- FreeSWITCH-users - FreeSWITCH Mailing Lists

freeswitch Scenario
by Yogesh Patel 17 Mar '08

17 Mar '08

HI Here what we want to do with the switch. Our scenario is like few wholesale carrier will sendus traffic for different destinations, which we will be sending to thedifferent vendors. So we were using 206.222.8.58 as our live switch to testfreeswitch. So we need to understand howfreeswitch is terming following things and where do it makes them so we cankeep adding more customers and vendors. Once it starts working for us. Scenario 1 customer name “Allianz” will be sending call to us from IP 209.62.77.114. presently we can configure this customer as a SIP but we also want to do h323 there. (we don’t authenticate any customer for user/pass, they only come with IP and prefix) they will send a call to us for destinations like India(91) and Egypt(20) so we have to create appropriate dialplan for them. for India(91) calls should go to indiavendor1 on ip address 122.252.236.10 as a 1st preference and once that is fill it sld go to indiavendor2 on IP address 122.252.236.11 calls for Egypt proper (20) and Egypt Mobile(201) should go to Egyptvendor1. Scenario 2 one SIP user created on freeswiotch which we can use to make test calls to the Gateways(vendors) we add to the switch for the testing purpose. that SIP user should be able to call any existing vendors/gateways for different destinations and also new one we will add in future so I think this one will have different dialplan. Scenario 3 1. where do we have to create SIP users which we can give it to retail customersfor them to cone in any ip, means they have to have user/pass and ipprovided by us to let them register in our system and once done they can make acall out NOTE: where it will dump CDR for all above calls ? Thanks Yogesh.

3 2

Re: [Freeswitch-users] freeswitch Scenario
by Yogesh Patel 17 Mar '08

17 Mar '08

#1 - Is your test server up and running right now? --> tets server is up and running but its on Private IP. #2 - Are you running with the default configs? --> we are running with default configured, configured wikipbx earlier with freeswitch but removed later when we found it can not help us with what we want. #3 - Have you made any changes or additions to the default configs? --> we have made few changes to make things working and did stuff here and there but nothing removed, and almost everything is same except added some more files. #4 - What revision are you on? --> we use SVN so I guess it will be the latest one. Let's get those nailed down first and then we'll take it from there. Yogesh

2 1

TEST TO -users LIST, IGNORE
by Michael Collins 17 Mar '08

17 Mar '08

1 0

Conceptual Question about Freeswitch and SIP
by Kurt Marasco 17 Mar '08

17 Mar '08

It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work? The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on. I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability. What am I missing....and I know that I am:) Kurt

6 7

404 Error on incoming calls
by Kurt Marasco 15 Mar '08

15 Mar '08

5 14

Overzealous 'nixevent'?
by freeswitch＠dalethatcher.com 14 Mar '08

14 Mar '08

I've been looking at improving the performance of an event socket app and have started trying to use 'nixevent' to restrict the traffic. However I've run into an issue where CUSTOM events go missing if I nix events. Environment: Freeswitch rev: 7886 Platform: Linux 2.6.22-14-generic #1 SMP (ubuntu) To reproduce I have this in my public.xml dialplan: <extension name="conference"> <condition field="destination_number" expression="^conference$"> <action application="answer"/> <action application="conference" data="conf"/> </condition> </extension> I set up socket A with the commands: auth ClueCon event plain all Socket B is set up with: auth ClueCon event plain all nixevent HEARTBEAT I then use a softphone to dial the 'conference' extension in the dialplan. On socket A I get the CUSTOM conference events, on socket B I don't get HEARTBEAT but I also get no CUSTOM conference events. I've also tried explicit enabling via: event plain CUSTOM conference::maintenence but still get no events (the spelling of maintenance is from the wiki). Can anyone else replicate or spot something I'm missing? thanks, - Dale Thatcher

3 2

Re: [Freeswitch-users] Incoming call audio problem
by Josip Djuricic 14 Mar '08

14 Mar '08

Ok, seems like I can't include any file larger then 100kb, so I can send the trace to people that want to check it out. As i said I still get no audio, but I'm rid of sip 415. From the trace you can see that there is a RTP from softphone trough FS, all the way to Cisco AS5300, but it doesn't get out on PSTN. Also FS get's no RTP from AS5300. This is all happening when I call from PSTN side into Voip network. When the call is coming from voip phone everything works well. Thank you all Djuricic wrote: > Hi there, > > I am including the new trace, this time with sip and rtp. > > I hope that somebody can see something I don't, cause there is still > no audio. > > This time I am doing the trace behind a NAT with a x-lite softphone > because I am at home. > > Josip Djuricic wrote: >> I meant GTD ;) >> >> Anyway, the traces now looks alot better, no SIP 415 anymore, but >> there is still no voice, and rtp seems to be going only from the >> phone to the FS >> >> Josip Djuricic wrote: >>> I can disable GDP but still no audio, I'm checkin the traces now >>> >>> Brian West wrote: >>>> Oh this bug has already been sent to MikeJ and Anthm. This is one >>>> strange invite. anyway its being worked on. You might want to see >>>> what in your config causes the addition of the application/gtd >>>> type. I suspect 'supplementary-service pass-through' >>>> >>>> /b >>>> >>>> >>>> INVITE sip:38514001148@89.18.32.19 >>>> <mailto:38514001148@89.18.32.19>:5080 SIP/2.0 Via: SIP/2.0/UDP >>>> 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: >>>> <sip:38513391900@10.0.0.19 >>>> <mailto:38513391900@10.0.0.19>>;tag=2D49486C-1855 To: >>>> <sip:38514001148@89.18.32.19 <mailto:38514001148@89.18.32.19>> >>>> Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: >>>> 253738A2-F04211DC-81719B71-D67790FB(a)10.0.0.19 >>>> <mailto:253738A2-F04211DC-81719B71-D67790FB@10.0.0.19> Supported: >>>> timer,100rel Min-SE: 1800 Cisco-Guid: >>>> 624255889-4030861788-2171509617-3598160123 User-Agent: >>>> Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, >>>> PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE >>>> Max-Forwards: 6 Remote-Party-ID: <sip:38513391900@10.0.0.19 >>>> <mailto:38513391900@10.0.0.19>>;party=calling;screen=yes;privacy=off >>>> Timestamp: 1205419179 Contact: <sip:38513391900@10.0.0.19 >>>> <mailto:38513391900@10.0.0.19>:5060> Expires: 180 Allow-Events: >>>> telephone-event MIME-Version: 1.0 Content-Type: >>>> multipart/mixed;boundary=uniqueBoundary Content-Length: 523 >>>> --uniqueBoundary Content-Type: application/sdp v=0 >>>> o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP >>>> Call c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 >>>> 10.0.0.19 a=rtpmap:8 PCMA/8000 a=ptime:20 --uniqueBoundary >>>> Content-Type: application/gtd Content-Disposition: >>>> signal;handling=optional IAM, PRN,isdn*,,NET5*, USI,rate,c,3,c,1 >>>> USI,lay1,alaw TMR,02 CPN,04,,1,14001148 CGN,04,,1,y,4,13391900 >>>> CPC,09 FCI,,,,,,,y, GCI,25356391f04211dc816e9b71d67790fb >>>> --uniqueBoundary-- >>>> On Mar 13, 2008, at 9:51 AM, Josip Djuricic wrote: >>>> >>>>> supplementary-service pass-through >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> 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 >>> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> 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 >> >

3 6

Re: [Freeswitch-users] 404 error
by Josip Djuricic 13 Mar '08

13 Mar '08

Dear Kurt, could you please send your dialplans, it seems like a routing problem. Thank you Josip

1 0

(no subject)
by Ben Levy 13 Mar '08

13 Mar '08

Is there any SS7 to sip functionality built into this project? Specifically I'm looking for a H324M to Sip gateway and don't want to use Asterisk. Thanks in advance for and answers you might have. Regards, Ben Levy President APEX Voice Communications, Inc THE APPLICATION SERVER COMPANY www.apexvoice.com Email: blevy(a)apexvoice.com Tel: +1.818.379.8400 x 103 Visit us at the following events... VON Mexico <http://www.von.com/2008/mexico/web/> Von Spring February 26 - 28, 2008 March 17 - 20, 2008 Mexico City, Mexico San Jose, CA, USA Booth 932 Booth 133 The information transmitted is intended only for the person(s) or entity to which it is addressed and may contain confidential and/or legally privileged material. Delivery of this message to any person other than the intended recipient(s) is not intended in any way to waive privilege or confidentiality. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by entities other than the intended recipient is prohibited. If you receive this in error, please contact the sender and delete the material from any computer.

2 1

Incoming call audio problem
by Josip Djuricic 13 Mar '08

13 Mar '08

Hi there, I was already writing about this problem, and I still didn't managed to fix it. I thought it was a NAT problem, but now without NAT the problem still remains. The error is like this, when I call from pstn, the call goes trough Cisco AS5300, then goes to FS, then to it's final destination - Ip Phone. Now you can answer the phone, but you can't hear anything on any side. And FS returns lots of SIP 415 (unsupported media type) to Cisco AS5300. When you call from Voip phone to the same pstn number it all works well, so I'm starting to be confused more and more. (and in this case there is no 415 messages). I believe cisco is sending something that FS can't understand, in the INVITE's but that's just a wild guess cause I ain't sure what can FS exactly understand very well. Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I can't send it to the list. Thank you all, Josip

5 14