August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

multiple sip profile , centrex like function
by Lee JJ 22 Aug '08

22 Aug '08

Hello Brian : I finally work something out ! # the dialplan <extension name="centrex testing"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> </condition> <condition field="${sip_from_host}" expression="^${sip_to_host}$"> <action application="bridge" data="sofia/${sofia_profile_name}/${dialed_ext}" /> <anti-action application="bridge" data="sofia/${sip_to_host}/${dialed_ext}" /> </condition> </extension> # the same EXT 1528@inter2 bridged with 1528@internal 2008-08-22 15:58:06 [NOTICE] sofia.c:2491 sofia_handle_sip_i_state() Channel [sofia/inter2/1528] has been answered 2008-08-22 15:58:06 [DEBUG] mod_sofia.c:430 sofia_answer_channel() Local SDP sofia/internal/1528(a)210.243.126.69: 2008-08-22 15:58:06 [NOTICE] switch_ivr_originate.c:1298 switch_ivr_originate() Channel [sofia/internal/1528(a)210.243.126.69] has been answered But there is one issue 2008-08-22 15:06:11 [WARNING] sofia.c:3130 sofia_handle_sip_i_invite() IP 219.81.16.134 Rejected by acl domains I have to disable the acl in profiles  otherwise it won't allow me to call "sip:1528@internal" from "sip:1528@inter2" ( PLS explain the security issue ) # the following diaplan tested , really nothing to do with ENUM very good DNS working <extension name="apnic Testing" continue="true"> <condition field="destination_number" expression="^(info|billing)"> <action application="bridge" data="sofia/default/$1@${sip_to_host}" /> </condition> </extension> 1 0.000000 210.243.126.69 -> 203.119.0.97 SIP/SDP Request: INVITE sip:billing@voip.apnic.net, with session description 2 0.246246 203.119.0.97 -> 210.243.126.69 SIP Status: 100 Trying 3 0.659271 203.119.0.97 -> 210.243.126.69 SIP Status: 180 Ringing 4 2.589506 210.243.126.69 -> 203.119.0.97 SIP Request: CANCEL sip:billing@voip.apnic.net Also been through these points. http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005531.h…

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Custom MOH
by mark morreny 22 Aug '08

22 Aug '08

Dear Freeswitch friends, I am working on making my own custom MOH specific to our company's need. I tried recording the content into a normal .wav file and then played it by setting ringback=<CUSTOMER MOH WAV PATH>. What I get is only a long series of beep sound instead of the recorded content. What did I do wrong? Is there any special format that I should use when creating a custom MOH? If I want to convert a mp3 file into MOH, how would I do that? Thanks, Mark

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How to originate a session from outside
by Adeel Ansari 22 Aug '08

22 Aug '08

Hi All, I had success in bridging a session, using my Java Program as a hook, and manage to connect one mobile phone to another. It goes like below. === --> originate sofia/gateway/gizmo2/01160176xxxxxx &java(net.digitalapps.freeswitch.CallTest) === In this case, freeswitch call my Java program. Now, My intension is to not use console at all, instead use my java program to originate a session in freeswitch and then bridge the session. So, call freeswitch from my program, not the other way around. Is it possible? Need suggestions in any case. I mean if I need to come up with my own client. Actually, I was looking JAIN-SIP to achieve this. But then thought of asking your suggestions. With JAIN I might go like this, place a call, hit the dial plan invoke my Java Hook Program and bridge the call. But I am afraid that it might connect my client with the mobile phone, not one phone to another. Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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Re: [Freeswitch-users] Fetching SIP Header Variable
by msp 22 Aug '08

22 Aug '08

Hi all, The solution is to add prefix "X-" before any custom variable, and this to be done on SIP proxy who is passing the call to Freeswitch. On the receiving site at freeswitch, we need to use prefix "sip_h_X-" to fetch those variables. This configuration works fine on my systems, thats all. Thanks you all for kind replys, msp

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Both phone rang, but no voice
by Adeel Ansari 22 Aug '08

22 Aug '08

Hi all, I have tried the command below, ==== originate sofia/gateway/gizmo1/01160176xxxxxx &bridge(sofia/gateway/gizmo2/01160122xxxxxx) ==== Its working, phone rang, but no voice. Am I missing something? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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test message please ignore
by Ilan Perez 22 Aug '08

22 Aug '08

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T1 RBS Support Revisited
by John Wehle 22 Aug '08

22 Aug '08

Anthony Minessale wrote: > It's more or less on hook and off hook and reading DTMF Hmmm ... I'm attempting to cobble together support and have hit a snag. Placing the channel off-hook and waiting for the wink is easier enough, however how do I sent the phone number? I believe e&m winkstart uses MF, * not * DTMF to send the phone number / extension ... at least my PBX doesn't respond when I send DTMF. Is there support in FreeSWITCH for MF? If not, then any pointers to how to go about adding support? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Custom events?
by James Green 21 Aug '08

21 Aug '08

I have noticed the wiki is pretty much non-existant on the subject of custom events. >From my understanding of the system thus far, I could create a number of custom events that my applications can react to and perform busienss logic. One specific example: I could connect to FS and fire an event "MYPROJECT::OUTBOUND_DIAL" which a separate application could listen for and process further. I might therefore have a chain of small apps developing these and reacting at various stages. Is my understanding correct? Can I therefore register an application to listen for MYPROJECT::MY_CUSTOM_EVENT and wait until another of my apps fires it? The next obvious question is how parameter handling works... But I shall try to walk before running... Thanks, James

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Query regarding FreeSWITCH
by Gayatri Kulkarni 21 Aug '08

21 Aug '08

Hi guys, Is there a way FressSWITCH exposes its list of APIs to third party clients? As in, I have a client and I want to invoke a FreeSWITCH API - Is there anywhere my application can look into to find out what is the list of parameters required? What would a SIP request to FressSWITCH querying about the API name and it's respective parameterlist look like? Does FressSWITCH support such a request basically? - I suppose it does, it has all of it's data in XML format. What would the response be like? Thanks a million, Gayatri Kulkarni.

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DNS SRV != ENUM
by Lee JJ 21 Aug '08

21 Aug '08

Hello Brian : >From the message http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005544.h… >* *>>* It should not query ENUM first . *>>* 21 11.653138 210.2x.x.x -> 202.y.y.y DNS Standard query NAPTR *>>* 2.9.9.9.e164.org * *<--- this is ENUM way , if I want to call an old fashion "Telephone No" you can see the domain always go to "E164.ORG"* >Again you told it to go to enum... isn't that exactly what it should >do. Why would we do a srv before that unless I'm missing your point >here. http://www.voip-info.org/wiki-DNS+SRV A Domain Name Server (DNS) SRV record helps connecting to a SIP user in a similar way that an MX record helps email delivery. http://mit.edu/sip/sip.edu/dns.shtml For example sip:billing@voip.apnic.net <sip%3Abilling(a)voip.apnic.net> sip:0702058888@sip.ttn.net <sip%3A0702058888(a)sip.ttn.net> The pure dns query will get either an "A" record ;; QUESTION SECTION: ;voip.apnic.net. IN A ;; ANSWER SECTION: voip.apnic.net. 3600 IN A 203.119.0.97 or a "SRV" record ;; QUESTION SECTION: ;sip.ttn.net. IN SRV ;; ANSWER SECTION: sip.ttn.net. 3600 IN CNAME nusip1.gogotalk.net.tw. for openser example forward(destination) Forward the SIP request to the given destination in stateless mode. This has the format of [proto:]host[:port]. Host can be an IP or hostname; supported protocols are UDP, TCP and TLS. (For TLS, you need to compile the TLS support into core). If proto or port are not specified, NAPTR and SRV lookups will be used to determine them (if possible). http://www.opensips.org/index.php?n=Resources.DocsCoreFcn#parameters for Asterisk sip.conf example srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet So they are totally different !

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