May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

My SIP provider is stripping phone numbers !
by rico-freeswitch＠ricozome.net 25 May '12

25 May '12

Hi list, I have a stupid issue with my SIP provider "SCT Telecom" (in France) : it strip the first digit of the caller pĥone number ! After many support requests they finally reconised the problem is at their side, but their answer is they'll not fix at their side and I must fix it at my side :-/ So I had a look on variables on public context and I see this : Channel-Name: [sofia/external/661864068(a)sct-voip.fr] Caller-Username: [661864068] Caller-Caller-ID-Name: [661864068] Caller-Caller-ID-Number: [661864068] Caller-ANI: [661864068] Caller-Channel-Name: [sofia/external/661864068(a)sct-voip.fr] variable_sip_from_user: [661864068] variable_sip_from_uri: [661864068(a)sct-voip.fr] variable_sip_from_user_stripped: [661864068] variable_sip_full_from: [<sip:661864068@sct-voip.fr;user=phone>;tag=21b05c16] variable_sip_contact_user: [661864068] variable_sip_contact_uri: [661864068(a)86.64.146.248] variable_channel_name: [sofia/external/661864068(a)sct-voip.fr] variable_caller_id_number: [661864068] The expected caller number is 0661864068, and as you can see is stripped to 661864068 So I added this in the begining of my public dialplan : <extension name="public_sct" continue="true"> <condition field="destination_number" expression="^01645327|^01645380|^01714170"> <action application="ring_ready"/> <action application="set" data="ringback=${fr-ring}"/> <action application="set" data="call_direction=inbound"/> <action application="set" data="caller_id_number=0${caller_id_number}"/> </condition> </extension> It works as expected as the variable caller_id_number is updated : variable_caller_id_number: [0661864068] But when it starts to the default context dialplan, the caller number is still truncated as 661864068. So I wonder I handle the problem the wrong way... Any help would be appreciated :) Thanks and regards, -- Rico

3 2

Weekly Conference Call
by Ken Rice 25 May '12

25 May '12

Hey Guys, Don¹t Forget today 1PM EST5EDT (that¹s 10AM Pacific for you guys on the West Coast of the US) FreeSWITCH community conference call. We¹ll be reviewing Dialplanning in C and doing an update on FreeSWITCH 1.2 release cycle. How to join you ask? See http://wiki.freeswitch.org/wiki/Weekly_Conference_Call there are several ways you can join from SIP to PSTN or even Skype. K

2 3

Re: [Freeswitch-users] destination sip port ?
by Peter Olsson 25 May '12

25 May '12

Try setting "register-proxy" instead. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Evtim Ivanov Skickat: den 25 maj 2012 14:22 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] destination sip port ? Hello All I don't want to waste your time , so I'll be as short as possible. My question is : How to change the destination sip port to one of the gateways (providers). This provider listen not on standard sip port 5060 but on 5069. I tried to set : <param name="proxy" value="provider.com:5069<http://provider.com:5069>"/> but it doesn't work this way. Again trying to send register on 5060 (I dumped). I also searched in the web but didn't find the answer till now. So, what is the proper way? Thanks in advance and Best regards Evtim !DSPAM:4fbf7d7a32761808049375!

2 1

destination sip port ?
by Evtim Ivanov 25 May '12

25 May '12

Hello All > > I don't want to waste your time , so I'll be as short as possible. > My question is : How to change the destination sip port to one of the > gateways (providers). This provider listen not on standard sip port 5060 > but on 5069. > I tried to set : <param name="proxy" value="provider.com:5069"/> but it > doesn't work this way. Again trying to send register on 5060 (I dumped). I > also searched in the web but didn't find the answer till now. > So, what is the proper way? > > Thanks in advance > and Best regards > Evtim >

1 0

FS as a dumb SBC (was: Re: Help on Explaination of the configurations parrametters)
by Paul Cupis 25 May '12

25 May '12

On 23/05/12 09:26, Albert Nguyen wrote: > I am new to FS and trying to setup FS as a "dumb" SBC as per > example 2 in the wiki website so that I can convert inband DTMF to > RFC 8233 and vice versa. This is the windows version of FS. Is it > possible that I can get some assistant from you guys to get it up and > running? Please advise how we can proceed. One way of doing this would be to setup two SIP profiles, one facing the device sending you RFC2833 and once facing the device which will only accept in-band. On the SIP profile facing the in-band device, disable RFC2833 support by setting the varaible "rfc2833-pt" to something <95, and then when you route a call from RFC2833 to in-band, add something like: <action application="export" data="execute_on_media=start_dtmf_generate"/> to your dialplan. Similarly you will want some config for calls going in-band->rfc2833 to do the conversion, perhaps something like: <action application="start_dtmf" /> Note that I've not specifically tested the above. If you are going down the above route, you might also want to read: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate http://wiki.freeswitch.org/wiki/DTMF Regards,

2 1

RTP media issue
by Nathan Downes 25 May '12

25 May '12

Hi, I had previous reported an issue with poor voice quality, appearing to stem from occasion wrong timestamps coming from provider, but the end user's experience was much worse than what I could see/hear in the trace. I have finally captured an event inbound and outbound. The thing I don't understand is I thought even though FS proxied the media it didn't touch it or change anything, but it appears it is. The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and http://www.nortec.com.au/outbound.pcap.gz Inbound is from my trunk provider to FS box and outbound is FS box to ATA in FTTH GPON. The event I am talking about, if both traces are open, is in the inbound one inbetween packet 8114 and 8117 the provider drops a packet or I don't receive it. In the corresponding outbound trace, between packet 8144 and 8152, it appears FS misses a whole heap of packets (.1 seconds) between 8146 and 8152 then it increases the timestamp only by 40 rather than 160 on packet 8152. This seems to not affect SIP phones themselves but causes issues with the FTTH GPON ATA. This causes a gap in the audio for the end user, and when they miss a high number of packets even though it sounds good on the inbound trace the end users experience is horrible. This trace is actually a good one, but the wrong timestamp can occur once per second, causing end user to lose 10%+ of incoming audio only. The issue only affects the audio coming from provider to FS to end user. I am chasing it up with the voice provider to try and eliminate the occasional packet loss, but if I could stop/fix FS from doing its adjustment/gap/something the end user wouldn't even notice it.

2 1

Can an inbound call be associated with a gateway?
by Richard Brady 24 May '12

24 May '12

Hi folks I'd like to know whether FreeSWITCH is able to associate an inbound call with a gateway. On the one hand I see documentation of variables which can be added to a gateway with the direction="inbound" attribute, which makes me think it's possible. On the other hand, I cannot see how FreeSWITCH binds an incoming call which is allowed by an ACL and therefore accepted but not challenged (so it is IP authenticated) to a gateway. If the binding is by IP address, where in the configuration is the association of the address or CIDR range to the gateway? I get a feeling that gateways are an outbound only concept, but the documentation seems to contradict this. Any clarification would be most helpful. Thanks, Richard -- Richard Brady

1 0

FreeSWITCH logs to C# application
by coopara 23 May '12

23 May '12

Hi, I would like to collect all FS logs to my C# application. Is there any possible way to do that? I use FreeSWITCH.EventBinding.Bind function and I'm looking for event_id which is switch_event_types_t.SWITCH_EVENT_LOG in handler but unfortunately it never occurs. Best regards, Coopara

2 1

How to set "keep-alives re-use the TCP connection" ?
by Mario G 23 May '12

23 May '12

Using Bria on iPad with a TCP connection, all works except after several hours it no longer registers to FreeSwitch. The Bria support says if this happens set the server PBX to use "keep-alives re-use the TCP connection". I could not find this options in the wiki, only thing close dealt with NAT but FS and the phones are all on local lan. Anyone know how to set this in the user definition? Thanks, Mario G

2 4

Re: [Freeswitch-users] Freswitch ODBC tables
by Peter Olsson 23 May '12

23 May '12

I don't think there is much documentation about this - since it's mostly data used internally by FS. What do you want to do more specifically? I guess most of the tables are quite self-explaining by their names.. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Gregor Nanger Skickat: den 23 maj 2012 09:13 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Freswitch ODBC tables Please give me hint what to search :-( I tried ODBC tables, sip_registration table, sofia tables, sofia ODBC and tables are mentioned, but no single description for what tables are used. I figure it out that in registration table are stored active sip registrations with expire date etc..., sip_registration tables has same information, but with more fields. In channels table are stored all legs information when call is active... In Calls are stored active calls when bridge happened between legs... Can you give me a hand for other tables? 2012/5/23 Gregor Nanger <gregor(a)infomedia.si<mailto:gregor@infomedia.si>> Ok Brian, you got me thank you... 2012/5/23 Brian Foster <bdfoster(a)endigotech.com<mailto:bdfoster@endigotech.com>> All over the wiki, my friend. -BDF On May 22, 2012 6:07 PM, "Gregor Nanger" <gregor(a)infomedia.si<mailto:gregor@infomedia.si>> wrote: Hi! I have FS in linux and moved sofia db via ODBC to MS Sql server and it works ok. Are there any documentation about tables and how they are used by sofia? Thanx, Gregor _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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 Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@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<mailto:consulting@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 Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbc8cbd32761286258325!

4 4