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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 20 Sep '08

20 Sep '08

Both suggestions work: If I include the effective_caller_id_number in the dialplan before bridging, or if I include {origination_caller_id_number=45161061} or [origination_caller_id_number=45161061] as a part of the first parameter after a originate command. But skipping the "set" command in the dialplan, and trying to use: <action application="bridge" data="[effective_caller_id_number=45161061]sofia/gateway/45161061/$1"/> or: <action application="bridge" data="[origination_caller_id_number=45161061]sofia/gateway/45161061/$1"/ > does not work. Same thing when using curled brackets. I need to be able to pass parameters individually to each of the bridge destinations, not just for the A-number, but also for other stuff to be used for post processing of the call. An alternative would be to emulate the Asterisk chan_local by making some kind of loopback to the dialplan, using IAX or SIP, but this is a inelegant way of solving it. Jon

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Error when running java script
by Suganya Rajarethinam 20 Sep '08

20 Sep '08

Hi, I am new to freeswitch and trying to run a javascript example calltest.js.I have put that code inside the folder /usr/local/freeswitch.I have started the freeswitch and from that console I have tried to run the program like jsrun /usr/local/freeswitch/calltest.js. but i am getting the following error: 2008-09-20 15:49:32 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1001(a)192.168.1.2[d1987c09-ebd2-4590-b072-ce851e4b5794] 2008-09-20 15:49:32 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/0000000000(a)192.168.1.2[c98bc58d-2ef9-4656-8f74-9c024000d18c] 2008-09-20 15:49:32 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSWITCH->1001@public 2008-09-20 15:49:32 [ERR] sofia.c:2845 sofia_handle_sip_i_refer() Cannot Blind Transfer 1 Legged calls 2008-09-20 15:49:32 [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup sofia/internal/ 0000000000(a)192.168.1.2 [CS_EXECUTE] [NORMAL_CLEARING] 2008-09-20 15:49:32 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/1001(a)192.168.1.2 [CS_ROUTING] [NO_USER_RESPONSE] 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 23 (sofia/internal/ 0000000000(a)192.168.1.2) Ended 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/ 0000000000(a)192.168.1.2 [CS_HANGUP] 2008-09-20 15:49:32 [WARNING] mod_spidermonkey.c:2933 session_originate() Cannot Create Outgoing Channel! [sofia/default/1001(a)192.168.1.2] 2008-09-20 15:49:32 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 22 (sofia/internal/1001(a)192.168.1.2) Ended 2008-09-20 15:49:32 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1001(a)192.168.1.2[CS_HANGUP] the code is: new_session = new Session(); new_session.originate(session, "sofia/default/1001(a)192.168.1.2"); new_session = new Session(session, "sofia/default/1001(a)192.168.1.2"); // wait for it ... answer new_session.waitForAnswer(10000); // IF everybody is ready, then bridge our current session & the new_session if (session.ready() && new_session.ready()) { bridge(session, new_session); } // log to the console console_log("***********Voila!\n"); // hangup when done new_session.hangup(); Is there anything i have to add to the program or any other requirements are needed to run this javascript successfully? please assist me to do this.

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billing platform
by xbipin 20 Sep '08

20 Sep '08

has any1 been able to use ASTPP billing with freeswitch and if so then my next question, is it possible to use it on a win2k3 platform and not linux as i have freeswitch running on a win2k3 server since a few days and works fine, i just need a prepaid type of script that basically stops a call once its credit is over. i will add the credit etc to the account manually. im ready to post a bounty if any1 willing to do it. the script just needs to end the call once it computes that remaining time is 0 based on available time which is based on the cost per minute and available credit and avoid calling again which needs to be done as soon as call is placed so the call is never connected if account has no credit. -- View this message in context: http://www.nabble.com/billing-platform-tp19549891p19549891.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 20 Sep '08

20 Sep '08

Yes I can, but somehow I start to repeat myself. An INVITE is sent allright every time, but one of the headers differs: Works: Remote-Party-ID: <sip:45161061@jbconsulting.v2tel.dk>;screen=yes;privacy=off Doesn't work: Remote-Party-ID: "Extension 1000" <sip:1000@jbconsulting.v2tel.dk>;screen=yes;privacy=off I have tried with different variable names in the bridge data: <action application="bridge" data="[effective_caller_id_number=45161061]sofia/gateway/45161061/$1"/> as well as: origination_caller_id_number data="[origination_caller_id_number=45161061]sofia/gateway/45161061/$1"/ > the latter combined with: <param name="caller-id-in-from" value="true"/>. But the only thing which works is setting the effective_caller_id_number in the dialplan before the bridge. That would be fine, unless I want to bridge to several destinations. Every time the call is rejected, the Remote-Party-ID-header hasn't the right value. I have tried to execute another dialplan at bridge, testing with other variables, and the info application doesn't reveal the variable I try to include in square brackets in the bridge data string. So maybe its not passed over to the B-leg. If you need more data, let me know, and I'll package it during the weekend. Jon

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 19 Sep '08

19 Sep '08

No the far side is BroadWorks, and as I said in the first mail, the only difference between an accepted and a rejected INVITE is in the contents of the Remote-Party-ID header. Generally in many European countries, the operators limit the A-number to the actual number of the line dialled out on.

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Asynchronous communication with FreeSWITCH's mod_event_socket
by Luke Graybill 19 Sep '08

19 Sep '08

I've been doing a lot of work recently with FreeSWITCH's mod_event_socket, and I wanted to comment a bit about the syntax used for commands through the socket while using asynchronous mode. I haven't tried the synchronous mode yet, as I always want to be free to be able to execute commands without waiting for other commands to finish. For instance, I need to be able to collect DTMF events while I'm playing a sound file, so that the user can do things like select menu items without listening to the entire menu first. Asterisk's AMI protocol allows you to specify an ActionID along with every command that you send. Asterisk then includes this ActionID with every event that is related to that command, making it cake to coordinate an asynchronous client. However, even in async mode, FreeSWITCH's mod_event_socket doesn't communicate any identifying information for command responses, or for events triggered by a previous command, unless one uses the bgapi command set. This command set is not applicable to every situation, though. It only applies to commands which manipulate the call; if one needs to manipulate the channel, then messages must be used through the sendmsg command set, which doesn't provide any specific identifying information. Now, to complicate things, with all commands (even bgapi) the protocol works something like this: you send a command, and wait for a response from mod_event_socket. This response is assumed to be immediate before anything else the client might receive from mod_event_socket, and in the case of bgapi, this response will contain a job-id to use for comparing job-related events later. For example, for the following command: sendmsg call-command: execute execute-app-name: answer\n\n The response is this: Content-Type: command/reply Reply-Text: +OK That response is generic to nearly every single command sent, and is only really saying "The last transmission was a valid command, and didn't immediately fail". The command may actually fail later, and command specific feedback is generally contained in later events (which have no unique identifying information). My issue here is that this seems to be forcing an asynchronous client to rely upon a synchronous ordering of response directly following command, thus violating the very concepts of an asynchronous protocol in which there should be no assumed order. Not only that, but this method increases the complexity of a client, which must be aware of limitations that wouldn't ordinarily be required by a true asynchronous protocol. An asynchronous client should be unconcerned with listening for a synchronous response to every command. My suggested solution is to apply the job-id concept from bgapi to messages as well, and to go a step further; borrow the Asterisk idea of transmitting an identifier along with each command. Every response and event related to that command should then contain the very same identifier in the header.

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Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
by imthiyaz＠peopletech.co.in 19 Sep '08

19 Sep '08

We don't see much samples for PRI outbound dialing . India the line is Euro ISDN. Has anyone tested sangoma A101 cards with Openzap ? I am tring to build a front end web application , to dial using JS in FS which will dial a outbound no and bridge the call to the extension. Thank you Imthiyaz Original Message: ----------------- From: Martin Joseph astmac(a)stillnewt.org Date: Fri, 19 Sep 2008 09:20:45 -0700 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch On Sep 19, 2008, at 6:23 AM, Gopal krishnan wrote: > Hi, > > Basically I just want to test outbound alone with freeswitch, so I > can use extensions.conf in the conf directory rite? > -- I would forget about the asterisk dialplan then. It's very simple to configure an outbound SIP provider in the XML config for FS. Look here for setting up your outbound provider: http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing Look here for a simple example (look for dialplan): http://wiki.freeswitch.org/wiki/Home_PBX_Example I set up a simple outbound SIP tester from these two pages in very little time. Good luck, hope this helps, Marty _______________________________________________ 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 -------------------------------------------------------------------- mail2web.com - Microsoft® Exchange solutions from a leading provider - http://link.mail2web.com/Business/Exchange

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Asiterisk Dialplan for Freeswitch
by Gopal krishnan 19 Sep '08

19 Sep '08

Hi, Is there any way that Asterisk dialplan can be used for freeswitch, since there is a flle extensions.conf in PREFIX/conf directory, is th possible with this file I can write a normal asterisk dialplan so that it will hit the freeswitch. If possible how can it be done any examples or any wiki is avaialble. -- Thank you with regards, Gopal,

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 19 Sep '08

19 Sep '08

None of the suggestions worked, and I still can't control the A-number individually when bridging to multiple destinations. I have tried to change the dialplan using [origination_caller_id_number=1234] As a part of the string in the bridge data, but an info after bridge did not show the variable. I also tried with an unknown variable such as new_var=1234, but it did not show up after the bridge using the info application. Do we have a bug, or has the format been changed..

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 19 Sep '08

19 Sep '08

Mike, you answered: effective_caller_id_number is meant to be set on the a leg, not the b-leg. Well I need to control the "effective_caller_id_number" (or whatever it is called in the B-leg) individually for each location when I bridge to many destinations. In my case, I need to have different identities (A-numbers) for local calls and for external calls. I have followed the WIKI: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#Calling_ multiple_destinations, though. Anyhow how is this solved? Could it be possible to loopback to the dialplan as in Asterisk local channel or to go through some steps before dial attemps are made, but after the bridge application? Jon ------------------------------ Message: 4 Date: Fri, 19 Sep 2008 09:22:51 -0400 From: Michael Jerris <mike(a)jerris.com> Subject: Re: [Freeswitch-users] dialpaln To: freeswitch-users(a)lists.freeswitch.org Message-ID: <B017FCD2-1BF2-4EA3-83AA-786BC26BE141(a)jerris.com> Content-Type: text/plain; charset="us-ascii" On Sep 19, 2008, at 9:18 AM, Gopal krishnan wrote: > Hi, > > Since I am not able to make the outbound call, when I use this > command <oz dump 1 a> in the console, I used to get the all the 31 > channels , for a refrence I am posing one channel block here, > > span_id: 1 > chan_id: 31 > physical_span_id: 1 > physical_chan_id: 31 > type: B > state: DOWN > last_state: DOWN > cid_date: > cid_name: > cid_num: > ani: > aniII: > dnis: > rdnis: > cause: NONE > > In this I am seeing that the state seems to be DOWN, but when I try > wantouer status, it show as connected like , > in openzap, state down is like on-hook... we should change that name maybe. > Wanrouter Status: > > Device name | Protocol | Station | Status | > wanpipe1 | AFT HDLC | N/A | Connected | > wanpipe2 | AFT HDLC | N/A | Connecting | > > where I am using is Sangoma A102D card with hardware echo > cancellation. And when I try to dial from the softphone in the > console I get this as, > > freeswitch(a)dialer.optimus.co.in> 2008-09-19 18:38:58 [NOTICE] > switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/1002(a)172.20.176.31 > [a75028c4-554a-473e-a8cc-8c52d7f72df4] > 2008-09-19 18:38:58 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() > Processing FreeSwitch->99841799874 in context default > 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538 > switch_channel_set_name() New Channel OpenZAP/1:1/9841799874 > [bd977e0d-e078-4193-a005-d751bdb26db8] > 2008-09-19 18:39:28 [NOTICE] sofia.c:2705 sofia_handle_sip_i_state() > Hangup sofia/internal/1002(a)172.20.176.31 [CS_EXECUTE] > [ORIGINATOR_CANCEL] > 2008-09-19 18:39:28 [NOTICE] switch_ivr_originate.c:1321 > switch_ivr_originate() Hangup OpenZAP/1:1/9841799874 > [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] > 2008-09-19 18:39:28 [INFO] mod_dptools.c:1814 > audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL > 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 > switch_core_session_thread() Session 1 (sofia/internal/1002(a)172.20.176.31 > ) Ended > 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 > switch_core_session_thread() Close Channel sofia/internal/1002(a)172.20.176.31 > [CS_HANGUP] > 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 > switch_core_session_thread() Session 2 (OpenZAP/1:1/9841799874) Ended > 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 > switch_core_session_thread() Close Channel OpenZAP/1:1/9841799874 > [CS_HANGUP] > > So where i am wrong, can you please correct me. > It looks like the hang-up is coming from the sip side of things, can you take a look at the sip trace and confirm there is really a cancel on sip. Mike

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