April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Connecting HP 8500 All in One FAX Machine to FS
by Nice Voip 06 Apr '11

06 Apr '11

Hi All, I have got HP 8500 All in One machine, including FAX, now i want to use it to send faxes to my VoIP provider through FS, how do i connect it to FS? it has wireless card, but i don't see any VoIP settings. Many thanks. - NV

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FS and Phone are in the same IP
by dunchan 06 Apr '11

06 Apr '11

Hi! My voip phone doesn't get the SIP bye request from the called party (via PSTN gateway) Phone is in the same machine with FS, IP addr is same too, Contact header is filled corretly. If phone is in other machine everything works fine. Should I change the NAT handling somewhere, or what? thanks, Viktor

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Need to enable NDLB_force_port to work properly with Windows PC Internet Gateway
by Vitalii Colosov 06 Apr '11

06 Apr '11

Hi list, I am testing the following configuration: Sip ATA adapter (Linksys PAP2) -> Windows PC (Internet Gateway) -> ... Internet ... -> FreeSWITCH Sip adapter sends REGISTER to the FS, from port 5060, however Windows Gateway transfers this packet to FS from a different port (for example 62000). FS replies to the port 5060 (it looks like it takes it from the SIP text information). So Windows Gateway receives the answer to the port 5060, but since it expects it to be received in 62000, it looks like it drops the packets. The only way to fix this, is to set NDLB_force_port= true in the internal profile configuration file. In this case, FreeSWITCH replies exactly to the port 62000, and windows forwards it to the Sip adapter on port 5060. Do you know whether it is possible to configure Windows to route properly? Or maybe there is a way to configure the Sip adapter? (Like it will say to windows PC - hey, don't change the port, use the same as I am using... :)) Or it is a fantastic wish) Otherwise I will have to create 2 internal sip profiles, one with NDLB_force_port = true (for those who uses Windows PC as the gateway), and another with NDLB_force_port = false (for those who use normal routers). And I need 2 default dialplans, etc... so it complicates the solution. Thank you, Vitalie

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Re: [Freeswitch-users] FS and Phone are in the same IP
by Viktor 06 Apr '11

06 Apr '11

There is no nat, preveously i got a tipp to check this section. Phone is a softphone. I think the problem is the FS doesn't forward the bye message to softphone which is listen on same ip but different port. FS suppuse the bye message sent to him. How can i avoid that? Thanks, Viktor Sent from Samsung Mobile Meftah Tayeb <tayeb.meftah(a)gmail.com> wrote: >where's nat in this path? >you say phone and fs have same ip >is this a hardphone or a softphone? >if this is a hardphone, how come a pc and a phone can fill in the same >network with same ip? >thanks >On 06/04/2011 07:27, dunchan wrote: >> Hi! >> >> My voip phone doesn't get the SIP bye request from the called party (via >> PSTN gateway) >> >> Phone is in the same machine with FS, IP addr is same too, Contact >> header is filled corretly. >> >> If phone is in other machine everything works fine. >> >> Should I change the NAT handling somewhere, or what? >> >> thanks, >> Viktor >> >> _______________________________________________ >> 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 >> > > >-- >Meftah Tayeb >inum: +883510001288000 >phone: +13477595883 > >

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SIP Proxy and DNS SRV options
by mazilo 06 Apr '11

06 Apr '11

Under the SIP Settings and Proxy and Registration sections in Line 1 and Line 2 TABs of a Linksys WRTP54G VoIP router with firmware v3.1.27, I have noticed the following options (see the attached image file below): 1. SIP Proxy-Require option in the SIP Settings section and Proxy option in the Proxy and Registration section. 2. Use DNS SRV, DNS SRV Auto Prefix, and Proxy Redundancy Method options in the Proxy and Registration section. The question I have is what are the equivalent settings in FS for the above mentioned options. Any examples will be a plus. http://freeswitch-users.2379917.n2.nabble.com/file/n6245865/SIP_DNS_SRV.png ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-Proxy-and-DNS-SRV-options… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Gateway ReInvite Problem with a=inactive in SDP
by Lars Bobka 05 Apr '11

05 Apr '11

Hi, I have a problem with ReInvites and bypass-media option. A call over a gateway comes in. The gateway is a shared line from a broadsoft application server. The Invite over the gateway shows an a=inactive in the SDP. v=0 o=BroadWorks 66603944 1 IN IP4 xxx.xxx.xxx.xxx s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 21568 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:30 a=fmtp:101 0-15 a=bsoft: 1 image udptl t38 a=inactive After sending the 200OK over the gateway ang getting an ACK, the broadworks server sends a ReInvite, because of the shared line. Now the FS should answer the ReInvite with a 200Ok and sendrecv, but the FS sends inactive to the broadworks server and I have no audio in the call. Only when I hold hold the call and unhold the call, the inactive messages is not given and I have 2way audio. I try <param name="NDLB-sendrecv-in-session" value="true"/> everywhere (external.xml, internal.xml, user.xml), but I had no luck. So could you please help me. 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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simultaneous voice conference question
by Madovsky 05 Apr '11

05 Apr '11

When at least 3 persons are in conference I noticed that we can't hear the 3 voices in same time, only one voice at a time. is it normal ? thanks

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