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ivr scripts and streamfile
by Jonas Gauffin 08 Apr '08

08 Apr '08

Hello Sometimes the sound files are not said properly. This usually happens for phrases that are said directly after session.answer, or after a "lenghty" function (as fetching xml using curl). For instance in my voicemail I have a phrase saying "You have no new messages" directly after session.answer. Sometimes I just hear "no new messages" when dialing in. What should I do? Thanks, Jonas

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preAnswer : API x JS
by Daniel Eli 08 Apr '08

08 Apr '08

Hi, first of all, congratulations to the FS team. I am trying to control FreeSwitch with a telnet client using the API. I can't find some way to pre-Answer (code 180) a channel in this interface, like the preAnswer method found in the session object. Another idea is calling an js script with the API, for example: bgapi jsrun preanswr.js ..., but there is another problem: the default session object is not created and I can't find how to get the session object from the uid of the channel (or from another channel variable). So, is there any way to get one session object from some channel variable ? Or, is there any way to pre-answer (generate the 180 code) the channel with the API ? Could you help me about that? Thanks in advance. Daniel. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

1 0

ENUM lookup and simultaneous dialing
by Mikael A. Bjerkeland 08 Apr '08

08 Apr '08

When doing an enum lookup where a number has two NAPTR records with the same preference and order enum_auto_route gets set to: sofia/gateway/openser/bordtelefon(a)192.168.1.154| sofia/gateway/openser/mikael(a)some-gw.com The "|" sets the dialstring for chained dialing. If the order and pref is identical for both records, shouldn't the dialstring be using "," to call both simultaneously? Is this a feature or a bug? Regards, Mikael

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About Dingaling
by Ali Jawad 08 Apr '08

08 Apr '08

Hi All I am trying to test the jingle features of FS. I have compiled and setup FS on debian all went well and I made a call using softphones from 1000 to 1001. After that I read http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working I did the changes and rebuilt I edited my config files my ./conf/jingle_profiles/client.xml looks as follows <include>  <x-profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="thisisme(a)gmail.com/talk"/> <param name="password" value="thisIsMyPwd"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/>  <param name="auto-login" value="true"/> <param name="auto-reply" value="Press *Call* to join my conference"/> <param name="exten" value="1000"/> <param name="ext-rtp-ip" value="stun:stun.xten.com"/>  <param name="sasl" value="plain"/>    <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/>  <param name="exten" value="888"/>    <param name="vad" value="both"/>  </x-profile> </include> As you can see above I used <param name="exten" value="1000"/> after that I logged in with my SIP softphone "Eyebeam" and tried to make a call to gmail account that did not work..but I was able to make a conference call to conf+888(a)conference.freeswitch.org The server is running on a PUBLIC IP So what must I do/change/add/check to able to make calls to my gmail accounts (or from my gmail account to my sip phone) as mentioned in the FAQ. And where can I debug this issue, if I call 9999 I can see the call setup and successful call in my putty shell, however when I call myaccount(a)gmail.com nothing happens. Thx Q: What? Did you say it can talk to GoogleTalk? Yes in March of 2006 I developed my own XMPP telephony signaling library that is capable of communicating with Google's GoogleTalk. With a single Jabber account you can receive endless simultaneous calls from GoogleTalk clients and gateway those calls to IVR or another voice protocol like SIP or H.323. When FreeSWITCH is on both ends of the call you can bypass NAT and send extended data such as Caller ID and DNIS. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 4

New LinkedIn Group
by Ken Rice 07 Apr '08

07 Apr '08

Hey Guys I have created a linked in Group for FreeSwitch... If you are a freeswitch user or developer and want to find other people on LinkedIn feel free to use the following link to join the group... http://www.linkedin.com/e/gis/84302/3EC65A7CD494 Ken http://www.cnam4voip.com/ krice(a)rmktek.com

1 0

freeswitch as a SIP Bridge
by Tim Meade 07 Apr '08

07 Apr '08

Greetings all, I've just stumbled upon your project and it may solve an issue we are having. I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need. We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk. Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now. One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together. So my question is: Can freeswitch do these things? Thanks and congratulations on the nice work!

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Route outbound-only calls through XML dialplan?
by tuhl＠ix.netcom.com 07 Apr '08

07 Apr '08

I've been using FS for almost a year now as a call-blaster type of device (originating SIP calls and playing wav/mp3 when user answers). In the past I was simply sending all calls to a single SIP switch, but now I need to think about routing to several different SIP switches, based on NPA-NXX for example. The XML dialplan would seem perfect for this, but that's only triggered by an incoming call, right? I am not sure how to cause 'outbound-only' calls to go through it (since Freeswitch is originating the call, it only has 1 leg, if that makes sense). Currently I am originating via XML-RPC from a Perl script, using a command like 'originate <sip address> &playback(wav file)'. The final caveat is that we not use javascript to originate the call because I am still having problems with garbage collection during capacity testing (the box is pushing out 20cps, 30-second calls, 600 simultaneous calls). Thanks, Tom =============== tuhl(a)ix.netcom.com

2 1

DTMF detect dialplan example
by jonathan augenstine 04 Apr '08

04 Apr '08

Can someone provide an example of how to collect DTMF in the dialplan? My search through docs and google has not clarified the steps to accomplish this task. I need to play a prompt and collect DTMF, with the ability to cut-through the prompt to start collecting digits. Jonathan

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Jingle To Sip Traffic
by Ali Jawad 04 Apr '08

04 Apr '08

Hi All We do have a jingle client and we do want to use that client to send traffic to sip devices, I have read in the FAQ that this is possible, has anyone done it ? Any advantages/disadvantages ? Any special issues I should be aware of ? Any good howto's ? Thx

2 1

My FS + Dingaling Experience With Qs
by Ali Jawad 04 Apr '08

04 Apr '08

Hi All I am trying to test the jingle features of FS. I have compiled and setup FS on debian all went well and I made a call using softphones from 1000 to 1001. After that I read http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working I did the changes and rebuilt I edited my config files my ./conf/jingle_profiles/client.xml looks as follows <include>  <x-profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="thisisme(a)gmail.com/talk"/> <param name="password" value="thisIsMyPwd"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/>  <param name="auto-login" value="true"/> <param name="auto-reply" value="Press *Call* to join my conference"/> <param name="exten" value="1000"/> <param name="ext-rtp-ip" value="stun:stun.xten.com"/>  <param name="sasl" value="plain"/>    <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/>  <param name="exten" value="888"/>    <param name="vad" value="both"/>  </x-profile> </include> As you can see above I used <param name="exten" value="1000"/> after that I logged in with my SIP softphone "Eyebeam" and tried to make a call to gmail account that did not work..but I was able to make a conference call to conf+888(a)conference.freeswitch.org Please note that my clients and server are both running in the same private LAN, I can put the server on a public IP if necessary. So what must I do/change/add/check to able to make calls to my gmail accounts (or from my gmail account to my sip phone) as mentioned in the FAQ Q: What? Did you say it can talk to GoogleTalk? Yes in March of 2006 I developed my own XMPP telephony signaling library that is capable of communicating with Google's GoogleTalk. With a single Jabber account you can receive endless simultaneous calls from GoogleTalk clients and gateway those calls to IVR or another voice protocol like SIP or H.323. When FreeSWITCH is on both ends of the call you can bypass NAT and send extended data such as Caller ID and DNIS.

2 1