September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Possible problem in adding channel variables to the bridge destinations
by Jon Bruel 19 Sep '08

19 Sep '08

I have tested the option of adding channel variables to the bridge string, and it does not work. This dialplan works: <extension name="External calls"> <condition field="destination_number" expression="^(\d{8})$"> <action application="set" data="effective_caller_id_number=45161061"/> <action application="bridge" data="sofia/gateway/45161061/$1"/> </condition> </extension> While this one does not work, it is rejected by the gateway: <extension name="External calls"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bridge" data="[effective_caller_id_number=45161061]sofia/gateway/45161061/$1"/> </condition> </extension> The INVITE Remote-Party-ID header differs. In the firs case it is: "Extension 1000" <sip:45161061@jbconsulting.v2tel.dk>;screen=yes;privacy=off And in the second case it is: "Extension 1000" <sip:1000@jbconsulting.v2tel.dk>;screen=yes;privacy=off Is this an error on my behalf, or do we have a bug?

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fs only in lan
by Rocco Lucente 19 Sep '08

19 Sep '08

Hello, mod_sofia don't start correctly when there isn't a internet connection. We can use fs only in lan (without internet connection)? Is there a particolar configuration for this situation? Regards, -- Rocco Lucente Vi preghiamo di considerare l'ambiente prima di stampare questa e-mail Please consider the environment before printing this e-mail

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SCCP aka Skinny
by Cavalera Claudio Luigi 19 Sep '08

19 Sep '08

Hello, is there a way to interconnect fs to a Cisco Call Manager which is configured to speak SCCP protocol (aka Skinny) and not SIP? I did not found a mod_SCCP in the docs :-) Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket
by Christian Jensen 19 Sep '08

19 Sep '08

I have not worked with FS events, but this seems like a sound premise. Anyone care to enlighten an alternative? -----Original Message----- From: Luke Graybill <killarny(a)gmail.com> Sent: September 18, 2008 2:49 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket 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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Speechtools and pocketsphinx always scoring zero
by Greg Thoen 18 Sep '08

18 Sep '08

Hi, I have the pizza demo running on my test box, but the debug shows that while it seems to recognize the words, the score is always zero so it never progresses. Any ideas? 2008-09-17 13:51:25 [DEBUG] mod_pocketsphinx.c:389 pocketsphinx_asr_get_results() Recognized: LARGE, Score: 0 2008-09-17 13:51:25 [DEBUG] SpeechTools.jm:150 console_log() ----XML: <interpretation grammar="pizza_size" score="0"> <result name="match">LARGE</result> <input>LARGE</input> </interpretation> 2008-09-17 13:51:25 [DEBUG] SpeechTools.jm:150 console_log() ----Heard [LARGE] 2008-09-17 13:51:25 [DEBUG] SpeechTools.jm:150 console_log() ----Hit score 0/40/70 2008-09-17 13:51:25 [DEBUG] SpeechTools.jm:365 console_log() ----We don't understand this -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260 http://www.cgicommunications.com/ http://www.elocallink.com/

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DID partially working
by Jair Santos 18 Sep '08

18 Sep '08

I have DID assigned to extension 1000 in public.xml <extension name="public_did"> <condition field="destination_number" expression="^(3462001)$"/> <action application="transfer" data="1000"/> </condition> </extension> Calling this DID from extension 1001 works but when calling from a regular phone it doesn't work. Does anybody have any idea about what I am missing ? thanks Jair Santos

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Hooking into the Audio Stream
by Robert Clayton 18 Sep '08

18 Sep '08

All, Due to the limited audio abilities implemented thus far in FreeSwitch I am needing to handle the audio streams outside of FreeSwitch. So can I use Lua to pass the audio back and forth to an outside library as streamRecord and FilePlay do not provide the functionality needed? And if so what FreeSwitch hooks would I use? Bob

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mod_openzap & PRI
by Evgeniy Zolotov 18 Sep '08

18 Sep '08

We have a lot of questions about mod_openzap & PRI. 1. If to see directory libs/openzap/src/isdn it is possible to see, there are defined such dialects : Q931, national, EvroISDN, DMS, 5ESS. $ ls *.c 5ESSmes.c DMSStateNT.c mfifo.c Q921.c Q931mes.c 5ESSStateNT.c DMSStateTE.c nationalmes.c Q931api.c Q931StateNT.c 5ESSStateTE.c EuroISDNStateNT.c nationalStateNT.c Q931.c Q931StateTE.c DMSmes.c EuroISDNStateTE.c nationalStateTE.c Q931ie.c Q932mes.c Into the file libs/openzap/src/isdn/include/Q931.h (line ~500) are defined only 4 dialects typedef enum // Dialect enum { Q931_Dialect_Q931 = 0, Q931_Dialect_National = 2, Q931_Dialect_DMS = 4, Q931_Dialect_5ESS = 6, // Coming soon to a PRI stack near you! Q931_Dialect_Count } Q931Dialect_t; #define DIALECT_STRINGS "q931", "", "national", "", "dms", "", "5ess", "" Is it possible to add our own dialect PRI (for another submission of SUBBADDRESS, for example), if we'll add new files *mess.c, *StateNT.c, *StateTE and definition * into the Q931Dialect_t ? ================================================================================ 2. We start zaptel : # ls /dev/zap 1 12 15 18 20 23 26 29 31 34 37 4 42 45 48 50 53 56 59 61 8 ctl 10 13 16 19 21 24 27 3 32 35 38 40 43 46 49 51 54 57 6 62 9 pseudo 11 14 17 2 22 25 28 30 33 36 39 41 44 47 5 52 55 58 60 7 channel timer - zaptel is started and configured normally. Next load FreeSWITCH... We use such configuration file for mod_openzap : <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="1"/> </settings> <pri_spans> <span id="1"> <param name="mode" value="user" /> <param name="dialect" value="dms" /> <param name="dialplan" value="XML"/> <param name="context" value="public" /> </span> <span id="2"> <param name="mode" value="user" /> <param name="dialect" value="q931" /> <param name="dialplan" value="XML"/> <param name="context" value="public" /> </span> </pri_spans> </configuration> Next we start (from FreeSWITCH console): freeswitch(a)opos8.altron.lan> console loglevel 9 freeswitch(a)opos8.altron.lan> load mod_openzap ... freeswitch(a)opos8.altron.lan> oz dump 1 16 API CALL [oz(dump 1 16)] output: span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE freeswitch(a)opos8.altron.lan> oz dump 2 16 API CALL [oz(dump 2 16)] output: span_id: 2 chan_id: 16 physical_span_id: 2 physical_chan_id: 47 type: DQ921 state: UP%! last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE In what manner ( using console or debugging messages ) we can see in run-time mode under what type of a dialect every "span" is configured ? If we'll specify for nonexistent dialects: <pri_spans> <span id="1"> <param name="dialect" value="ABC" /> </span> <span id="2"> <param name="dialect" value="XYZ" /> </span> </pri_spans> - then >load mod_openzap is carried out successfully and >oz dump <i> <j> gives us same result like before. What dialect is used by "span" in this case? How it can be looked?

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At the least
by Robert Clayton 18 Sep '08

18 Sep '08

All, To overcome audio limitations at the least can one use a callback in Lua to pull and push raw audio? And if so could both the push and pull be done simultaneously per caller? Thanks, Bob

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