- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Scheduled Hangup Issue
by Anthony Minessale 16 Apr '08

16 Apr '08

I added support in the latest trunk to set the variable execute_on_answer to an application of your choice <action application="set" data="execute_on_answer=sched_hangup +20 alloted_timeout"/> -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

9 29

Asterisk vs. Freeswitch
by Pete Kay 16 Apr '08

16 Apr '08

Hi, This question may have come up a few times already. I am working on a application to provide IVR, voicemail, and tailored call routing services. The SIP registration will be handled by Openser, and Asterisk is only doing the media function. We are talking about over 100 users. Is Freeswitch better than Asterisk in terms of functionality, ease-of-maintain, and ease-of-use? Thanks alot in advance for your inputs. Pete

7 10

What is Freeswitch equivalent of Asterisk AMI and Realtime
by Pete Kay 16 Apr '08

16 Apr '08

Hi, I have studied the Freeswitch doc and can't find any info related to setting up the sip users, queue, ans voicemail as a realtime DB, like Asterisk does. Is this feature available? If not, would it be a bit too much work to write to the XML all the times? Also, is there an Asterisk AMI-equivalent feature in Freeswitch? What is that called? Thanks for your input. Pete

3 4

Re: [Freeswitch-users] Asterisk vs. Freeswitch - what about functionality
by Pete Kay 16 Apr '08

16 Apr '08

Hi, I have been seeing quite a few articles that mention Freeswitch is much more efficient than Asterisk. What about in terms of functionality? Can Freeswitch perform the same kind of functionality provided by Asterisk such as Queue, Conf call, MOH, Voicemail Main, etc? Can anyone give some inputs on this regards? Thanks, Pete

2 1

javascript originate timeout problem
by Jonas Gauffin 16 Apr '08

16 Apr '08

Hello again =) I've made a call queue in javascript where the js takes orders from a server through a socket. The server monitors all extensions (through eventsocket) in the queue and tells the javascript to originate the call to a extension as soon as it becomes idle. Everything works fine except when the extension do not answer. Freeswitch seems to kill the javascript and executes the dialplan instead. This only happens when the destination do not answer. Small log (from when entering the javascript and forward) 2008-04-16 15:21:01 [CONSOLE] mod_spidermonkey_socket.c:109 socket_connect() Connecting to: 127.0.0.1:23341. 2008-04-16 15:21:01 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/default/anonymous(a)anonymous.invalid [BREAK] 2008-04-16 15:21:01 [NOTICE] mod_spidermonkey.c:1948 session_answer() Channel [sofia/default/anonymous(a)anonymous.invalid] has been answered 2008-04-16 15:21:01 [DEBUG] mod_spidermonkey.c:3132 js_api_use() Loading Socket 2008-04-16 15:21:01 [DEBUG] log.js:7 console_log() DEBUG: Sending register 2008-04-16 15:21:01 [DEBUG] sofia.c:1648 sofia_handle_sip_i_state() Channel sofia/default/anonymous(a)anonymous.invalid entering state [completed] 2008-04-16 15:21:01 [DEBUG] sofia.c:1648 sofia_handle_sip_i_state() Channel sofia/default/anonymous(a)anonymous.invalid entering state [ready] 2008-04-16 15:21:02 [DEBUG] log.js:7 console_log() DEBUG: Receivied: transfer 20 192.168.1.110 u1000022 2008-04-16 15:21:02 [DEBUG] log.js:7 console_log() DEBUG: Calling sofia/192.168.1.110/u1000022 2008-04-16 15:21:02 [DEBUG] log.js:7 console_log() DEBUG: DESTINATION: {ignore_early_media=true,originate_timeout=15}sofia/192.168.1.110/u1000022 2008-04-16 15:21:02 [DEBUG] switch_ivr_originate.c:607 switch_ivr_originate() variable string 0 = [ignore_early_media=true] 2008-04-16 15:21:02 [DEBUG] switch_ivr_originate.c:607 switch_ivr_originate() variable string 1 = [originate_timeout=15] 2008-04-16 15:21:02 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/u1000022 [91daaf25-4f6f-8848-8d10-596f45c13238] 2008-04-16 15:21:02 [DEBUG] mod_sofia.c:1830 sofia_outgoing_channel() sofia/default/u1000022 State Change CS_NEW -> CS_INIT 2008-04-16 15:21:02 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/u1000022 [BREAK] 2008-04-16 15:21:02 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/u1000022 Running State Change CS_INIT 2008-04-16 15:21:02 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() (sofia/default/u1000022) State INIT 2008-04-16 15:21:02 [DEBUG] mod_sofia.c:80 sofia_on_init() sofia/default/u1000022 SOFIA INIT 2008-04-16 15:21:02 [DEBUG] mod_sofia.c:107 sofia_on_init() sofia/default/u1000022 State Change CS_INIT -> CS_RING 2008-04-16 15:21:02 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/u1000022 [BREAK] 2008-04-16 15:21:02 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() (sofia/default/u1000022) State INIT going to sleep 2008-04-16 15:21:02 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/u1000022 Running State Change CS_RING 2008-04-16 15:21:02 [DEBUG] sofia.c:222 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/default/u1000022 2008-04-16 15:21:02 [DEBUG] sofia.c:1648 sofia_handle_sip_i_state() Channel sofia/default/u1000022 entering state [calling] 2008-04-16 15:21:02 [DEBUG] switch_ivr_originate.c:951 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2008-04-16 15:21:02 [DEBUG] switch_core_state_machine.c:418 switch_core_session_run() (sofia/default/u1000022) State RING 2008-04-16 15:21:02 [DEBUG] mod_sofia.c:119 sofia_on_ring() sofia/default/u1000022 SOFIA RING 2008-04-16 15:21:02 [DEBUG] switch_ivr_originate.c:55 originate_on_ring() sofia/default/u1000022 State Change CS_RING -> CS_HOLD 2008-04-16 15:21:02 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/u1000022 [BREAK] 2008-04-16 15:21:02 [DEBUG] switch_core_state_machine.c:418 switch_core_session_run() (sofia/default/u1000022) State RING going to sleep 2008-04-16 15:21:02 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/u1000022 Running State Change CS_HOLD 2008-04-16 15:21:02 [DEBUG] switch_core_state_machine.c:440 switch_core_session_run() (sofia/default/u1000022) State HOLD 2008-04-16 15:21:02 [DEBUG] sofia.c:222 sofia_event_callback() event [nua_r_invite] status [180][Ringing] session: sofia/default/u1000022 2008-04-16 15:21:02 [DEBUG] sofia.c:222 sofia_event_callback() event [nua_i_state] status [180][Ringing] session: sofia/default/u1000022 2008-04-16 15:21:02 [DEBUG] sofia.c:1648 sofia_handle_sip_i_state() Channel sofia/default/u1000022 entering state [proceeding] 2008-04-16 15:21:02 [NOTICE] sofia.c:1686 sofia_handle_sip_i_state() Ring-Ready sofia/default/u1000022! 2008-04-16 15:21:02 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/default/anonymous(a)anonymous.invalid [BREAK] 2008-04-16 15:21:02 [DEBUG] sofia.c:350 sofia_event_callback() nua_r_prack: unknown event 39: 200 OK 2008-04-16 15:21:33 [DEBUG] sofia.c:1648 sofia_handle_sip_i_state() Channel sofia/default/u1000022 entering state [calling] 2008-04-16 15:21:33 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1202(a)192.168.1.110 [e59265ba-8486-7546-af60-8da7b25e87 2008-04-16 15:21:33 [DEBUG] sofia.c:1648 sofia_handle_sip_i_state() Channel sofia/default/1202(a)192.168.1.110 entering state [received] 2008-04-16 15:21:33 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/1202(a)192.168.1.110 Running State Change CS_NEW 2008-04-16 15:21:33 [DEBUG] sofia.c:1652 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 4934107701406015977 4314135494661389300 IN IP4 192.168.1.110 s=FreeSWITCH c=IN IP4 192.168.1.110 t=0 0 a=sendrecv m=audio 25866 RTP/AVP 0 8 9 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2008-04-16 15:21:33 [DEBUG] sofia_glue.c:2011 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2008-04-16 15:21:33 [DEBUG] sofia_glue.c:1333 sofia_glue_tech_set_codec() Set Codec sofia/default/1202(a)192.168.1.110 PCMU/8000 20 ms 160 samples 2008-04-16 15:21:33 [DEBUG] switch_core_state_machine.c:368 switch_core_session_run() (sofia/default/1202(a)192.168.1.110) State NEW 2008-04-16 15:21:33 [DEBUG] sofia_glue.c:1975 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2008-04-16 15:21:33 [DEBUG] sofia_glue.c:1983 sofia_glue_negotiate_sdp() Set comfort noise payload to 13 2008-04-16 15:21:33 [DEBUG] sofia.c:1789 sofia_handle_sip_i_state() sofia/default/1202(a)192.168.1.110 State Change CS_NEW -> CS_INIT 2008-04-16 15:21:33 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/1202(a)192.168.1.110 [BREAK] 2008-04-16 15:21:33 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/1202(a)192.168.1.110 Running State Change CS_INIT 2008-04-16 15:21:33 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() (sofia/default/1202(a)192.168.1.110) State INIT 2008-04-16 15:21:33 [DEBUG] mod_sofia.c:80 sofia_on_init() sofia/default/1202(a)192.168.1.110 SOFIA INIT 2008-04-16 15:21:33 [DEBUG] mod_sofia.c:107 sofia_on_init() sofia/default/1202(a)192.168.1.110 State Change CS_INIT -> CS_RING 2008-04-16 15:21:33 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/1202(a)192.168.1.110 [BREAK] 2008-04-16 15:21:33 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() (sofia/default/1202(a)192.168.1.110) State INIT going to sleep 2008-04-16 15:21:33 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/1202(a)192.168.1.110 Running State Change CS_RING 2008-04-16 15:21:33 [DEBUG] switch_core_state_machine.c:418 switch_core_session_run() (sofia/default/1202(a)192.168.1.110) State RING 2008-04-16 15:21:33 [DEBUG] mod_sofia.c:119 sofia_on_ring() sofia/default/1202(a)192.168.1.110 SOFIA RING 2008-04-16 15:21:33 [DEBUG] switch_core_state_machine.c:65 switch_core_standard_on_ring() Standard RING sofia/default/1202(a)192.168.1.110 2008-04-16 15:21:33 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Arne Rubin->82378009@public 2008-04-16 15:21:35 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test conditions destination_number(82378009) =~ /^\+4623666(12[01][0-9])$/ 2008-04-16 15:21:35 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch 2008-04-16 15:21:35 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test conditions destination_number(82378009) =~ /^(5551212)$/ 2008-04-16 15:21:35 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch 2008-04-16 15:21:35 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test conditions destination_number(82378009) =~ /1202/ 2008-04-16 15:21:35 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch 2008-04-16 15:21:35 [INFO] switch_core_state_machine.c:113 switch_core_standard_on_ring() No Route, Aborting

2 3

ivr script, originating a call
by Jonas Gauffin 16 Apr '08

16 Apr '08

Hello I'm launching a javascript that originates a call to a user and then records a file. The problem is that the soundfiles ("record after the beep" and "message saved") sounds strange (choppy/vibrating, yeah I know, sucky explanation). The same phrases sounds fine when played for an incoming call. Is it a bitrate issue or something like that? Regards, Jonas

7 14

is it support RFC 3428 feature?
by 이성훈 16 Apr '08

16 Apr '08

Hi freeSwitch users ! I was installed freeSwitch window version on my notebook, all functionality features are doing desirable. but RFC 3428(instant message) feature is not. see below, i describe my setting. UAS - freeSwitch window version, in my notebook (ip:192.168.1.100) UAC1 - just simple voip phone, in my notebook (ip:192.168.1.100) UAC2 - hardware voip phone (ip:192.168.1.101) both UACs are the same setting that sip server (freeSwitch UAS). UAC1 <-> UAC2 call, take answer, talk are doing good. but UAC1 <-> UAC2 instant message send, receive are not. UAC1 sends message to UAC2, UAC1 receives 200 ok message from UAS, but UAC2 receives nothing, reverse actions too. anyone help me please ! is it support RFC 3428 feature in freeSwitch ? (denotes: UAC - user agent client, in this case voip phone UAS - user agent server, in this case sip server(freeSwitch)) (example sip packets) ------------------------------------------------------------------------ recv 791 bytes from udp/[localaddress]:5090 at 01:57:15.187500: ------------------------------------------------------------------------ MESSAGE sip:1001@localaddress SIP/2.0 Via: SIP/2.0/UDP localaddress:5090;rport;branch=z9hG4bKPj4f841d9b229c4353957a63f44b097214 Max-Forwards: 70 From: <sip:1000@localaddress>;tag=7c5eab80dedb4d7c8fc2cc5b44a53b70 To: <sip:1001@localaddress> Call-ID: 97e135971d86481ca15dae3acc0272d0 CSeq: 26500 MESSAGE Accept: text/plain, application/im-iscomposing+xml Contact: <sip:1000@localaddress:5090;transport=UDP> User-Agent: PJSUA v0.8.0/win32 Content-Type: application/im-iscomposing+xml Content-Length: 264 <?xml version="1.0" encoding="UTF-8"?> <isComposing xmlns="urn:ietf:params:xml:ns:im-iscomposing" xmlns:xsi=" http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:im-composing iscomposing.xsd"> <state>active</state> </isComposing> ------------------------------------------------------------------------ send 522 bytes to udp/[localaddress]:5090 at 01:57:15.187500: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP localaddress:5090;rport=5090;branch=z9hG4bKPj4f841d9b229c4353957a63f44b097214 From: <sip:1000@localaddress>;tag=7c5eab80dedb4d7c8fc2cc5b44a53b70 To: <sip:1001@localaddress>;tag=m461UjHUm02yN Call-ID: 97e135971d86481ca15dae3acc0272d0 CSeq: 26500 MESSAGE User-Agent: FreeSWITCH-mod_sofia/1.0.rc1-7946 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Content-Length: 0 ------------------------------------------------------------------------ recv 509 bytes from udp/[localaddress]:5090 at 01:57:17.953125: ------------------------------------------------------------------------ MESSAGE sip:1001@localaddress SIP/2.0 Via: SIP/2.0/UDP localaddress:5090;rport;branch=z9hG4bKPj770d2b3e6a5248b5bf0aca1abc976b9c Max-Forwards: 70 From: <sip:1000@localaddress>;tag=8eb10cec2ced420fbee0a3f1c004a82c To: <sip:1001@localaddress> Call-ID: 9a795b147818437d8b0cb4e11c9e6e6a CSeq: 15724 MESSAGE Accept: text/plain, application/im-iscomposing+xml Contact: <sip:1000@localaddress:5090;transport=UDP> User-Agent: PJSUA v0.8.0/win32 Content-Type: text/plain Content-Length: 2 HI ------------------------------------------------------------------------ send 522 bytes to udp/[localaddress]:5090 at 01:57:17.953125: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP localaddress:5090;rport=5090;branch=z9hG4bKPj770d2b3e6a528b5bf0aca1abc976b9c From: <sip:1000@localaddress>;tag=8eb10cec2ced420fbee0a3f1c004a82c To: <sip:1001@localaddress>;tag=ND0tXD2yH9rHH Call-ID: 9a795b147818437d8b0cb4e11c9e6e6a CSeq: 15724 MESSAGE User-Agent: FreeSWITCH-mod_sofia/1.0.rc1-7946 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Content-Length: 0 ------------------------------------------------------------------------ send 655 bytes to udp/[221.132.67.173]:5060 at 01:57:17.968750: ------------------------------------------------------------------------ MESSAGE sip:1001@221.132.67.173:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP localaddress;rport;branch=z9hG4bKFQ85jBK4v24mg Max-Forwards: 70 From: <sip:1000@localaddress>;tag=8eb10cec2ced420fbee0a3f1c004a82c To: "UAS2" <sip:1001@221.132.67.173:5060;transport=udp> Call-ID: 48d60e4c-85fb-122b-2780-39a48cb53b8d CSeq: 98037182 MESSAGE Contact: <sip:mod_sofia@localaddress:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.rc1-7946 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Content-Type: text/html Content-Length: 2 HI ------------------------------------------------------------------------ recv 377 bytes from udp/[221.132.67.173]:5060 at 01:57:18.015625: ------------------------------------------------------------------------ SIP/2.0 415 Unsupported Media Type Call-ID: 48d60e4c-85fb-122b-2780-39a48cb53b8d CSeq: 98037182 MESSAGE From: <sip:1000@localaddress>;tag=8eb10cec2ced420fbee0a3f1c004a82c To: "UAS2" <sip:1001@221.132.67.173:5060>;tag=205e2a1a5f55968 Via: SIP/2.0/UDP localaddress;branch=z9hG4bKFQ85jBK4v24mg Content-Length: 0 User-Agent: Stonehenge IP255-S 1.30.191

4 5

Startup pointers
by David Villasmil 16 Apr '08

16 Apr '08

Hello all, I just downloaded FS from svn, compiled, everything went smooth. (all default) I can actually register 2 UAs, 1000 and 1001. I can call the conference, i can call 888, etc. BUT, I can NOT call the UAs! The question: How does FS actually work? I mean, how does SIP UAs area mapped to contexts, diaplans, extensions, etc? Something else, I do NOT want to use radius, so, how can I configure MYSQL for cdr? mod_cdr is out of the tree... any hints? Thanks to all. David

2 2

Re: [Freeswitch-users] [OpenSER-Users] FreeSWITCH and OpenSER
by Mikael A. Bjerkeland 16 Apr '08

16 Apr '08

ons, 16.04.2008 kl. 11.35 +0300, skrev Juha Heinanen: > Mikael A. Bjerkeland writes: > > > All SIP traffic goes to the OpenSER proxy which in turn routes the call > > with the LCR module to a FreeSWITCH server. This has proven to work fine > > in my limited testing environment. I am using ENUM on the FreeSWITCH > > server to look up dialed numbers. If the dialed number was found in our > > ENUM registry and belongs to the proxy, we call back to OpenSER: > > why don't you do enum lookup already in openser? Sure, I can do this. Should the NAPTR record point to my FreeSWITCH server or OpenSER? If it points to OpenSER then I could just use dbaliases, right? Should the call be looked up with ENUM in OpenSER and then routed via FreeSWITCH? If so, how is this done? > > > number_to_transfer_to@openser-registrar. The call is routed to OpenSER > > which complains that it cannot find the subscriber/alias, which makes > > sense as there isn't configured an alias. But this also happens when I > > transfer to a PSTN number. > > hard to say without more info, but if the number is not found in openser > registrar, should you then use lcr module to route it to one of your > gws? Yes, I've been trying this, but couldn't get it to work. I will try once more, unless you have another suggestion to my questions above. > -- juha Thanks, Mikael

1 0

FreeSWITCH and OpenSER
by Mikael A. Bjerkeland 16 Apr '08

16 Apr '08

Hi, I am trying to look up information on the use of FreeSWITCH together with OpenSER. I am doing a configuration on my own without much experience on the configuration of OpenSER. In my setup OpensSER acts as a registrar and proxy. I have multiple FreeSWITCH servers for media, routing and PSTN termination, but I am not sure if my approach is the "right way". All SIP traffic goes to the OpenSER proxy which in turn routes the call with the LCR module to a FreeSWITCH server. This has proven to work fine in my limited testing environment. I am using ENUM on the FreeSWITCH server to look up dialed numbers. If the dialed number was found in our ENUM registry and belongs to the proxy, we call back to OpenSER: SIP UA (subscriber 56195122) -> INVITE 25929511 -> FreeSWITCH -> ENUM lookup -> Found -> Routed back to OpenSER -> SIP UA (subscriber (25929511) If the call is not found: SIP UA (subscriber 56195122) -> INVITE 95018615 -> FreeSWITCH -> ENUM lookup -> Not found -> Routed to PSTN gateway or SIP peer I am not using aliases in OpenSER as it seems that ENUM provides the same functionality, but please correct me if I'm wrong. I also made this decision because it seems easier to implement features like call forwarding (unconditional, busy, no answer and timed) in FreeSWITCH rather than in OpenSER. This brings me back to my current issue: Call transfer Outbound and inbound calling with ENUM lookups in FreeSWITCH to my test phones works fine. So does calls to the PSTN gateway. My problem has to do with transfers. I can do a an attended transfer from my Polycom phone when I'm in a call. The second call is set up and I can talk to the person I want to transfer to. I hit the transfer button. Nothing happens, and FS and OpenSER try to look up number_to_transfer_to@openser-registrar. The call is routed to OpenSER which complains that it cannot find the subscriber/alias, which makes sense as there isn't configured an alias. But this also happens when I transfer to a PSTN number. Do I need to take special actions on REFERs? Are there any examples of a working OpenSER + FreeSWITCH setup out there? I believe FreeSWITCH and OpenSER complement each other great but I found the lack of information on the two applications used together a showstopper. Has anyone implemented phone service based on OpenSER and FreeSWITCH who can explain their setup a bit? Regards, Mikael

1 0