- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
by Mitul Limbani 26 Aug '09

26 Aug '09

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; }Infact coming from Asterisk world is not too hard to figure out FS, but yeah i wont recommend a newbie that route :) Although an Asterisk dude /dudette can check out the Rosetta_Stone page on wiki Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com On Wed 26/08/09 19:35 , Bradley Brashier bjbrashier(a)gmail.com sent: Speaking as someone who went through this recently myself, my suggestion is to learn by doing. Choose some simple-sounding task you want to accomplish in FS, and try to carry it out. Use the wiki as a reference and the mailing list if you can't figure out what you need from that. I don't suggest you start with Asterisk or anything else, as I'd guess that would just confuse the issue. I came from a BIOS background, went directly into Freeswitch, and am so far glad I did. BB On Wed, Aug 26, 2009 at 7:25 AM, Merle J. Ebbert wrote: > > Hi, > > I'm trying to avoid taking up a lot of peoples valuable time. > > SIP & FS have brought some ideas for some commercial products but I > need to know where to start. > > Having once written a proprietary DOS & helped with writing a RTOS, I > consider myself capable > of learning. I (we) just need to know where to start to come up to > speed rapidly. > > Is there a FreeSWITCH tutorial available? > Should someone new start with Asterisk and then possibly move to FS? > > Thanks, > Merle > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org [2] > 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 [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------- Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ Links: ------ [1] mailto:se02005-mec@yahoo.com [2] mailto:FreeSWITCH-users@lists.freeswitch.org [3] mailto:FreeSWITCH-users@lists.freeswitch.org

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Re: [Freeswitch-users] I would like to send an IM. From a Windows PC. To a BlackBerry @vzw.blackberry.com
by Merle J. Ebbert 26 Aug '09

26 Aug '09

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Questions about att_xfer (freeswitch version 1.0 trunk 14633M)
by Anatoliy Kounitskiy 26 Aug '09

26 Aug '09

Hello everybody! I have few questions about the att_xfer application. First, what i want to accomplish is: user A calls user B, after that user B makes attended transfer to user C. In the dialplan i have: <context name="vpbx"> <extension name="local_number"> ... <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::attented_xfer XML features"/> .... </condition> </extension> So when user B answers the call, he sends *4 and the extensions for the attended transfer is started - the usual - plays message and read the input dtmf: features.xml ... <extension name="attented_xfer"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^attented_xfer$"> <action application="info"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="call_timeout=15"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> ... To this problems everything is perfect. But here comes the questions, so if you can give some tips would be great. 1) when user B enters the extension number of C - the C's phone starts ringing in the tcpdump i can see that the phone is sending 180 ringing, BUT user B does not hear the ringing. 2) as mentioned in the http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer quote: "If the other leg is a voicemail or doesn't answered you can hangup that leg by pressing dtmf # (fixed in r14438) " It doesn't work. The option 0 is working even before C answering the phone - after he answers it's a threeway conference :) - i like this feature. I'm using FreeSWITCH Version 1.0.trunk (14633M) Also I tried to set call timeout to see if I can go back the user A, who is listening to MOH - no luck here. Probably I'm missing something. Tried to look in the source of att_xfer to understand why the feature i want is not working - but it seems my C/C++ skills are not so good, as i want :( . Thank you in advance, Anatoliy Kounitskiy

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FreeSWITCH 1.0.4 windows installer - great but I have a little problem
by Raffaele P. Guidi 26 Aug '09

26 Aug '09

I had the sweet surprise to find the installer packaged with FreePBX... really great! Why it has not been advertised as it deserves? It worked like a breeze once launched, with the automatic configuration and all of that., Only thing: once stopped I cannot get it to load sofia profiles anymore - issueing sofia status doesn't show anything. I had to copy internal.xml and default.xml from a previous installation and everything got to work again - but no FreePBX anymore :( I'm sure I'm missing something important. Can you give me a hint? Should sofia profiles be served by curl or something? Thanks, Raffaele

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Moved Freeswitch to new subnet.
by Mike Peace 26 Aug '09

26 Aug '09

I just installed FS on a fresh Centos 5.3 install, everything went perfect. I loaded x0lite up on a couple of hosts and they registered and worked fine. Then I moved the Centos server to a new subnet on out network that we have tons of boxes on and have all the routes already configured on and now the Sip phones will no longer register. No other changes have been made on the FreeSwitch server other than changing the IP. BTW I can ping to and from the FS server and the hosts and the firewall is off on the FS server and the hosts. I plugged a laptop with X-lite into a switch on the same network so the packets don't even need to route and it still won't register! BTW I also downloaded 3CX VoIP phone and it behaves the same way. Any ideas? TIA =-=-=-=-=-=-=-=-=-=-=-=-=-=-= EDCO Group, Inc. Phone: (800) 999-3456 Fax: (800) 999-3551 Web: http://www.edcogroupinc.com/ Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

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Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message
by Tihomir Culjaga 25 Aug '09

25 Aug '09

Hello Brian, Dave Still nothing... i've changed ip_addresses (remote_ip, local_ip) and changed branch within ACK message to meet INVITE's one....but it is still not enough... Also i checked RFC and this is how should it be ... (ACK without contact taking care to have correct TAGs and branch)... what can it be? ------------------------------------------------------------------------ INVITE sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 Max-Forwards: 70 Contact: <sip:22222238515000403@10.4.4.252<sip%3A22222238515000403(a)10.4.4.252> > From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7079(a)10.4.4.252 CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 131 v=0 o=user1 53655765 2353687637 IN IP4 10.4.4.252 s=- c=IN IP4 10.4.4.252 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ------------------------------------------------------------------------ send 325 bytes to udp/[10.4.4.252]:5060 at 21:56:08.152812: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7079(a)10.4.4.252 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0 ------------------------------------------------------------------------ send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.159929: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060 >;tag=cFS6jHj9DgjjF Call-ID: 1-7079(a)10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 342 bytes from udp/[10.4.4.252]:5060 at 21:56:08.160166: ------------------------------------------------------------------------ ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@110.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060 >;tag=cFS6jHj9DgjjF Call-ID: 1-7079(a)10.4.4.252 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.661299: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060 >;tag=cFS6jHj9DgjjF Call-ID: 1-7079(a)10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 here is a scenario i use: <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC"> <send retrans="500"> <![CDATA[ INVITE sip:[service]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip];branch=[branch] Max-Forwards: 70 Contact: <sip:[field1]@[local_ip]> From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="302" rtd="true"> </recv> <send> <![CDATA[ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 ]]> </send>  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> </scenario> ---------- Forwarded message ---------- From: Brian West <brian(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Date: Mon, 24 Aug 2009 15:42:31 -0500 Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 Use that.. your scenario has some hard coded IP's in the fields that shouldn't be there. /b On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote: Hello Brian, it doesn't work .. tried this today as well:

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RTP Packet Routing
by Jerry Richards 25 Aug '09

25 Aug '09

Hello All, I noticed Freeswitch becomes the "middle-man", handling RTP traffic for an active call. How do I configure it so it allows the two SIP endpoints to send RTP packet to each other directly? Best Regards, Jerry

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Scalability
by Jerry Richards 25 Aug '09

25 Aug '09

Hello All, Does anyone know what the capacity of a stand-alone Freeswitch, in terms of how many users? Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Best Regards, Jerry

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reload user data
by Juan Backson 25 Aug '09

25 Aug '09

Hello, I would like to dynamically add user to freeswitch. If I add a new file to the directory dir, is there anyway to have freeswitch to read the new user xml file without having to restart freeswitch? Other than using flat file, is there anyway to add user to freeswitch user api command? Thanks, JB

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Freeswitch & DAHDI Kernel drivers / Hardware questions
by Raimund Sacherer 25 Aug '09

25 Aug '09

Hello, i was reading through the openzap wiki page and searched the wiki, the only thing I found about dahdi is a note from January that freeswitch does not work with dahdi right now. Is this current information? Can't I use dahdi kernel drivers for freeswitch? As the last update to the zaptel drivers is now nearly a year old (1.4.12) and development has, AFAIK, ceased and new stuff only gehts to dahdi, correct me please if I am wrong. Second question, is it possible to use this hardware: 00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 10 I/O ports at e000 [disabled] [size=8] Memory at ee001000 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 Kernel modules: hisax with openZap? I read in the wiki (openZap.conf_Examples) something about HFC 4 BRI, but i do not know if this is the same as the above mentioned card. I have access to some pcmcia fxo, and pcmcia isdn cards, could they be used? I have as well an soekris board with two pcmcia slots, could i use this with an fxo pcmcia card to actually drive a small home - pbx with freeswitch? I did not find answers for this questions in the wiki :-) best regards -- Raimund Sacherer - RunSolutions Open Source It Consulting - Email: rs(a)runsolutions.com tel: 625 40 32 08 Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares

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