Re: [Freeswitch-users] Freeswitch and Kamailio (OpenSer) Integration
Yes it is possible but there is no documentation on how to do it. You will have to learn SIP and understand what you are doing. Forwarding the call to FS for nat may cause issue as FS will then not have direct connection to the phone and may not be able to always detect it is behind NAT. Have a look at SIP PATH Extension as it is what you need to force traffic back to KAM from FS. Regards, Thomas On 6 Mar 2009, at 08:51, Ramu wrote: Hi All, I would like to setup freswitch and kamailio as follows: Kamailio acts as Proxy and Registrator Freeswitch acts as a SBC and MediaServer (voicemail) Users will be reigstered to Kamailio Kamailio forwards calls to FS to NAT FS sends back INVITE to Kamailio Kamailio will dial-out user. Bob calls Alice Bob ==INVITE ==> Kamailio ==INVITE==> FS ==INVITE==> Kamailio ==INVITE ==> Alice How can I achieve this scenario? Can you please direct me to any documentation which is available? Thanks, Ramu ___ Freeswitch-users mailing list 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 ___ Freeswitch-users mailing list 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
Re: [Freeswitch-users] FreeSWITCH VPSs
Hello Brian / Everyone,
Like Nik and Nicolas, I created a openvz box to test things in a 'near
production' environment. The box does only take 'test' calls, ie it
never saw more that a few calls at a time.
The design was 100% openser/opensips/kamailio but I since replaced the
pstn gateways with FS and it has been working perfectly and I am not
looking back.
I am now planning to use freeswitch as the registrar/voicemail/media
servers to only keep openser as a proxy inbound proxy (as it is
possible to program it to assign a RTP proxies topologically near the
caller and you can use it to fix some really broken sip packets - like
LLU operators cheap DSL routers badly NAT fixing the contact header).
Following this thread I am wondering if I should/could expect some
issues with my setup which is 32 bits or if your comments are only
related to the performance/behaviour of FS once under load (in which
case I need not to worry).
voip-master:~# uname -a
Linux voip-master 2.6.18-ovz-028stab053.5-smp #1 SMP Sat Mar 1
12:19:31 CET 2008 i686 GNU/Linux
voip-master:~# vzlist
VEID NPROC STATUS IP_ADDRHOSTNAME
1001 24 running A.B.C.Aproxy1.sip (openser phone
outbound proxy - accept REGISTER - range locked)
1002 24 running A.B.C.Bin1.sip (openser incoming
calls from the net - enum, no REGISTER - open)
1003 19 running A.B.C.Cout1.sip (openser outgoing
calls to the net - enum - to be FS)
1004 4 running A.B.C.Drtp1.nat (rtpproxy nat)
1005 3 running A.B.C.Emedia1 (was sems for
voicemail/media)
1006 29 running A.B.C.Fdatabase1
1107 23 running A.B.C.Gregistrar1.sip (openser)
1108 26 running A.B.C.Hregistrar2.sip (FS)
1109 8 running A.B.C.Ins1(auth DNS for the tested
zone with ENUM info)
1110 8 running A.B.C.Jinternal1.cache (cache with
internal ENUM routing)
8 running A.B.C.Kexternal1.cache (normal DNS
cache)
1112 21 running A.B.C.Lpstn-out-1 (FS gateway out
to pstn)
1113 21 running A.B.C.Mpstn-in-1 (FS gateway in
from pstn)
voip-master:~# cat /proc/cpuinfo | grep model
model : 4
model name : Intel(R) Xeon(TM) CPU 3.00GHz
model : 4
model name : Intel(R) Xeon(TM) CPU 3.00GHz
voip-master:~# free -m
total used free sharedbuffers
cached
Mem: 2023 1902120 0431
1009
-/+ buffers/cache:460 1562
Swap: 2588 3 2585
yep, memory is short :p)
Regards,
Thomas
On 11 Feb 2009, at 22:54, Brian West wrote:
> Your VE must be 64bit also.
>
> http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora
>
> If you need the set util listed on that page let me know I have a
> copy of it.
>
> /b
>
> On Feb 11, 2009, at 4:47 PM, Nik Martin wrote:
>
>> I think the VE I've built is too, but uname is a bit cryptic:
>> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009
>> i686 i686 i386 GNU/Linux
>>
>> I can easily change it if FS will run better.
>
>
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Re: [Freeswitch-users] Clustering FreeSWITCH
Hi, Sorry for jumping in without reading the whole thread correctly but I will most likely not have the time to do so before Saturday but would still provide some feedback. When using Openser as a customer phone proxy, it is my understanding that unless you use the PATH extension you will have problem with firewalls as the packet from Freeswitch will not match the opened IP address for the UDP port. In the scenario where the phone send a REGISTER this way : Phone -> Firewall (create an association) -> OpenSer -> Freeswitch. If Freeswitch answers directly to the packet, it will not match the association on the firewall for the return packet (different IP) and the connection may/will be blocked, so the reply to the REGISTER must come back via OpenSER. Same thing for the INVITE message, if FS tries to connect the firewall directly the packet will be blocked. As well, the firewall mapping need to be kept open with some kind of SIP pinging using OPTIONS - for example - and going through OpenSER as well from the FS server When quickly looking into it a few weeks ago - before I was side tracked, I found that the way OpenSER and FS way of using SIP PATH are not working together very well (I was planning a mail to the list with more info once I checked this more extensively). I can see the point with trying to not use OpenSER and use DNS SRV but you may not always be able to do without. Regards, Thomas ___ Freeswitch-users mailing list 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
Re: [Freeswitch-users] Newbie: Avaya SES <>Freeswitch 407 Proxy Authentication error
Hi, it is for loose route for the details see http://www.faqs.org/rfcs/rfc3327.html Thomas On 15 Oct 2008, at 09:00, Gayatri Kulkarni wrote: Record-Route: Record-Route: what's the 'lr' next to the port number? Did you notice the Content-Length for your INVITE is 0? Usually taking an etheral trace and analyzing it helps in case of Avaya SES - even for administration issues -- Regards, Gayatri Kulkarni On Tue, Oct 14, 2008 at 7:48 PM, Gerry Hull <[EMAIL PROTECTED]> wrote: On Mon, Oct 13, 2008 at 12:59 PM, Brian West <[EMAIL PROTECTED]> wrote: You need to add otherwise the register contact will be the username. aka 3824, its trying to route to 3824 in context public. /b On Oct 13, 2008, at 9:53 AM, Gerry Hull wrote: Hi Brian, Still no luck. I guess I'm still missing something. Let me explain and provide more details. We have several Avaya extensions registered in FS; we have the Avaya switch direct inbound calls to these extensions. The idea is to park the inbound calls in FS; we will then transfer the calls later using event_socket. However, we cannot get FS to answer the calls due to the proxy authentication error. Here's the configuration: /sip_profiles/internal/AvayaInternal.xml: /dialplan/public.xml: and here's the debug info: INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE From: "Station 216" 5061>;tag=05c8e85a1a5dd14da549c5a6a00 Record-Route: ,> To: "3823" Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 Content-Length: 150 Content-Type: application/sdp Contact: "Station 216" Max-Forwards: 69 User-Agent: Avaya CM/R014x.00.1.731.2 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS Accept-Contact: *;+avaya-cm-line=1 Supported: 100rel,timer,replaces,join,histinfo Alert-Info: ;avaya-cm-alert-type=internal Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: "Station 216" History-Info: ;index=1,"3823" [EMAIL PROTECTED]>;index=1.1 v=0 o=- 1 1 IN IP4 10.0.2.151 s=- c=IN IP4 10.0.2.152 t=0 0 m=audio 3188 RTP/AVP 0 127 a=rtpmap:0 PCMU/8000 a=rtpmap:127 telephone-event/8000 tport(01833120): msg 01872960 (1194 bytes) from udp/10.0.2.154:5060/ sip next= nta: received INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 (CSeq 1) nta: canonizing sip:[EMAIL PROTECTED]:5060 with contact nta: INVITE (1) going to a default leg nta: timer set to 200 ms soa_clone(static::017FD508, 017FED48, 0188E450) called soa_set_params(static::0189BB50, ...) called nta_leg_tcreate(00FEDDE8) soa_init_offer_answer(static::0189BB50) called soa_set_remote_sdp(static::0189BB50, , 0189E804, 150) called nua(0188E450): adding session usage tport_tsend(01833120) tpn = UDP/10.0.2.154:5060 tport_resolve addrinfo = 10.0.2.154:5060 tport(01833120): not found by name UDP/10.0.2.154:5060 tport_vsend(01833120): 515 bytes of 515 to UDP/10.0.2.154:5060 tport_vsend returned 515 send 515 bytes to udp/[10.0.2.154]:5060 at 14:09:50.666454: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.154:5060;branch=z9hG4bK03030353535336363671ef.0,SIP/2.0/TLS 10.0.2.151;psrrposn=2;received=10 .0.2.151;branch=z9hG4bK05c8e85a1a5dd14fa549c5a6a00 Record-Route: Record-Route: From: "Station 216" 5061>;tag=05c8e85a1a5dd14da549c5a6a00 To: "3823" Call-ID: 05c8e85a1a5dd14ea549c5a6a00 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9570M Content-Length: 0 nta: sent 100 Trying for INVITE (1) nua(0188E450): event i_invite 100 Trying nua(0188E450): call state changed: init -> received, received offer soa_get_remote_sdp(static::0189BB50, [029EFC10], [029EFC0C], []) called nua(0188E450): event i_state 100 Trying nua(0188E450): sent signal r_respond nua(0188E450): sent signal r_destroy nua(0188E450): event i_state dropped nua(0188E450): recv signal r_respond 407 Proxy Authentication Required soa_set_params(static::0189BB50, ...) called soa_clear_remote_sdp(static::0189BB50) called tport_tsend(01833120) tpn = UDP/10.0.2.154:5060 tport_resolve addrinfo = 10.0.2.154:5060 tport(01833120): not found by name UDP/10.0.2.154:5060 tport_vsend(01833120): 893 bytes of 893 to UDP/10.0.2.154:5060 tport_vsend returned 893 send 893 bytes to udp/[10.0.2.154]:5060 at 14:09:50.697704:
Re: [Freeswitch-users] Newbie Questions
Another way would be to export to a Radius server using a MySQL backend. I am not using the feature so I can not report how well it works but I am confident that "it should just work" and that if it does not, it will be fixed :) Thomas Vito Andolini wrote: > Michael, > > Thanks a lot for the answers, it answers my questions, I guess the next > thing would be downloading and trying FS :) > > Also, as I saw in the wiki, there is no built in support for mysql > right? The solution that was suggested was setting up a cron to import > the csv files into mysql, is this still what's available or are there > any updates on this? > > Thank, > > Vito A. > > > *From:* [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] *On Behalf Of > *Michael Jerris > *Sent:* Friday, October 03, 2008 7:06 AM > *To:* freeswitch-users@lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Newbie Questions > > > On Oct 3, 2008, at 4:54 AM, Vito Andolini wrote: > >> Hi All, >> >> I am familiar with Asterisk and doing some testing for my next >> project. I have had some difficulties Asterisk, and now researching >> FreeSwitch hoping that it has some out-of-the box answers for my >> questions. >> >> Basically I want to implement a Click 2 Call service. Very simple, >> user types in his/her number on a website, that number is called, >> after it's answered, the company number is called and they are bridged >> together. >> >> 1) Is there a way to communicate with FreeSwitch programatically and >> issue commands such as initiate calls etc ? (ver much like manager API >> in Asterisk) > > There is an interface that we call the fsapi interface that can be > accessed in many ways, including over a socket method similar to a > combination of AMI and FAGI: > > http://wiki.freeswitch.org/wiki/Event_Socket > > and xmlrpc: > > http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC > >> 2) If you initiate a call from the software and then once its answered >> call a 2nd number. How do you bridge them? > > You can do this all in one command: > > http://wiki.freeswitch.org/wiki/Mod_commands#originate > >> 3) After the 2 numbers talk and hang up. How does your cdr look like? >> Do you have 2 cdr's that correspond to both calls or just 1 after both >> numbers are bridged together? This is one of the problems I can't >> solve with Asterisk as it generates only 1 cdr after the 2 calss are >> bridged. The reason for this request is, in case of a Click2Call >> service, you are charged for both calls by your SIP provider therefore >> you need to be able to track both calls for invoices/payments etc. > > We can do either per leg or combined cdr's > > http://wiki.freeswitch.org/wiki/Channel_Variables#process_cdr > > We have multiple supported formats for cdr: > > http://wiki.freeswitch.org/wiki/Mod_xml_cdr > http://wiki.freeswitch.org/wiki/Mod_cdr_csv > >> 4) Is there a way to programatically know if a call has been asnwered >> or not and act based upon that. I understand the cdr contains that >> information. But what I want is, if the call is not answered maybe I >> can play a prerecorded message or take them to the voicemail or >> whatnot. So I need a way to do a flow-control based on if the call has >> been asnwered or not in the dialplan. Does that exist? If so can you >> point me to some resources? > > There are several approaches you could take to this. You could do this > all in dialplan if there is not any real forking other than if the call > worked. You can use the variables: > > http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge > http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail > > and just playing the sounds in the dial-plan after a bridge line. > > > Mike > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.173 / Virus Database: 270.7.5/1702 - Release Date: > 10/2/2008 9:35 PM > > > > > ___ > Freeswitch-users mailing list > 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 ___ Freeswitch-users mailing list 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
Re: [Freeswitch-users] Default config changes
Jeremy,
Short answer:
The commercial version of zoiper http://www.zoiper.com/ (30 euros) has
the features you request listed but I only ever used their 'lite' version.
Long answer:
I found the 'lite' version on Leopard useful but it has is weirdness -
some I would attribute to be a stripped down test product.
I had codec negotiation problems with x-lite: things getting confused
between a and u-law.
The lite version had clear limitation once you start to use it for more
than have a look:
* Some configuration options are not read and some other changes on
restart (a way to enforce it 'lite' status I assume).
* some URI characters restrictions ("-+" are in an configurable invalid
characters list, which you can not change on the 'lite' product)
* no support for the [EMAIL PROTECTED] syntax - it becomes user%40domain !
thank you openser for fixing that for me (... oups wrong list :) however
nothing can fix correctly a stripped '-' in the domain name.
When I contacted their support about the last point, they told me the
sip:[EMAIL PROTECTED] syntax should work but I guess it is only available on
their commercial version.
On the plus side, it does not take days to start like x-lite does on my
macbook when I use wireless.
Thomas
Jeremy Kiffiak wrote:
> Brian,
>
> For those of us who would like to check out the changes but who don't
> inherently know how to "check it out" would you mind giving some
> simple step-by-step instructions? Much appreciated.
>
> Additionally, do you know of an OS X (Leopard) softphone that will let
> me test out TLS and SRTP? Thank you.
>
> Jeremy
>
>
> On 28-Sep-08, at 6:44 PM, Brian West wrote:
>
>> FreeSWITCHers,
>>
>> I have been working to make the default configuration multi-domain
>> aware. So please check it out. I have made a few more changes that
>> were recommended by the community and added some documentation into
>> the config and some warnings that shouldn't be ignored! ;)
>>
>> The extensions no longer plays the bong tone. It plays a file
>> saying "this call has been secured" and it only does so if TLS and
>> SRTP are active. Before it would do it only on SRTP which wasn't
>> really secure in my opinion since you just sent the crypto key in
>> plain text only milliseconds before.
>>
>> Please check it out and let me know if you find any bugs or have input
>> for additions.
>>
>> /b
>>
>>
>>
>>
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