[Freeswitch-users] Passwords in clear text
I've seen in the XCML files that passwords and credentials e.g. for directory entries are always stored in clear text. Is there a way to use encrypted passwords? Bet regaerds Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Passwords in clear text
On Thursday 16 October 2008 20:40:29 Peter P GMX wrote: I've seen in the XCML files that passwords and credentials e.g. for directory entries are always stored in clear text. Is there a way to use encrypted passwords? As mentioned on this page; http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide you can use a1-hash values instead. hads ___ Freeswitch-users mailing list Freeswitch-users@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] Passwords in clear text
Hi, Yes, you can just return an a1-hash instead of a password. The a1-hash consists of md5(username:domain:password) see: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide regards, Leon On Oct 16, 2008, at 9:40 AM, Peter P GMX wrote: I've seen in the XCML files that passwords and credentials e.g. for directory entries are always stored in clear text. Is there a way to use encrypted passwords? Bet regaerds 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 mailing list Freeswitch-users@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] busy tone detection
Hi, I am using event socket to originate calls. I need to originate the calls thru console and need to detect the tone. In Asterisk we used to detect thru BackgroundDetect and VMDetect. In freeswitch I found that the tones.conf which will detect the tones that we are dialing. I am not sure how to integrate with the event socket to detect the tone with console dialing. Any help will be appreciated. My tones.conf [in] generate-dial = v=-7;%(1000,0,375,425) detect-dial = 375,425 generate-ring = v=-7;%(2000,4000,440,480) detect-ring = 440,480 generate-busy = v=-7;%(500,500,480,620) detect-busy = 480,620 generate-attn = v=0;%(100,100,1400,2060,2450,2600) detect-attn = 1400,2060,2450,2600 generate-callwaiting-sas = v=0;%(300,0,440) detect-callwaiting-sas = 440 generate-callwaiting-cas = v=0;%(80,0,2750,2130) detect-callwaiting-cas = 2750,2130 detect-fail1 = 913.8 detect-fail2 = 1370.6 detect-fail3 = 776.7 thanks -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@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] capture DTMF
there is a read application that will use a file to prompt you with an audio file, ask for a certain number of digits and place then into a channel variable. It's probably on the wiki On Thu, Oct 16, 2008 at 12:57 AM, Noah Silverman [EMAIL PROTECTED]wrote: Hi, I'm looking for a way to capture DTMF in the dialplan. My goal is to have a way for a user to enter an account number, pin code, etc. I've seen some examples in javascript and/or perl, but would really like to avoid using an outside language. Does FS have any kind of function to capture the DTMF. (The format for the javascript function looks great if there was something similar in FS. It allows you to set the number of digits, timeout, etc..) Thanks, -N ___ Freeswitch-users mailing list Freeswitch-users@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:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] 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] busy tone detection
Hi,
Thanks for the mail. I tried in this format to detect the busy signal but
I cant.
I am using javascript file like,
session1 = new Session();
session1.originate(session1,{ignore_early_media=true}sofia/internal/+argv[0]+,30);
session1.execute(bridge, sofia/default/+argv[1]+@172.20.176.254);
session1.execute(transfer, argv[1]);
function on_event(new_session, type, event_obj, user_data)
{
if (type == event)
{
console_log(notice, Event: + event_obj.serialize() +
\n);
}
}
session1.execute(tone_detect,busy 480,620 r);
session1.streamFile(/usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav,
on_event);
//session1.execute(stop_tone_detect);
but i cant able to detect the tone.
--
Thank you with regards,
Gopal,
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Re: [Freeswitch-users] busy tone detection
gotta start tone_detect right after originate and before bridge.
On Thu, Oct 16, 2008 at 7:53 AM, Gopal krishnan [EMAIL PROTECTED] wrote:
Hi,
Thanks for the mail. I tried in this format to detect the busy signal but
I cant.
I am using javascript file like,
session1 = new Session();
session1.originate(session1,{ignore_early_media=true}sofia/internal/+argv[0]+,30);
session1.execute(bridge, sofia/default/+argv[1]+@172.20.176.254);
session1.execute(transfer, argv[1]);
function on_event(new_session, type, event_obj, user_data)
{
if (type == event)
{
console_log(notice, Event: + event_obj.serialize()
+ \n);
}
}
session1.execute(tone_detect,busy 480,620 r);
session1.streamFile(/usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav,
on_event);
//session1.execute(stop_tone_detect);
but i cant able to detect the tone.
--
Thank you with regards,
Gopal,
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED] [EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED]
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED][EMAIL PROTECTED]
pstn:213-799-1400
___
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[Freeswitch-users] Help on call transfer
Hi, I am unable to do call transfer when media is in bypass mode. Please let me know how it can be done. Also why is it implemented in such a way that call transfer works only when media flows through freeswitch. Appreciate your help Regards Sridhar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] How to get DISA working ?
-- View this message in context: http://www.nabble.com/How-to-get-DISA-working---tp20014047p20014047.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] How to get DISA working ?
I'm trying to get DISA working. I've done this: exten name= condition field=destination_number expression=^$ action application=javascript data=/usr/local/freeswitch/conf/disa/disa.js/ /extension The file disa.js is here: http://wiki.freeswitch.org/wiki/Examples_disa.js calling gives me busy tone: [EMAIL PROTECTED] 2008-10-16 15:54:29 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [f3d6e460-9b89-11dd-910f-55dc10d13151] 2008-10-16 15:54:29 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing 2000- in context default 2008-10-16 15:54:29 [NOTICE] switch_ivr.c:1116 switch_ivr_session_transfer() Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED] 2008-10-16 15:54:31 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-16 15:54:31 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/[EMAIL PROTECTED] [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-16 15:54:31 [NOTICE] switch_core_session.c:833 switch_core_session_thread() Session 12 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-10-16 15:54:31 [NOTICE] switch_core_session.c:835 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] And API call gives: [EMAIL PROTECTED] jsrun /usr/local/freeswitch/conf/disa/disa.js 2008-10-16 15:56:42 [ERR] disa.js:27 mod_spidermonkey() TypeError: session.ready is not a function API CALL [jsrun(/usr/local/freeswitch/conf/disa/disa.js)] output: OK I'm I doing something wrong ? Henk -- View this message in context: http://www.nabble.com/How-to-get-DISA-working---tp20014227p20014227.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] Help on call transfer
If you want to do transfer then fs should stay in media path. Thanks, Ruchir Brahmbhatt Director Ecosmob Technologies Pvt. Ltd. -Original Message- From: Rajagopal, Sridhar (Sridhar) [EMAIL PROTECTED] Reply-to: freeswitch-users@lists.freeswitch.org To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Help on call transfer Date: Thu, 16 Oct 2008 15:49:05 +0530 Hi, I am unable to do call transfer when media is in bypass mode. Please let me know how it can be done. Also why is it implemented in such a way that call transfer works only when media flows through freeswitch. Appreciate your help Regards Sridhar ___ Freeswitch-users mailing list Freeswitch-users@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] Help on call transfer
We need to be in media path to do the ringback in an attended transfer. There are some new params in trunk that will make it pop back out of the media path on the completion of transfer. Mike On Oct 16, 2008, at 11:41 AM, Ruchir Brahmbhatt wrote: If you want to do transfer then fs should stay in media path. Thanks, Ruchir Brahmbhatt Director Ecosmob Technologies Pvt. Ltd. -Original Message- From: Rajagopal, Sridhar (Sridhar) [EMAIL PROTECTED] Reply-to: freeswitch-users@lists.freeswitch.org To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Help on call transfer Date: Thu, 16 Oct 2008 15:49:05 +0530 Hi, I am unable to do call transfer when media is in bypass mode. Please let me know how it can be done. Also why is it implemented in such a way that call transfer works only when media flows through freeswitch. Appreciate your help Regards Sridhar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to get DISA working ?
2008-10-16 15:54:29 [NOTICE] switch_ivr.c:1116 switch_ivr_session_transfer() Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED] 2008-10-16 15:54:31 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting Your routing to enum for extension and there is no enum route for that number. Mike On Oct 16, 2008, at 9:59 AM, henkoegema wrote: I'm trying to get DISA working. I've done this: exten name= condition field=destination_number expression=^$ action application=javascript data=/usr/local/freeswitch/conf/disa/disa.js/ /extension The file disa.js is here: http://wiki.freeswitch.org/wiki/Examples_disa.js calling gives me busy tone: [EMAIL PROTECTED] 2008-10-16 15:54:29 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [f3d6e460-9b89-11dd-910f-55dc10d13151] 2008-10-16 15:54:29 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing 2000- in context default 2008-10-16 15:54:29 [NOTICE] switch_ivr.c:1116 switch_ivr_session_transfer() Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED] 2008-10-16 15:54:31 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-16 15:54:31 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/[EMAIL PROTECTED] [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-16 15:54:31 [NOTICE] switch_core_session.c:833 switch_core_session_thread() Session 12 (sofia/internal/[EMAIL PROTECTED] ) Ended 2008-10-16 15:54:31 [NOTICE] switch_core_session.c:835 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] And API call gives: [EMAIL PROTECTED] jsrun /usr/local/freeswitch/conf/disa/disa.js 2008-10-16 15:56:42 [ERR] disa.js:27 mod_spidermonkey() TypeError: session.ready is not a function API CALL [jsrun(/usr/local/freeswitch/conf/disa/disa.js)] output: OK I'm I doing something wrong ? ___ Freeswitch-users mailing list Freeswitch-users@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] failover on landline vs cellular
I'm having a problem with failover. I have the call_timeout set to 15 seconds which results in a landline ringing 4 rings, and INSTANTLY after the 4th ring fails over to the failover destination. However, if the number being dialed is a cell phone I've had one of two things happen: it will either ring and ring until voicemail, or what seems to happening now (I'm not sure what setting caused this) it will rings twice and there will be dead air and eventually the calling side will time out and fail. timeout=15 desk phone: 4 rings - failover sprint mobile: 5 rings - voicemail verizon mobile: 5 rings - voicemail section name=dialplan description= context name=public extension name=did condition field=destination_number expression=^($did)$ !-- set the caller id -- action application=set data=effective_caller_id_name=Bob Smith/ action application=set data=effective_caller_id_number=8005551234/ !-- stops processing the dialplan after call is bridged -- !-- action application=set data=hangup_after_bridge=true/ -- !-- this is needed to allow call_timeout to work after bridging to a gateway -- action application=set data=ignore_early_media=true/ !-- continue to process the dialplan after failure -- action application=set data=continue_on_fail=NO_ANSWER/ action application=set data=continue_on_fail=true/ !-- action application=set data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION,NO_ANSWER/ -- !-- try the client shop -- action application=set data=call_timeout=15/ action application=bridge data=sofia/vitelity/number to try@carrier/ !-- failover destination -- action application=bridge data=sofia/vitelity/$0@failover dest/ /condition /extension /context /section call to desk: 2008-10-15 15:47:13 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/caller@carrier [1022c19a-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:47:13 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing caller-[EMAIL PROTECTED] 2008-10-15 15:47:13 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/4078023342@carrier [1026f4ae-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:47:14 [NOTICE] sofia.c:2167 sofia_handle_sip_i_state() Ring-Ready sofia/vitelity/4078023342@carrier! 2008-10-15 15:47:14 [NOTICE] mod_sofia.c:1058 sofia_receive_message() Ring-Ready sofia/vitelity/caller@carrier! 2008-10-15 15:47:14 [NOTICE] switch_ivr_originate.c:1148 switch_ivr_originate() Ring Ready sofia/vitelity/caller@carrier! 2008-10-15 15:47:28 [NOTICE] switch_ivr_originate.c:1277 switch_ivr_originate() Hangup sofia/vitelity/4078023342@carrier [CS_CONSUME_MEDIA] [NO_ANSWER] 2008-10-15 15:47:28 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NO_ANSWER 2008-10-15 15:47:28 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 181 (sofia/vitelity/4078023342@carrier) Ended 2008-10-15 15:47:28 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/vitelity/4078023342@carrier [CS_HANGUP] 2008-10-15 15:47:28 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/4073921320@failover dest [18d07116-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:47:28 [NOTICE] sofia.c:2456 sofia_handle_sip_i_state() Channel [sofia/vitelity/4073921320@failover dest] has been answered 2008-10-15 15:47:28 [NOTICE] sofia.c:2469 sofia_handle_sip_i_state() Channel [sofia/vitelity/caller@carrier] has been answered 2008-10-15 15:47:45 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/vitelity/caller@carrier [CS_HIBERNATE] [NORMAL_CLEARING] 2008-10-15 15:47:45 [NOTICE] switch_ivr_bridge.c:586 signal_bridge_on_hangup() Hangup sofia/vitelity/4073921320@failover dest [CS_HIBERNATE] [NORMAL_CLEARING] 2008-10-15 15:47:45 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 180 (sofia/vitelity/caller@carrier) Ended 2008-10-15 15:47:45 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/vitelity/caller@carrier [CS_HANGUP] 2008-10-15 15:47:45 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 182 (sofia/vitelity/4073921320@failover dest) Ended 2008-10-15 15:47:45 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/vitelity/4073921320@failover dest [CS_HANGUP] call to cell phone: 2008-10-15 15:51:09 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/caller@carrier [9cce820a-9af2-11dd-bf67-89c26635687a] 2008-10-15 15:51:09 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing caller-[EMAIL PROTECTED] 2008-10-15 15:51:09 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/vitelity/3216956446@carrier
[Freeswitch-users] Getting terminator value from PlayAndGetDigit
Hi, I am looking for ways to extract the actual terminator value being pressed by user in the case where user can press # or * to signal the end of DTMF input. By extracting this terminator value, I would be able to perform action accordingly. For instance, if * is pressed, it means the user is aware that he/she has entered the wrong number and would like to retry. Could someone tell me if it is something possible within Freeswitch? Thanks, 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] Can't see any Sofia messages
2008/10/16 Michael Jerris [EMAIL PROTECTED]: If you are seeing nothing at all on the console with all that set, then the packets are never getting to FreeSWITCH. My first guess would be either firewall or bound to the wrong ip/port. iptables rules wrong. 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] Couple Quick Questions
First incredibly nice work. I've been beating on a freeswitch box in an SBC type config with a hardware call generator and it's holding up extremely well. I just have 2 questions at the moment. 1) Is there any place to get call statitics for reporting/trending puproses, specifically around complettion rates of gateways/endpoints. If it already exists is it switch wide or can I collect them per endpoint? Apart from parsing the cdrs files of course. If not, any pointers as to where it should live in the code ? I haven't dug into the source much yet, which is the only reason I ask. 2) I think this may have already been answered but I couldn't find the previous email I thought I saw. The calls per second limit that exists today is only for a switch wide config and is not configurable, at this point, on a per endpoint basis right ? Thanks for you time -e ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to get DISA working ?
In other words, you don't have your extension in the right spot. Check to see that it isn't in the default.xml dialplan file after the enum extension. It needs to be *before* the enum extension because the enum extension is the catch-all - it grabs everything that hasn't already been matched in the dialplan. HINT: You might want to consider putting your extension in the conf/dialplan/default directory in its own XML file. Extensions in that directory are always inserted into the dialplan prior to the enum extension. If you want your specific XML file to be parsed prior to the other files in there then be sure to name it with a low number, like 001_My_Extensions.xml. Note that there are a few files in there already. The file you are particularly interested in is 9_enum.xml. As long as your number is lower than 9 then you'll get your file parsed first. NOTE: A filename beginning with an alpha character is not smaller than 9! Definitely name your files with at least one digit first in the filename. (Ask me how I learned that one!;) HtH -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Jerris Sent: Thursday, October 16, 2008 8:57 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to get DISA working ? 2008-10-16 15:54:29 [NOTICE] switch_ivr.c:1116 switch_ivr_session_transfer() Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED] 2008-10-16 15:54:31 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting Your routing to enum for extension and there is no enum route for that number. Mike On Oct 16, 2008, at 9:59 AM, henkoegema wrote: I'm trying to get DISA working. I've done this: exten name= condition field=destination_number expression=^$ action application=javascript data=/usr/local/freeswitch/conf/disa/disa.js/ /extension The file disa.js is here: http://wiki.freeswitch.org/wiki/Examples_disa.js calling gives me busy tone: [EMAIL PROTECTED] 2008-10-16 15:54:29 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [f3d6e460-9b89-11dd-910f-55dc10d13151] 2008-10-16 15:54:29 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing 2000- in context default 2008-10-16 15:54:29 [NOTICE] switch_ivr.c:1116 switch_ivr_session_transfer() Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED] 2008-10-16 15:54:31 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-16 15:54:31 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/[EMAIL PROTECTED] [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-16 15:54:31 [NOTICE] switch_core_session.c:833 switch_core_session_thread() Session 12 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-10-16 15:54:31 [NOTICE] switch_core_session.c:835 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] And API call gives: [EMAIL PROTECTED] jsrun /usr/local/freeswitch/conf/disa/disa.js 2008-10-16 15:56:42 [ERR] disa.js:27 mod_spidermonkey() TypeError: session.ready is not a function API CALL [jsrun(/usr/local/freeswitch/conf/disa/disa.js)] output: OK I'm I doing something wrong ? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to get DISA working ?
This hint only works in the default configs in SVN trunk as of the past two weeks if I recall. Also in your version you'll be conf/ diaplan/extensions/ /b On Oct 16, 2008, at 2:30 PM, Michael Collins wrote: HINT: You might want to consider putting your extension in the conf/ dialplan/default directory in its own XML file. Extensions in that directory are always inserted into the dialplan prior to the enum extension. If you want your specific XML file to be parsed prior to the other files in there then be sure to name it with a low number, like 001_My_Extensions.xml. Note that there are a few files in there already. The file you are particularly interested in is 9_enum.xml. As long as your number is lower than “9” then you’ll get your file parsed first. NOTE: A filename beginning with an alpha character is not “smaller” than 9! Definitely name your files with at least one digit first in the filename. (Ask me how I learned that one!;) ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to get DISA working ?
Oops, my directory layout is older. I will get a proper update put into place right now. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Thursday, October 16, 2008 2:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to get DISA working ? This hint only works in the default configs in SVN trunk as of the past two weeks if I recall. Also in your version you'll be conf/diaplan/extensions/ /b On Oct 16, 2008, at 2:30 PM, Michael Collins wrote: HINT: You might want to consider putting your extension in the conf/dialplan/default directory in its own XML file. Extensions in that directory are always inserted into the dialplan prior to the enum extension. If you want your specific XML file to be parsed prior to the other files in there then be sure to name it with a low number, like 001_My_Extensions.xml. Note that there are a few files in there already. The file you are particularly interested in is 9_enum.xml. As long as your number is lower than 9 then you'll get your file parsed first. NOTE: A filename beginning with an alpha character is not smaller than 9! Definitely name your files with at least one digit first in the filename. (Ask me how I learned that one!;) ___ Freeswitch-users mailing list Freeswitch-users@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 on call transfer
Very interesting. Could you point out those params and when they apply? Also, does FS differentiate between an attended and a blind transfer? Just curious if the type of transfer matters or not. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Jerris Sent: Thursday, October 16, 2008 8:55 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Help on call transfer We need to be in media path to do the ringback in an attended transfer. There are some new params in trunk that will make it pop back out of the media path on the completion of transfer. Mike On Oct 16, 2008, at 11:41 AM, Ruchir Brahmbhatt wrote: If you want to do transfer then fs should stay in media path. Thanks, Ruchir Brahmbhatt Director Ecosmob Technologies Pvt. Ltd. -Original Message- From: Rajagopal, Sridhar (Sridhar) [EMAIL PROTECTED] mailto:%22Rajagopal,[EMAIL PROTECTED] -lucent.com%3e Reply-to: freeswitch-users@lists.freeswitch.org To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Help on call transfer Date: Thu, 16 Oct 2008 15:49:05 +0530 Hi, I am unable to do call transfer when media is in bypass mode. Please let me know how it can be done. Also why is it implemented in such a way that call transfer works only when media flows through freeswitch. Appreciate your help Regards Sridhar ___ Freeswitch-users mailing list Freeswitch-users@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] Auto Changing port
Hi I'm using openser+FS, FS as B2BUA, so, the call goes to openser and then to FS. The problem is, when customer is behind nat, some times work and some times not. When freeswitch show this message, work: 2008-10-17 09:50:25 [INFO] switch_rtp.c:1289 rtp_common_read() Auto Changing port from x.x.x.x:1122 to x.x.x.x:1244 So, I can make 10 calls from the same voip, in sequence, and sometimes work and sometimes not, all time that work I see this message. There are any way to force FS to do that all time? Thanks Thiago ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Problem with PlayAndGetDigits
Hi, I am finding some problems when using PlayAndGetDigits within lua. As seen in the log below, I set the retry count to '3' and valid digit to be 1,2,3, or 5. I pressed 9 for every retry attempt, and found that Freeswitch actually let me retry for more than 3 times. Moreover, Freeswitch reports DTMF of 99 is received even though I only pressed 9. At the end, it reports 99 as the final result. Does anyone know if this is a config problem or a bug? Thanks, Keith 2008-10-17 20:07:20 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [test9] destination_number(rule) =~ /(.*)/ 2008-10-17 20:07:20 [NOTICE] switch_core_session.c:1219 switch_core_session_execute_exten() Execute answer() start ivr 2008-10-17 20:07:20 [NOTICE] switch_core_session.c:1219 switch_core_session_execute_exten() Execute lua(run_ivr.lua 1 1 3 3000 # /audio/admin_menu.wav /audio/invalid_input.wav 1|2|3|5 admin_selection ) 2008-10-17 20:07:20 [DEBUG] switch_ivr_play_say.c:1455 switch_play_and_get_digits() switch_play_and_get_digits(session, 1, 1, 3, 3000, #*, /audio/admin_menu.wav, /audio/invalid_input.wav, digit_buffer, 512, 1|2|3|5) 2008-10-17 20:07:20 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms 2008-10-17 20:07:20 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-17 20:07:21 [DEBUG] switch_ivr_play_say.c:1218 switch_ivr_play_file() done playing file 2008-10-17 20:07:21 [DEBUG] switch_ivr_play_say.c:1473 switch_play_and_get_digits() play gave up 9 2008-10-17 20:07:21 [DEBUG] switch_ivr_play_say.c:1483 switch_play_and_get_digits() Checking regex [1|2|3|5] on [9] 2008-10-17 20:07:21 [DEBUG] switch_regex.c:198 switch_regex_match() number of matches: -1 2008-10-17 20:07:21 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms 2008-10-17 20:07:21 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-17 20:07:25 [DEBUG] switch_ivr_play_say.c:1218 switch_ivr_play_file() done playing file 2008-10-17 20:07:25 [DEBUG] switch_ivr_play_say.c:1507 switch_play_and_get_digits() Calling more digits try 3 2008-10-17 20:07:25 [DEBUG] switch_ivr_play_say.c:1524 switch_play_and_get_digits() Checking regex [1|2|3|5] on [99] 2008-10-17 20:07:25 [DEBUG] switch_regex.c:198 switch_regex_match() number of matches: -1 2008-10-17 20:07:25 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms 2008-10-17 20:07:25 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-17 20:07:30 [DEBUG] switch_ivr_play_say.c:1218 switch_ivr_play_file() done playing file 2008-10-17 20:07:30 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms 2008-10-17 20:07:30 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-17 20:07:30 [DEBUG] switch_ivr_play_say.c:1218 switch_ivr_play_file() done playing file 2008-10-17 20:07:30 [DEBUG] switch_ivr_play_say.c:1473 switch_play_and_get_digits() play gave up 9 2008-10-17 20:07:30 [DEBUG] switch_ivr_play_say.c:1483 switch_play_and_get_digits() Checking regex [1|2|3|5] on [9] 2008-10-17 20:07:30 [DEBUG] switch_regex.c:198 switch_regex_match() number of matches: -1 2008-10-17 20:07:30 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms 2008-10-17 20:07:30 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-17 20:07:34 [DEBUG] switch_ivr_play_say.c:1218 switch_ivr_play_file() done playing file 2008-10-17 20:07:34 [DEBUG] switch_ivr_play_say.c:1507 switch_play_and_get_digits() Calling more digits try 2 2008-10-17 20:07:35 [DEBUG] switch_ivr_play_say.c:1524 switch_play_and_get_digits() Checking regex [1|2|3|5] on [99] 2008-10-17 20:07:35 [DEBUG] switch_regex.c:198 switch_regex_match() number of matches: -1 2008-10-17 20:07:35 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms 2008-10-17 20:07:35 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-17 20:07:38 [DEBUG] sofia_reg.c:121 sofia_reg_check_gateway() registered nine 2008-10-17 20:07:39 [DEBUG] switch_ivr_play_say.c:1218 switch_ivr_play_file() done playing file 2008-10-17 20:07:39 [DEBUG] switch_ivr_play_say.c:928 switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms 2008-10-17 20:07:39 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 2008-10-17 20:07:39 [DEBUG] switch_ivr_play_say.c:1218 switch_ivr_play_file() done playing file 2008-10-17 20:07:39 [DEBUG] switch_ivr_play_say.c:1473
Re: [Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error
Didn't get that On Thu, Oct 16, 2008 at 6:13 PM, Anthony Minessale [EMAIL PROTECTED] wrote: You could always set up an acl so FS won't send auth to that particular host. On Thu, Oct 16, 2008 at 12:58 AM, Gayatri Kulkarni [EMAIL PROTECTED] wrote: UA1 discovers P1 as a result of configuration, DHCP assignment or other similar operation, also outside the scope of this document. REGISTRAR has a similar *default outbound proxy* relationship with P3. Check if you have this configured on the Avaya SES as your FS. To do that, login to admin interface of Avaya SES and in the setup side bar, there's a link Called 'Hosts' click on the link for this home and check what outbound proxy has been specified Is your SES box a Home or Edge or Home/Edge combo?The Hosts link appears only on Edge or home/Edge combo Thanks Thomas! -- Regards, Gayatri Kulkarni On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull [EMAIL PROTECTED] wrote: Gayatri, Any idea on how to enable this response in Freeswitch? David, Not sure of the lr... On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni [EMAIL PROTECTED] wrote: Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems! On Wed, Oct 15, 2008 at 1:52 PM, David Knell [EMAIL PROTECTED] wrote: On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote: Record-Route: sip:10.0.2.154:5060;lr Record-Route: sip:10.0.2.151:5061;lr;transport=tls what's the 'lr' next to the port number? short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html --Dave ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards, Gayatri Kulkarni ___ Freeswitch-users mailing list Freeswitch-users@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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] 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 -- Regards, Gayatri Kulkarni ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
