- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to limit the max number of registration users
by fieldpeak 07 Apr '11

07 Apr '11

Could anyone help advise how to limit the max number of registration users? thanks if any advise. Regards, Charles

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Recording transfer audio
by Eric Beard 07 Apr '11

07 Apr '11

Hello, I asked this question yesterday over irc. My apologies if someone answered there already, my irc client kept crashing so I might have missed it. I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, except when the call gets transferred. When I do a transfer, I see two WAV files, one of which is the original session in which I can hear the call up until the transfer. The other file is all silence, which I assume is the session up to the point where the INVITE is answered. But I get no audio of the conversation after the transfer. My suspicion is that freeSwitch is not hairpinning the audio, so it does not have access to the RTP. I pasted everything from the console at http://pastebin.freeswitch.org/16027. Thanks! ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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Freeswitchsolutions.com?
by Alex Crow 07 Apr '11

07 Apr '11

Hi list, Anthony, We are investigating a possible move from a proprietary phone system to Freeswitch. Advertised on the FS site is the above address which appears to say it provides commercial support for FS. In the last two weeks both me and a colleague filled in the contact form asking about prices, SLAs etc but have had no response. Is the company real or is there a problem with the contact form? Thanks Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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What FreeSwitch version i am using after updating git pull?
by Shamun toha md 07 Apr '11

07 Apr '11

How do we know if i am using the latest Git version (nightly builids)? How to know if i am really using 1.0.7 or 1.0.6 oldest? So far tried as > git pull && make current > make current (showed some error) > make clean modwipe (completed successfully) > ./configure && make & make install > /usr/local/freeswitch/bin/freeswitch > > preesed F12 shows: > FreeSWITCH Version 1.0.head (git-828960a 2010-09-25 12-51-42 -0500) Let me know plz, is this the 1.0.7 ?

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Cannot compile freetdm
by Valery K 07 Apr '11

07 Apr '11

Enter: # cd /usr/local/freeswitch/libs/freetdm # ./configure --with-libisdn # make ... compile ... cc1: warnings being treated as errors src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34': src/ftmod/ftmod_isdn/ftmod_isdn.c:982: warning: unused variable 'cplen' src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'isdn_configure_span': src/ftmod/ftmod_isdn/ftmod_isdn.c:2794: warning: passing argument 2 of 'Q931SetLogCB' from incompatible pointer type make: *** [ftmod_isdn_la-ftmod_isdn.lo] Error 1 Why? libisdn-0.0.1 installed -- View this message in context: http://old.nabble.com/Cannot-compile-freetdm-tp31340769p31340769.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Hylafax server emulation
by Steve Underwood 07 Apr '11

07 Apr '11

Hi, It has always been clear that a HylaFAX compatible FAX job submission server would add considerably to the value of the FAX facilities in Asterisk and Freeswitch, but somehow it hasn't happened until now. I recently found that in 2005 someone produced something fairly basic for Asterisk in Perl, but it doesn't seem to have been well publicised, and it looks like development stalled long ago. I now have the skeleton of HylaFAX compatible FAX job submission server, in C, working. It accepts FAX submissions from sendfax and a couple of the windows HylaFAX clients, though it needs a lot more polishing. Now I need to look at the best thing to do on the Freeswitch side. I aim to make the server maintain its own database of FAX jobs. It will attach to Freeswitch, by ESL; push the jobs through FS; deal with scheduling, retries, etc; and report the final result to the user, just as HylaFAX does. The thing I am rather unsure about is the best way to handle the accounts used to accept FAX jobs? Should I maintain a separate database of FAX accounts, or hook into an existing database? I would welcome suggestions for what would be the most useful approach. Steve

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softphone to outbound help
by Dave Bracken 06 Apr '11

06 Apr '11

Can anyone tell me what i need to do to be able to dial a 7 digit number on my softphone and have it dial out to a phone i have sitting on my desk? I guess what i really need to know is what all files do i need to set up and where, etc. I have never done anything like this, so i need your help. Thanks in advance, Dave

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Correct way to determine bridge result?
by Dmitry Sytchev 06 Apr '11

06 Apr '11

Hi all! How to correctly determine bridge result (answer/no answer and reason) after executing Bridge on already answered channel? endpoint_disposition shows 'ANSWER' as call is answered. Call goes to FS, FS answers, reads digits and attempts bridge to some destination. But I can't reliable determine whether this call was answered or rejected with some status, as originate_disposition is set to SUCCESS even if we just get early media, and endpoint_disposition contains 'ANSWER' related to current channel. -- Best regards, Dmitry Sytchev, IT Engineer

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Re: [Freeswitch-users] uuid_hold is not send hold message
by 张明 06 Apr '11

06 Apr '11

Michael Collins，您好！　　use step, freeswitch version 1.07 1. fs_cli 2. /event plain CHANNEL_HOLD CHANNEL_UNHOLD 3. show calls freeswitch@10.108.226.220@internal> show calls call_uuid,call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid,hostname b0a7bdfd-430d-4746-91a4-8c318f9e1695,2011-04-08 04:22:37,1302207757,switch_ivr_multi_threaded_bridge,700000,700000,700001,sofia/internal/700000@10.108.226.220,b0a7bdfd-430d-4746-91a4-8c318f9e1695,Outbound Call,700001,700001,sofia/internal/sip:700001@10.108.226.44:18692,e2faae72-ef43-4019-b7e6-1120d7e42e1e,localhost.localdomain 1 total. 4. uuid_hold b0a7bdfd-430d-4746-91a4-8c318f9e1695 2011-04-08 04:27:55.539115 [DEBUG] switch_channel.c:1373 (sofia/internal/700000(a)10.108.226.220) Callstate Change ACTIVE -> HELD 2011-04-08 04:27:55.539115 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/700000(a)10.108.226.220 [BREAK] 2011-04-08 04:27:55.539115 [DEBUG] switch_ivr.c:1272 接收消息完毕 2011-04-08 04:27:55.539115 [DEBUG] switch_core_session.c:954 Send signal sofia/internal/sip:700001@10.108.226.44:18692 [BREAK] 2011-04-08 04:27:55.539115 [DEBUG] sofia.c:4646 Channel sofia/internal/700000(a)10.108.226.220 entering state [calling][0] 2011-04-08 04:27:55.544824 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:700001@10.108.226.44:18692 [BREAK] 2011-04-08 04:27:55.544824 [INFO] sofia.c:729 sofia/internal/700000(a)10.108.226.220 Update Callee ID to "Outbound Call" <700000> 2011-04-08 04:27:55.547789 [DEBUG] sofia.c:4646 Channel sofia/internal/700000(a)10.108.226.220 entering state [ready][200] 2011-04-08 04:27:55.547789 [DEBUG] sofia.c:4654 Duplicate SDP v=0 o=Idefisk_user 47685 6056184806875839522 IN IP4 10.108.226.44 s=Idefisk_user c=IN IP4 10.108.226.44 t=0 0 m=audio 8000 RTP/AVP 8 97 0 110 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:110 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:2690 Already using PCMA 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4571 Set 2833 dtmf send/recv payload to 101 2011-04-08 04:27:55.725181 [DEBUG] switch_ivr.c:563 sofia/internal/sip:700001@10.108.226.44:18692 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/sip:700001@10.108.226.44:18692 playback(local_stream://moh) 2011-04-08 04:27:55.725181 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2011-04-08 04:27:55.725181 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16@8000hz 1 channels 20ms 5. it 's not return hold message, if user Zoiper hold button freeswitch return hold message Zoiper hold freeswich return message: 2011-04-08 04:34:51.330157 [DEBUG] sofia.c:4646 Channel sofia/internal/700000(a)10.108.226.220 entering state [received][100] RECV EVENT Event-Name: CHANNEL_HOLD Core-UUID: 5e94c1c9-305d-4c4d-b00c-ea24f6346559 FreeSWITCH-Hostname: localhost.localdomain FreeSWITCH-IPv4: 10.108.226.220 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2011-04-08 04:34:51 Event-Date-GMT: Thu, 07 Apr 2011 20:34:51 GMT Event-Date-Timestamp: 1302208491330157 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_mark_hold Event-Calling-Line-Number: 642 Channel-State: CS_EXECUTE Channel-Call-State: HELD Channel-State-Number: 4 Channel-Name: sofia/internal/700000(a)10.108.226.220 Unique-ID: eb1222c9-4175-469a-8549-667c0dd921a6 Call-Direction: inbound Presence-Call-Direction: inbound Channel-Presence-ID: 700000(a)10.108.226.220 Channel-Call-UUID: eb1222c9-4175-469a-8549-667c0dd921a6 Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Read-Codec-Bit-Rate: 64000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Channel-Write-Codec-Bit-Rate: 64000 Caller-Direction: inbound Caller-Username: 700000 Caller-Dialplan: XML Caller-Caller-ID-Name: 700000 Caller-Caller-ID-Number: 700000 Caller-Callee-ID-Name: Outbound Call Caller-Callee-ID-Number: 700001 Caller-Network-Addr: 10.108.226.44 Caller-ANI: 700000 Caller-Destination-Number: 700001 Caller-Unique-ID: eb1222c9-4175-469a-8549-667c0dd921a6 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/internal/700000(a)10.108.226.220 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1302208456152936 Caller-Channel-Created-Time: 1302208456152936 Caller-Channel-Answered-Time: 1302208463040930 Caller-Channel-Progress-Time: 1302208456340619 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Other-Type: originatee Other-Leg-Direction: outbound Other-Leg-Username: 700000 Other-Leg-Dialplan: XML Other-Leg-Caller-ID-Name: 700000 Other-Leg-Caller-ID-Number: 700000 Other-Leg-Callee-ID-Name: Outbound Call Other-Leg-Callee-ID-Number: 700001 Other-Leg-Network-Addr: 10.108.226.44 Other-Leg-ANI: 700000 Other-Leg-Destination-Number: 700001 Other-Leg-Unique-ID: 1595a55f-d97c-47c1-aa6c-42c8efcb5b27 Other-Leg-Source: mod_sofia Other-Leg-Context: default Other-Leg-Channel-Name: sofia/internal/sip:700001@10.108.226.44:18692 Other-Leg-Screen-Bit: true Other-Leg-Privacy-Hide-Name: false Other-Leg-Privacy-Hide-Number: false 2011-04-08 04:34:51.330157 [DEBUG] sofia.c:4657 Remote SDP: v=0 o=Idefisk_user 6056184806875821816 6056184806875816806 IN IP4 10.108.226.44 s=Idefisk_user c=IN IP4 0.0.0.0 t=0 0 m=audio 8000 RTP/AVP 8 97 0 110 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:110 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly 2011-04-08 04:34:51.330157 [DEBUG] switch_channel.c:1373 (sofia/internal/700000(a)10.108.226.220) Callstate Change ACTIVE -> HELD 2011-04-08 04:34:51.330157 [DEBUG] switch_core_session.c:954 Send signal sofia/internal/sip:700001@10.108.226.44:18692 [BREAK] 2011-04-08 04:34:51.346542 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:700001@10.108.226.44:18692 [BREAK] 2011-04-08 04:34:51.524112 [DEBUG] switch_ivr.c:563 sofia/internal/sip:700001@10.108.226.44:18692 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/sip:700001@10.108.226.44:18692 playback(local_stream://moh) 2011-04-08 04:34:51.524112 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz 2011-04-08 04:34:51.524112 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16@8000hz 1 channels 20ms 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2690 Already using PCMA 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4571 Set 2833 dtmf send/recv payload to 101 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2976 Audio params changed for sofia/internal/700000(a)10.108.226.220 from 10.108.226.44:8000 to 0.0.0.0:8000 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2987 AUDIO RTP [sofia/internal/700000(a)10.108.226.220] 10.108.226.220 port 19022 -> 0.0.0.0 port 8000 codec: 8 ms: 20 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:3017 AUDIO RTP CHANGING DEST TO: [0.0.0.0:8000] 2011-04-08 04:34:51.580870 [DEBUG] sofia.c:5057 Processing updated SDP 2011-04-08 04:34:51.583974 [DEBUG] sofia.c:4646 Channel sofia/internal/700000(a)10.108.226.220 entering state [completed][200] 2011-04-08 04:34:51.583974 [DEBUG] sofia.c:4646 Channel sofia/internal/700000(a)10.108.226.220 entering state [ready][200] uuid_hold command user problem: 1. uuid_hold uuid is error still return OK 2. uuid_hold <uuid> OR uuid_hold off <uuid> freeswitch is not return hold or unhold message ======== 2011-04-05 03:23:09 您在来信中写道： ======== I just tried this on latest git and it worked fine for me. Can you pastebin the console debug output when you use it? -MC 2011/4/2 张明 <sc_zhangming(a)sina.com> freeswitch-users，您好！ uuid_hold command , freeswitch is not send HOLD message. who know it. 　　　　　　　　致礼！　　　　　　　　张明　　　　　　　　sc_zhangming(a)sina.com 　　　　　　　　　　2011-04-02 _______________________________________________ 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 = = = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　　　　　　　张明　　　　　　　　　　　　　　sc_zhangming(a)sina.com 　　　　　　　　　　　　　　　2011-04-06

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Gateway ReInvite Problem with a=inactive in
by Lars Bobka 06 Apr '11

06 Apr '11

ok, I understand, that the sdp is media and so untouched by the freeswitch. But isn´t there a possibility to change this behaviour, for example in the dialplan, or somewhere else. It could also be an automatic hold/unhold, after answering the call, that the call becomes active. Is something like this possible? regards Lars ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

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