- FreeSWITCH-users - FreeSWITCH Mailing Lists

Understanding Freeswitch
by Robert Clayton 13 Aug '08

13 Aug '08

In trying to learn the functionality of FreeSwitch does anyone have any suggestions for a hypothetical: To allow multiple users to call a single number and simultaneously and interactively leave recordings. I assume the line needs to bridge to other extensions which act as an ivr/recording connection. Initially it seems that Lua offers much functionality but would Lua answer the incoming line and branch to recording extensions in separate threads? Or is there other functionality (XML?) that should provide the branching and Lua answer the ivr/recording extensions? Does Lua even offer control over branching? Thanks in advance for the kick start. Bob

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Help with call forwarding
by Diego Viola 13 Aug '08

13 Aug '08

Hi everyone, I'm trying to do call forwarding in FS, this is what I have in my default dialplan: <extension name="10000"> <condition field="destination_number" expression="^10000$"> <action application="set" data="call_timeout=10"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER"/> <action application="bridge" data="sofia/internal/scot.clausing"/> <action application="bridge" data="sofia/gateway/teliax/blah"/> </condition> </extension> Whenever I dial 10000, I get a NO_ROUTE_DESTINATION error message because the UA is not registered. If I add NO_ROUTE_DESTINATION to the continue_on_fail line, it doesn't work too. I need it to pass to the next call even if the UA (scot.clausing) is not registered, or if there is no route to destination. Thanks, Diego

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Re: [Freeswitch-users] GUI
by Traun Leyden 13 Aug '08

13 Aug '08

> > > On Fri, Aug 1, 2008 at 8:02 AM, Lito Manansala > <lmanansala(a)tigercom.com.sg>wrote: > > > www.wikipbx.com > > > > Hi > > its asking Login, how can i download > > ram You can download from wikipbx.org The IVR's (for voicemail) are out of date due to recent mod_python changes, but the rest should work. I recently fixed a bug in the communication with freeswitch (event socket stuff).

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Installation: good, but some issues
by David Baggett 13 Aug '08

13 Aug '08

I've gotten FreeSWITCH running on this config: - Intel Atom D945GCLF in 1U rackmount - RedHat FC8 - Xorcom USB Astribank-8 (FXO) - Xorcom USB Astribank-8 (FXS) - Trunk FreeSWITCH code from a few days ago - zaptel-1.4.9.2.xpp.r5566 - Grandstream BT-100 SIP phones - Grandstream GXV-3000 SIP videophones - Various analog phone extensions First I tried installing on FC9. This didn't work because FC9 couldn't deal with the on-board NIC card on the D945GCLF. FC8 installed fine, though, so I just used that instead. Overall I am very impressed with FreeSWITCH. It is vastly easier to set up than Asterisk, and much cleaner. The XML config files are *so* much better than the ad hoc Asterisk config syntax. I like that I can make it run like a regular RedHat service, though I could not get it to run as user freeswitch -- it seems to want to run as root. I read through a bunch of the code and it is really well done. Thanks for a great open source project! Most things work: I set up the SIP phones and added the analog extensions, and can dial out on any phone over my POTS lines via Astribank FXO. I can receive incoming calls as well. I can videoconference between the videophones by using the code in the sample dialplan (default.xml) for "intercom" mode. Neat! Problems: Weirndess with FXS-originated calls ----------------------------------- If I call from an analog (FXS) extension to a BT-100, it rings normally. But if I then pick up the SIP phone, I can hear the ring on the analog extension get choppy. Soon after, the call is dropped. If I call from an analog (FXS) extension to a GXV-3000, it rings normally, but when I pick up the SIP phone, I get an immediate busy on the analog extension. If I call from an analog (FXS) extension to an X-Lite SoftPhone, it rings normally, but when I pick up the SIP phone, no audio comes through from the analog extension. If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone *to* an analog (FXS) extension, it works fine. The dialplan is simple (note that I have 1- and 2-digit extensions):  <extension name="OpenZAP extensions"> <condition field="destination_number" expression="^(9|1[0-6])$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="OpenZAP/${dialed_ext}/1"/> </condition> </extension>  <extension name="local-extension"> <condition field="destination_number" expression="^([3-6]\d)$"> <action application="set" data="dialed_ext=$1"/> <action application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action> <action application="export" data="sip_invite_params=intercom=true"/> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_ext}@$${domain}"/> </condition> </extension> In general, I found it a bit strange that you can't (it seems) put analog extensions into the directory. Is it true that you have to define analog extensions manually in the dialplan? That's kind of counterintuitive, especially for Asterisk users; the Asterisk Now GUI makes analog and SIP extensions look the same for all practical purposes. Echo ---- I was getting massive distortion on analog connections involving any GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf. The default value was 64. I changed it to 0 and the distortion went away, but of course I get lots of annoying echo now. Other SIP phones don't seem to have this problem. I have read in earlier posts to this list that the GXV-3000 has known problems. But the interesting thing is that it was working fine with Asterisk and the Astribanks (though on a different CPU & M/B). The Astribank seems to have its own EC, but I can't figure out how to turn it on. And I have no idea what to set the value in zt.conf to; 32 seems to work better than either 16 or 64 (!). I tried fxotune but that didn't do anything: it ran for a long time and generated an /etc/fxotone that seemed to have all zero values. Basically, it would be great to have an "echo cancellation HOWTO" for OpenZap users since EC seems to be offered in several different layers. I also notice a bunch of different EC modes in the OpenZap code -- does anyone know how to try different ones out? FAX detection ------------- This just doesn't seem to work. Here's the relevant code from my diaplan. (It looks a little odd because I send all unanswered calls from my POTS lines to the ext. 31 voicemail box.)   <extension name="incoming-astribank"> <condition field="destination_number" expression="^([1-4])$"> <action application="answer" /> <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" /> <action application="set" data="dialed_ext=31"/> <action application="export" data="dialed_ext=31"/> <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::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <action application="set" data="call_timeout=20"/> <action application="sleep" data="1000"/> <action application="bridge" data="user/30@$${domain},user/31@$${domain},user/33@$${domain},user/36@$${domain},user/37@$${domain},OpenZAP/9/1,OpenZAP/10/1,OpenZAP/11/1"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension>  <extension name="fax"> <condition field="destination_number" expression="^fax$"> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="OpenZAP/12/1"/> </condition> </extension> Thanks for any help with these issues. Dave

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setting contact header
by Craig Guy 13 Aug '08

13 Aug '08

Hi, First, have been lurking for about a week and I feel Freeswitch is a great project, coming from a non-programmer background and heavy familiarity with Asterisk and Callweaver it's a bit of a learning curve to come to grips with the XML but I'm slowly getting there J I'm currently doing some interop testing with Freeswitch and the provider has come back to me stating that they require the Contact header to exhibit: a) e.164 format b) The From, RPID and Contact headers to match Try as I might I am seemingly unable to set the user portion of the Contact header for the outbound leg to the termination provider. The contact is always mod_sofia@<ip:port>. I have found the sip-force-contact setting however it seems to have no effect on the contact header (and I might be using it incorrectly in any case). The ideal behaviour for me is for the contact to take the form of callerid@<ip:port> For example if my sip profile is bound to 192.168.0.1:5060 and my handset callerid is +13035555555 I would like to set the contact to sip:+13035555555@192.168.0.1:5060 Is there an ability within Freeswitch to set the Contact header based on callerid? Craig

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Very high cpu utilization under load test of FS
by Erol Akarsu 13 Aug '08

13 Aug '08

I am trying a load on FS. I am making originate call between two FS boxes. Originator endpoint is playing one minute sound file with raw format, the other endpoint is sleeping during this time. I am using Xeon quad X5355, 2.66 Ghz box as originator, the other remote endpoint is hp laptop. The system can handle 500 simultaneous originate calls, successfully, without any NO_ANSWER event. But when I start more than 600 originate calls, the Xeon box is having high cpu utilization (almost 100%) and system almost freezes and getting a lot of no-answer calls. I thought this is due to RTP streams and I made Xeon to sleep instead of playing sound file. But I got same behavior, that means it did not help. I am thinking either there is some thread code block that eats CPU or something else. Could you please guide me so that I can achieve 1000 simultaneous originate calls? I appreciate your help, Best Regards Erol Akarsu

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Re: [Freeswitch-users] SIP/2.0 407 Proxy Authentication Required
by Shehzad Pankhawala 13 Aug '08

13 Aug '08

Hi all, I have solved the above problem of routing call from SIP client (Yate) to Freeswitch: All the configuration was same as my previous request, but instead of creating a local user and gateway, i only created one gateway in directory/default folder with all the settings like (used <include> <user id="4444"> <gateways> <gateway name="yate"> <param name="username" value="4444"/> <param name="realm" value="192.168.1.23"/> <param name="password" value="''your-password''"/> <param name="proxy" value="192.168.1.23"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </gateways> <params> <param name="password" value="1234"/> </params> </user> </include> Enjoy; Shehzad.

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SDP issue receiving calls from SIP connection
by Kirk Bateman 13 Aug '08

13 Aug '08

Afternoon everyone, I have a bit of a problem with Freeswitch receiving RTP INVITEs from my SIP provider (tesco's internet phone ... I know SIP isn't supported technically, but it sort of works...) Freeswitch (sofia) seems to be whinging about the INVITE SDP format ... but I'm not sure why ... here's a dump of my log / trace, anyone got any clues, I've tried adding extra codecs to my list and setting the late negotiation but that doesn't seem to fix it ... (registration all works fine by the way ...) ------------------------------------------------------------------------ INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER(a)sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER(a)sip.tescointernetphone.com> ;user=phone> Max-Forwards: 69 Content-Type: application/sdp Contact: <sip:ANOTHER_PHONE_NUMBER@77.75.1.10:5060;transport=udp> Allow: INVITE,CANCEL,ACK,BYE Accept: application/sdp Content-Length: 840 v=0 o=- 3250588766 0 IN IP4 77.75.1.10 s=- c=IN IP4 77.75.1.10 t=0 0 m=audio 54424 RTP/AVP 18 99 101 102 103 15 104 4 105 106 107 108 125 109 100 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G.729a/8000 a=rtpmap:101 G.726-16/8000 a=rtpmap:102 G.726-24/8000 a=rtpmap:103 G.726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:104 G.723.1-H/8000 a=rtpmap:4 G723/8000 a=rtpmap:105 G.723.1-L/8000 a=rtpmap:106 G.729b/8000 a=rtpmap:107 G.723.1a-H/8000 a=rtpmap:108 G.723.1a-L/8000 a=rtpmap:125 G.nX64/8000 a=rtpmap:109 AMR/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=maxptime:20 a=maxptime:30 a=ptime:20 a=ptime:30 a=X-cap: 1 audio RTP/AVP 100 a=X-cap: 2 image udptl t38 a=X-sqn:0 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 ------------------------------------------------------------------------ tport_deliver(0x80fcbf0): msg 0x8129468 (1416 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100) nta: canonizing sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060 with contact nta: INVITE (100) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x80ef908, 0x80eccb8, 0x8120fe0) called soa_set_params(static::0x8111d08, ...) called nta_leg_tcreate(0x81200c0) soa_init_offer_answer(static::0x8111d08) called soa_set_remote_sdp(static::0x8111d08, (nil), 0x810ed20, 840) called nua(0x8120fe0): INVITE server: error parsing SDP nua: nua_invite_server_respond: entering tport_tsend(0x80fcbf0) tpn = UDP/77.75.1.10:5060 tport_resolve addrinfo = 77.75.1.10:5060 tport_by_addrinfo(0x80fcbf0): not found by name UDP/77.75.1.10:5060 tport_vsend returned 661 send 661 bytes to udp/[77.75.1.10]:5060 at 13:16:28.358779: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER(a)sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER(a)sip.tescointernetphone.com> ;user=phone>;tag=yjay36ZrycNmD Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9235 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk Content-Length: 0 ------------------------------------------------------------------------ nta: sent 400 Bad Session Description for INVITE (100) nta_leg_destroy(0x81200c0) soa_destroy(static::0x8111d08) called tport_wakeup_pri(0x80fcbf0): events IN tport_recv_event(0x80fcbf0) tport_recv_iovec(0x80fcbf0) msg 0x8111d08 from (udp/192.168.1.10:5060) has 442 bytes, veclen = 1 recv 442 bytes from udp/[77.75.1.10]:5060 at 13:16:28.377628: ------------------------------------------------------------------------ ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 CSeq: 100 ACK Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 From: "ANOTHER_PHONE_NUMBER" < sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER(a)sip.tescointernetphone.com> ;user=phone>;tag=SDvke8201-b2b.31deb49 To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER@sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER(a)sip.tescointernetphone.com> ;user=phone>;tag=yjay36ZrycNmD Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x80fcbf0): msg 0x8111d08 (442 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100) nta: ACK (100) is going to INVITE (100) nta: timer set next to 4820 ms nta: timer I fired, terminate 400 response incoming_reclaim_all((nil), (nil), 0xb174a1fc) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set sofia loglevel 0 API CALL [sofia(loglevel 0)] output: Sofia-sip log level set to [0] Any clues would be welcomed ... Cheers Kirk Bateman

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presence states
by Jonas Gauffin 13 Aug '08

13 Aug '08

Is it possible to get FS to just report open/closed (avail/not avail) instead of all specific call states? Maybe a toggle for that or something?

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Call broadcasting
by Ruchir Brahmbhatt 13 Aug '08

13 Aug '08

Hi, Which is the best method of doing call broadcasting in freeswitch? In asterisk we can do this by creating call files and asterisk handles the rest. How does freeswitch offers this feature? Do we need to use mod_event_socket only? Is it faster enough for this requirement?

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