April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Cannot compile freetdm!
by Valery Kalinin 08 Apr '11

08 Apr '11

# cd /usr/local/freeswitch/libs/freetdm # ./configure --with-libisdn # make bla-bla-bla cc1: warnings being treated as errors src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34': src/ftmod/ftmod_isdn/ftmod_isdn.c:982: warning: unused variable 'cplen' src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'isdn_configure_span': src/ftmod/ftmod_isdn/ftmod_isdn.c:2794: warning: passing argument 2 of 'Q931SetLogCB' from incompatible pointer type make: *** [ftmod_isdn_la-ftmod_isdn.lo] Error 1 Why? libisdn-0.0.1 installed

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Re: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5 with latest GIT head
by JRichey 08 Apr '11

08 Apr '11

What do you see if you run "chkconfig --list | grep freeswitch"? # chkconfig --list | grep freeswitch freeswitch 0:off 1:off 2:on 3:on 4:on 5:on 6:off If you don't get anything you'll need to add it with "chkconfig --add freeswitch" and "chkconfig freeswitch on". -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of fieldpeak Sent: Friday, April 08, 2011 9:21 AM To: FreeSWITCH-users Cc: 13910936628(a)139.com Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5 with latest GIT head I'm a newbie, and trying use scritp for FS to auto startup when OS start, i follow below link, http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix <http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix> http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora <http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora> However, after the OS started for some time, the FS still not startup (by register from eyebeam failure and use './freeswitch status' to check), if i manually excute the scritp (./freeswitch start), it works well. Could anyone advise any clue to to resolve it? Attached is the startup script, FS running on CentOS 5.5 with latest GIT head. Thanks Regards, Charles

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New FreeSWITCH IVR coming, but need HELP!
by Zac Wolfe 08 Apr '11

08 Apr '11

Hi guys, First some good news: we're finally close to releasing our free IVR Development platform SafiServer/SafiWorkshop (www.safisystems.com) with FreeSWITCH support! It's happening much later than we originally anticipated as we've been unexpectedly busy with contracting opportunities but I think it will be worth the wait. Currently everything is working fine with one minor exception: if the user-created script (we call them Saflets) doesn't explicitly hang up the call, the call will remain parked until the caller hangs up. Some details: In Asterisk we invoke our server-side scripting applications via the extensions.conf using the following syntax: exten = 1111,1,Agi(agi:// 192.168.0.10:3573/safletEngine.agi?saflet=project1/callflow1) Here '192.168.0.10' is the IP address of the SafiServer and project1/callflow1 identifies the Saflet to be executed. Asterisk creates a socket connection to the SafiServer and, once the socket is disconnected, the call proceeds to the next entry in the dialplan (typically 'hangup'). For FreeSWITCH, the process is slightly different. Currently, rather than create a separate socket connection for each incoming call, we're invoking an event that informs the SafiServer that there is a new incoming call. The event contains the contextual information including the Saflet name. For example: <extension name="9194"> <condition field="destination_number" expression="9194$"> <action inline="true" application="set" data="saficall=true"/> <action application="event" data="Event-Subclass=saficall::incoming,Event-Name=CUSTOM,saflet_project=test,saflet=flow1,new_saficall=true"/>  <action application="park" /> </condition> </extension> So once the event is fired, the call is parked to prevent further execution within the dialplan. From there on, SafiServer is controlling the call via Inbound Mod event socket. So this works perfectly, except that if the invoked Saflet doesn't explicitly hang-up the call it will remain parked until the caller hangs up. My question is, is there a better way to do this? Is there some better alternative to park in this case? Ideally I'd like to initiate a 'session' of some kind when the SafiServer is "controlling" the call and then exit that session as soon as the Saflet is complete, at which point the call would continue on to the next entry in the dialplan. I understand I could use Outbound sockets to achieve this but, as I mentioned, I'd like to avoid the overhead of a separate socket connection for each incoming call. I actually have a mod_saficall.c app that does basically the the same thing as I described in the dialplan entry. Perhaps there's something more I could do in code that would allow me to be notified when the session is complete. Here's the relevant code I have so far: switch_channel_t *channel = NULL; switch_event_t *event; const char *safiCallFlag = NULL; channel = switch_core_session_get_channel(session); safiCallFlag = switch_channel_get_variable(channel, "saficall"); if (!safiCallFlag) switch_channel_set_variable(channel, "saficall", "true"); if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, "saficall::incoming") == SWITCH_STATUS_SUCCESS) { switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "new_saficall", safiCallFlag ? "false" : "true"); switch_channel_event_set_data(channel, event); switch_event_fire(&event); switch_ivr_park(session, NULL); } Any ideas you might have on this are welcome. Thanks, Zac Wolfe Safi Systems LLC www.safisystems.com

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incoming call stop working after a few minutes
by Philippe Le Toquin 08 Apr '11

08 Apr '11

Hi, I have the latest FS installed on my Guruplug and it was working fine overall Last Friday I decided to rename the hostname of the guruplug and since then I have problem with incoming calls no longer going through. I can't see how the 2 can related but that is the only thing I can think of. Also if I make a outgoing call then incoming will start working again for a few minutes and stops again The only thing I can see in the debug is that message: 2011-04-03 12:39:00.820680 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [8926(a)172.21.0.20] from ip 172.21.0.50 But I was seeing it before as well so it can't be only that. Is there some other trace I can activate to see more? Regards /Philippe

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Setting SIP request URI in bridge
by Pete Kelly 08 Apr '11

08 Apr '11

Hi I need to bridge a call to an IP a.b.c.d, however the request URI needs to be sent to a domain rather than an IP e.g. the UDP packet is sent to a.b.c.d, but the INVITE looks like "INVITE sip:12345@my.domain.com" This is the syntax I am using for the bridge: <action application="bridge" data="{sip_invite_req_uri= sip:12345@my.domain.com,sip_from_uri=sip:$1@my.domain.com ,codec_string='G729,PCMU,PCMA'}sofia/mediaserver/12345(a)a.b.c.d"/> however freeswitch just 503's the bridge with "NORMAL_TEMPORARY_FAILURE". If I remove the sip_invite_req_uri part, the call is bridged fine, but the INVITE line is incorrect. Is it possible to do what I need in Freeswitch? Thanks Pete

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how to send a "uuid_displace" command in C++ code?
by Frankie Yiu 08 Apr '11

08 Apr '11

Hi there, I would like to know how I can send a "uuid_displace" command in my c++ code. In C#, I can call Api.ExecuteString("uuid_displace " + <uuid string> + " start " + <strFileName> + " 20"); but in C++ how can I send this command? Hope someone can help me. Thanks in advance!! Frankie

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How to replay an audio when a ' * " (star key) is pressed?
by Frankie Yiu 08 Apr '11

08 Apr '11

Hi there, I would like to play an audio to the channel and then if the callee presses a ' * ' and the audio would replay immediately from the beginning, how can I do that? I am using C#. Currently I am calling PlayAndGetDigits() and would play an audio. I am also subscribing a DTMF event, when I find a DTMF = ' * ' I would call this command "uuid_displace " + <unique ID> + " start " + <file name> + " 20" " to play the audio again. The problem is that it seems like the event come in after 6 or 7 sec after the callee presses the key. What am I doing wrong or is there a better way to do this? Thank you. Frankie

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FS -How to limit the max number of registration users
by fieldpeak 08 Apr '11

08 Apr '11

Could anyone help advise how to limit the max number of registration users on FS? Thanks if any advice. Regards, Charles

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Transfer attempt for a previously a replaced call fails
by Joegen E. Baclor 08 Apr '11

08 Apr '11

Hi List, I have a scenario where a bridged call has been replaced due to a consultative transfer. This works pretty well and audio is bidirectional. I have the original uuid of the call in a var somewhere. The trouble begins when I uuid_deflect the bridged call once again to attempt another transfer. Sofia disconnects the channel. I am using the original uuid of the call (uuid prior to replaces). Is this the right way of doing it? Joegen

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Freeswitch
by Kaushal Shriyan 07 Apr '11

07 Apr '11

Hi, I have couple of questions regarding Asterisk. a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of phone numbers? b) Does it Support VoiceXML ? c) What PRI Card is recommended for using Asterisk ? Thanks Kaushal

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