April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL
by Guido Kuth 29 Apr '09

29 Apr '09

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 Message processed 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.minessale(a)gmail.com> To: gk(a)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 <gk(a)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/1001@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(a)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_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400

1 0

Serious Problem detecting DTMF in bridged SIP call using ESL
by Guido Kuth 29 Apr '09

29 Apr '09

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/1001@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

3 2

Re: [Freeswitch-users] Very confusing startup error
by Guido Kuth 29 Apr '09

29 Apr '09

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 <gerry(a)pstn2.net> To: gk(a)exram.de All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitch>freeswitch 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?

3 2

HELP 3-way network access
by Edward Q. 29 Apr '09

29 Apr '09

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 Apache 2.2.3 MySQL 5.0.45 SSH OpenSSH 4.3 freeswitch@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 .. freeswitch@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

2 1

Very confusing startup error
by Gerry Hull 29 Apr '09

29 Apr '09

All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitch>freeswitch 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?

2 1

Condition and custom variables
by Alex Gusak 29 Apr '09

29 Apr '09

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_extension}@${domain_name}"/> <action application="hangup"/> </condition> </extension> -- Alex Gusak

3 2

SNMP or Cacti to graph perrformance?
by Ron McCarthy 29 Apr '09

29 Apr '09

Hi List, I have searched and searched and have found nothing. Id like to be able to keep track of current calls and the CPS and then chart these via Cacti. Is their some a SNMP plugin that I have not seen for FS that can do some of these things? Also, if I have to make some custom plugin that montiors output the of show channels /show calls via one of the API's, anyway we can get a realtime stream so I don't have to query it every second to get correct data? Thanks!

3 5

ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available
by Michael Collins 28 Apr '09

28 Apr '09

The latest pre-release of 1.0.4 is available. More information here: http://www.freeswitch.org/node/177 The FreeSWITCH development team would like to thank everyone for their help in reporting bugs and testing features. Please keep the feedback coming and spread the word about what a cool project FreeSWITCH really is! -Michael See you at ClueCon - http://www.cluecon.com

3 2

Voice mail: skip announcement
by paul.degt 28 Apr '09

28 Apr '09

Is there a way to skip "the person at extension..." announcement when forwarding a call to voice mail, so that it starts recording immediately, or just says something like "start recording..."? Also, is it possible to turn off voice mail play back feature, meaning when somebody tries to leave a voice mail one needs to press a button to end message and then VM suddenly plays it back? I see there's sounds.xml but not sure it help me to do what I need , is there any write up on how to customize it? Would appreciate some directions.

2 3

Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available
by Brian West 28 Apr '09

28 Apr '09

Fixed. /b On Apr 28, 2009, at 3:38 PM, Alex Rambau wrote: > The link in the post refers to pre5, not pre6. Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

1 0