- FreeSWITCH-users - FreeSWITCH Mailing Lists

Where do I find the ANI in SIP packets?
by Don Coe 07 Dec '11

07 Dec '11

I have a couple of carriers that terminate calls for me. Most send my ANI and it shows up, for instance, if I call my cell. However, some carriers are telling me that I am not sending ANI and on those calls I am seeing unknown numbers on my cell. How can I verify I am actually sending ANI and where would I find the ANI in the SIP packets? --Don

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183 Session Progress Missing Video Codecs
by Jerry Richards 07 Dec '11

07 Dec '11

Using Nov 28 Freeswitch, if I make a Bria video call that includes H.263, H.263-1998 and H.264 in the caller's INVITE, Freeswitch sends 183 Session Progress back to caller with H263/90000 only. I am running will all default configs, except I added "H263-1998,H263,H264" to the list of global_codecs and outbound codecs in vars.xml. I'm thinking this might be a bug? Thanks, Jerry

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FreeSWITCH Conference Call Today
by Michael Collins 07 Dec '11

07 Dec '11

Hello all! We are having a nice conference call today. First off, Brian West has a few items to discuss with the community. After that, I will be doing a brief primer on using the FreeSWITCH event socket and ESL. Agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_07 Talk to you soon! -Michael

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Does I need jitter buffer
by Boris Kovalenko 07 Dec '11

07 Dec '11

Hello! Would You please explain does I need jitter or not? As I wrote about a month ago I offten have an echo when calling local_user -- freeswitch -- localuser. And _never_ have an echo when calling local_user -- freeswitch -- remote_gateway. My local network is gigabit based. remote_gateway is on internet. Will the jitter help me and what settings should I start from? -- Regards, Boris

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FS supports RFC 5626
by Beppe Grillo 07 Dec '11

07 Dec '11

Hy all, FreeSWITCH supports RFC 5626 ? I FS with the last git. Thanks, Giuseppe

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Need dial plan for Analogue FreeTDM
by Ryan V 07 Dec '11

07 Dec '11

Hi, We have 1 PRI and 3 Analog lines. We are using Sangoma A200 with 4FXS and 4FXO ports. Also, we have a Sangoma A101 card. Incoming calls are working fine. But, we are unable to dial out. Here is our dial plan <extension name="outgoing_bsnl"> <condition field="destination_number" expression="^9(\d{10})$"> <action application="bridge" data="freetdm/3/1/$1"/> </condition> </extension> Here are the logs from mod_freetdm. 2011-12-07 14:03:54.429502 [DEBUG] mod_freetdm.c:1227 Connect outbound channel FreeTDM/3:1/9449905000 2011-12-07 14:03:54.429502 [DEBUG] mod_freetdm.c:1236 (FreeTDM/3:1/9449905000) State Change CS_NEW -> CS_INIT 2011-12-07 14:03:54.429502 [DEBUG] mod_freetdm.c:1253 Attached session 32ad9cea-20ae-11e1-b313-21737ba18370 to channel 3:1 2011-12-07 14:03:54.929503 [DEBUG] mod_freetdm.c:435 (FreeTDM/3:1/9449905000) State Change CS_INIT -> CS_ROUTING 2011-12-07 14:03:54.929503 [DEBUG] mod_freetdm.c:458 FreeTDM/3:1/9449905000 CHANNEL ROUTING 2011-12-07 14:03:59.489509 [DEBUG] mod_freetdm.c:1992 got FXO sig 3:1 [STOP] 2011-12-07 14:03:59.489509 [NOTICE] mod_freetdm.c:2013 Hangup FreeTDM/3:1/9449905000 [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] I don't know where we are making mistake. Thanks, Ryan

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Google Voice
by Jacob Smith 07 Dec '11

07 Dec '11

I am near admitting defeat on this. If anyone can tell me what I am doing wrong, I would really appreciate it. I call my Google voice account, the FS phone (ext 1000) rings, I answer the phone but the calling phone keeps ringing and after 8 or so rings disconnects. I can make outgoing calls with no problems. Here is the full log, all I did was start it up/call/shutdown and replaced the calling phone number with "phone number" http://pastebin.freeswitch.org/17954 Here is my dial plan: http://pastebin.freeswitch.org/17956 Here is my Jingle profile: http://pastebin.freeswitch.org/17955 Thank you! Jacob

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pagd dynamic conference
by Rodney 07 Dec '11

07 Dec '11

>>>> I changed it to 10000 and 15000 with same results. it does send me to "conference" but not with any numbers. seems the read feature worked without givng me conf options but pagd doesnt really work at all. i must still be missing something. Hmmm, maybe the timeout is too short. Change the 2000 (2 seconds) to 7000 (7 seconds) and see if that makes any difference. -MC On Mon, Dec 5, 2011 at 11:41 PM, Rodney <notlikeme75(a)yahoo.com> wrote: >>> Michael, I tried what you suggest and it just plays conf-pin.wav then goes right to bad-pin.wav then hangups without actually putting me in conference.before i tried this i tried using "read" instead of pagd and it accepted the pin and put me into the conference but didn't allow me any conference options or the ability to back out to main ivr. I am still stuck :( . my system is ivr based incoming calls only so this feature is essential. all help is appreciated. > > > condition destination_number ^3001$ > action answer > action play_and_get_digits 4 4 3 # 2000 conf-pin.wav conf-bad-pin.wav \d+ > action conference ${digits}-${domain}@default > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >I believe you just need to call play_and_get_digits (PAGD) prior to dropping them into the conference. If I understand the question correctly, the four-digits that the caller enters represents the "conference number" or whatever you call it. You could do this: > > ><action application="play_and_get_digits" data="4 4 3 # plz_enter_conf.wav invalid.wav digits \d+ 2000"/> ><action application="conference" data="${digits}-${domain}@default"/> > > >naturally you'll need to read up on PAGD so that you understand what all that stuff is doing. Also, be sure to specify real sound files, not the fake ones that I used. The ${digits} value is what the caller actually dials. In a production environment you'll need to handle the scenario where the caller never actually enters a valid 4-digit number. > > >-MC > > >On Sun, Dec 4, 2011 at 11:44 AM, Rodney <notlikeme75(a)yahoo.com> wrote: > >I have tried the nb_conference example in the default.xml but I think I am doing something wrong. when i transfer to that extension it drops me into a single static conference of the variable . what do i need to add or change to prompt my callers transfered to the nb_conference to enter a 4 digit conference number without pin that gives them the same options as my static conferences on the default profile? I know play and get digits must be involved but would appreciate the help. thanks. >> >> >> <extension name="nb_conferences"> >> <condition field="destination_number" expression="^\*(30\d{2})$"> >> <action application="answer"/> >> <action application="conference" data="$1-${domain_name}@default"/> >> </condition> >> </extension> >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>http://www.cudatel.com >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>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 >> >> > >_______________________________________________ >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 > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > Chad, What is your event listener? Just curious. It seems to me that you could do an inbound event socket to generate the call and then let that call hit the dialplan and send it to the socket app which in turn would send the call to your event listener. (I suppose you could even do something like "bgapi originate sofia/foo/bar@baz &socket("127.0.0.1:8084")" if you wanted to.) -MC On Tue, Dec 6, 2011 at 8:10 AM, Chad Phillips -- Apartment Lines <chad(a)apartmentlines.com> wrote: I'm designing an application that will originate many calls via the event system which I also want to attach event listeners to. i want each event listener to only listen for events particular to each originated call. > >Since I want FreeSWITCH to control the call after I originate it, it seems that I should use an inbound connection. The two possible workflows I see are: > > a) Issue an originate command via inbound, then turn it into an outbound-type connection via a 'myevents <uuid>' call (or 'handlecall' using mod_erlang_event) once the channel is up. With this method, it seems that I could miss critical channel events between the time the call is originated and the listener is attached. > > b) Set up an inbound listener filtered on a pre-determined UUID for the call, then originate the call with that UUID. This would seem to guarantee that no events are missed by the listener, but I'm wondering about the performance of having potentially hundreds of inbound connections with filters going at the same time. > >So my questions are: > > 1) Is there a way to accomplish a) without potentially missing any channel events? > > 2) Is b) vastly less efficient than a)? > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > On 05/12/11 17:56, Rob Hutton wrote: >http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence > > > Rob, Indeed, that seems to have done the trick. Many thanks for that! Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) _______________________________________________ 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

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Voicemail Transcription Roundup
by Avi Marcus 06 Dec '11

06 Dec '11

I did some investigating and thought I'd share... Some info on pricing - free (google speech API) to $0.10/minute to approximately $2.80/minute are the options. See here: http://wiki.freeswitch.org/wiki/Transcribing_Voicemail And.. if you use fusionpbx, you can use the google speech API pretty easily with my patch from here: http://wiki.fusionpbx.com/index.php?title=Voicemail#Transcribing_Voicemails… Please share your experience or methods... you can update the wiki page. I'm not quite sure how to get the voicemail file to the transcription service with something other than the mailer script in the middle.. unless you are using ESL or polling the database of VMs and keep track if it's been processed yet. -Avi Marcus

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Created new profile for double-natted phones, calling these fails with USER_NOT_REGISTERED
by Alex Crow 06 Dec '11

06 Dec '11

Hi all, I have set up a new profile for phones behind NAT connecting to an FS box behind NAT. Calls in from a NAT'ed phone works fine, however calls out to one fail with USER_NOT_REGISTERED. All the phones are set up in /opt/freeswitch/conf/directory/default/<sip id>. I have added the following in the new profile config hoping it would resolve the issue but it does not: <param name="challenge-realm" value="$${domain}"/> <param name="force-subscription-domain" value="$${domain}"/> <param name="force-register-domain" value="$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> Any clues? Thanks Alex

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