension name = sample-1. Freeswitch gets the first extension the 2nd also trigger it. When the calls finds the match it suits perfectly but I just want that I do not want to view the bridges with CS_DESTROY or hangup_after_false if not found. Nandy Dagondon wrote: ed, i mean you use separate extension names: extension name=prefix-51 condition field=destination_number data=(51\d*)\ action application=bridge data=sofia/sip1/$...@222.333.444.555/ /extension extension name=prefix-63 condition field=destination_number data=(63\d*)\ action application=bridge data=sofia/sip2/$...@111.222.333.333/ /extension btw, you should also use separate gateway names sip1 and sip2. so differentiate them in the bridge application. On Mon, Jul 27, 2009 at 4:16 PM, Jason White ja...@jasonjgw.net wrote: Edmar Cruz darklio...@yahoo.com wrote: Not working just the same both of them are running Do you have them as separate extensions in the dial plan? ___ 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 -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674260p24691020.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] Skypiax, Skype endpoint and trunk, revamped
Thanks for the great work. Just want you know that 20 channels with the same username works well on my server. And echo() works without any problem. An updated version of Round Robin hunt and a minor bug posted on jira. Thanks again. 2009/7/27 Giovanni Maruzzelli gmar...@celliax.org Ciao FreeSWITCHers, please have a look at the much changed wiki page: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and checkout and test the code in svn. Much has happened, various bug fixes and features added. Most notable: - multiple instances of the same Skype username on Linux (eg: running 20 concurrent channels as Bob Skype user) - adding and removing interfaces on the fly (patch sent by Muhammad Shahzad) - easier creation of Skype clients configuration directory - reduced latency - better robustness - running as Windows Service - customized ALSA driver for more devices with less IRQs and context switches - custom kernel tickless and 100HZ (eg. solves high load problems in CentOS and in virtualization) - interactive command line client for prototyping Also, please note that ALSA drivers version 1.0.20 seems to be much more stable in our kind of usage (snd-dummy). Various other enhancements will come, but in the mean time please give feedback on the current svn code (we want to be robust for the 1.0.4 Release :-) ) See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm ! -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 ___ 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] Multiple gateways: Priority the first bridge with prefix
Edmar Cruz darklio...@yahoo.com wrote: Yes, I actually just want to not be able to communicate with the other bridges. I have already this extension name = sample-1. Freeswitch gets the first extension the 2nd also trigger it. When the calls finds the match it suits perfectly but I just want that I do not want to view the bridges with CS_DESTROY or hangup_after_false if not found. The above text is absolutely incoherent and incomprehensible, so I don't understand what you are trying to say. Try setting action application=set data=continue_on_fail=true/ on the first extension and see whether that does what you want. I hope this help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
