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moh is not transcoded for some CODECs
by Bzzz 01 Apr '12

01 Apr '12

Hi list, I'm playing with the FS original diaplan and noticed something strange, when twinkle has G711(a|u) or GSM into its list the moh's played, but when I only leave iLBC & SPEEX it fails. I modified vars.xml and added iLCB@30i on both global_codec_prefs & outbound_codec_prefs, so AFAI understand how FS works it should be Ok. But the log shows comparisons only against G7221s, PCMU, PCMA & GSM: 2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[G7221:115:32000:20:48000] 2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[G7221:107:16000:20:32000] 2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[G722:9:8000:20:64000] 2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[PCMU:0:8000:20:64000] 2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[PCMA:8:8000:20:64000] 2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/ [GSM:3:8000:20:13200] does MOH only work with these codecs, or did I miss another place than vars.xml to set the MOH available CODECs? JY -- "Surely you can't be serious." "I am serious, and stop calling me Shirley." -- "Airplane"

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Re: [Freeswitch-users] VM sound very choppy
by Peter Olsson 01 Apr '12

01 Apr '12

Another case of virtual machine setup maybe, or is this real hardware? /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Bzzz Skickat: den 23 mars 2012 00:44 Till: freeswitch-users(a)lists.freeswitch.org Ämne: Re: [Freeswitch-users] VM sound very choppy On Thu, 22 Mar 2012 10:21:58 -0700 Michael Collins <msc(a)freeswitch.org> wrote: That the same thing when I put a call on hold: http://pastebin.freeswitch.org/18724 -- For some reason, this fortune reminds everyone of Marvin Zelkowitz. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:4f6bb86f32761437815508!

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Api function to check if media is currently being played
by Michael Lutz 01 Apr '12

01 Apr '12

Hi All, I am using an remote host to control my parked call using an inbound socket connection. I am sending "uuid_broadcast" to play an audio file to the call. Before it is being bridged. Everything works ok. The problem is I need to continue my code (ESL) to instruct the bridge etc, but I need to do that when the audio file being played is finished. Is there an api function for this? I tried "uuid_buglist" to check if that would return me some useful data, but that doesn't seem to be working (at least for what I need ..) Is the something like "uuid_mediaReady" ? show api didn't show me something similar.. Thanks!, Mike. Ps. I believe tricking with checking file length of al wavs and calculating the duration upfront would not be the way doing this. ;-)

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Skypopen problem on centos5.8
by tangdu 31 Mar '12

31 Mar '12

First, Giovanni, Thank You for the new skypopen installer。 With mod skypopen wiki，I'm trying to install skypopen on my centOS server witout success. when run /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh， error message： ERROR: Module snd_pcm_oss does not exist in /proc/modules ERROR: Module snd_mixer_oss does not exist in /proc/modules ERROR: Module snd_seq_oss does not exist in /proc/modules mknod: `/dev/dsp': File exists insmod: error inserting '/usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko': -1 File exists how could i solved this problem？ Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skypopen-problem-on-centos5-8… Sent from the freeswitch-users mailing list archive at Nabble.com.

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When using nimbuzz, call disconnects after 90 seconds with RECOVERY_ON_TIMER_EXPIRE
by Anton VG 31 Mar '12

31 Mar '12

Please help, While other SIP clients works well, there is a problem with nimbuzz SIP client, which quite frequently used by 'not-advanced' users. seems It which works through their services (asterisk 1.6) This problem was not happening while I have been using Asterisk too, but after recent migration to FS - it does. nimbuzz client stays like online, just sound disappears, but there is a confirmation from nimbuzz server side, regarding the BYE most likely a NAT issue, but if the same time using Asterisk on the same server - there is not problems. So might be there any options to tune? there are some SIP activity like this before calls disconnects, no replies: send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: ------------------------------------------------------------------------ INVITE sip:anton@192.168.0.20:5061;transport=udp SIP/2.0 Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe Max-Forwards: 70 From: "3777077" <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS To: "anton(a)62.122.137.32" <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 26214721 INVITE Contact: <sip:3777077@62.122.137.32:5060;transport=udp> User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 272 v=0 o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 s=FreeSWITCH c=IN IP4 62.122.137.32 t=0 0 m=audio 20062 RTP/AVP 18 106 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:20 and than 2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup sofia/internal/anton(a)62.122.137.32 [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------

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Trouble setting "caller_id_name", "caller_id_number" on an inbound call
by Brian Foster 31 Mar '12

31 Mar '12

As the subject says, I'm having issues setting the caller id name and number on an inbound call when doing some cleanups and a lookup. These dialplans are based on what information is listed for Mod_callcenter, and I have a feeling that they are wrong. I'll volunteer myself to change them if they are. Alright, so here's the dialplan: <extension name="cid_number_cleanup" continue="true"> <condition field="caller_id_number" expression="^(?:\+)(\d+)$"> <action application="set" data="caller_id_number=$1" inline="true"/> </condition> </extension> <extension name="cid_name_cleanup" continue="true"> <condition field="caller_id_name" expression="^(?:\+)(\d+)$"> <action application="set" data="caller_id_name=$1" inline="true"/> </condition> </extension> <extension name="cid_lookup-country_code_1" continue="true"> <condition field="${module_exists(mod_cidlookup)}" expression="true"/> <condition field="caller_id_name" expression="^${caller_id_number}$|^$"/> <condition field="caller_id_number" expression="^(?:/+1)?([2-9]\d\d[2-9]\d{6})$"> <action application="cidlookup" data="$1"/> </condition> </extension> Call log here: http://pastebin.freeswitch.org/18786 I've tried setting the effective_caller_id_(name/number) but no luck there. Is there something I'm doing wrong here? -BDF -- Brian D. Foster Endigo Computer LLC Email: bdfoster(a)endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

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Re: [Freeswitch-users] Setting Hangup Hook on api originate
by Michael Collins 30 Mar '12

30 Mar '12

On Fri, Mar 30, 2012 at 9:37 AM, Bob McCarthy <bob.mccarthy(a)experient.com>wrote: > I am having trouble getting a hangup hook to execute after an api > originate.**** > > **** > > bgapi expand originate > {api_hangup_hook=confhanguphook.lua,session_in_hangup_hook=true}sofia/internal/2001@${switchvox} > &conference(25bce61d-fb32-44d7-8d99-7ceafc599f22@sla)**** > > ** ** > > I get into the conference just fine but the script does not execute when I > hang up. Any Idea's of what I am doing wrong? > I have an idea: how about actually calling 'lua' and not just the name of your script? ;) bgapi expand originate {api_hangup_hook='lua confhanguphook.lua',session_in_hangup_hook=true}sofia/internal/2001@${switchvox} &conference(25bce61d-fb32-44d7-8d99-7ceafc599f22@sla) -MC

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Could not locate channel type skinny
by Luis Daniel Lucio Quiroz 30 Mar '12

30 Mar '12

Helo, Strange thing. I'm mapping a DID to a extension. Using a softphone incomming call is working using a hwrdware atcome phone it fails with this log: EXECUTE sofia/external/18886498985(a)voxbone.com bridge(skinny/internal/1100) 2012-03-30 14:47:36.320320 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-03-30 14:47:36.320320 [ERR] switch_core_session.c:427 Could not locate channel type skinny 2012-03-30 14:47:36.320320 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [skinny] cause: [CHAN_NOT_IMPLEMENTED] 2012-03-30 14:47:36.320320 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] 2012-03-30 14:47:36.320320 [INFO] mod_dptools.c:2916 Originate Failed. Cause: CHAN_NOT_IMPLEMENTED 2012-03-30 14:47:36.320320 [DEBUG] switch_channel.c:2848 (sofia/external/18886498985(a)voxbone.com) Callstate Change RINGING -> HANGUP Is it a missing thing ? Thanks LD

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Specifying Port on Record-Route header
by Thomas McCarthy-Howe 30 Mar '12

30 Mar '12

Hi all - I'm trying to configure my FS box to run on the same server as an OpenSIPs instance, and I think I'm having a problem because, when FS adds a Record-Route header to the 200 OK, it does not specify the port. So, my endpoint is sending the ACK back to OpenSIPs, not to FS, and it keeps looping on me. The 200 OK from FS running on port 5070: 15:21:29.330683 IP 192.168.153.13.vtsas > 192.168.153.13.sip: SIP, length: 1044 E..0.N..@. ........SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.153.13;branch=z9hG4bK36e1.59a34dc3.0 Via: SIP/2.0/UDP 192.168.153.28;branch=z9hG4bKebe2bd1a967399c3d Record-Route: <sip:192.168.153.13;lr=on> From: <sip:D05@192.168.153.13>;tag=8739e6ff4f To: <sip:MEETME01@192.168.153.13>;tag=F5yXe7FeFH6jg Call-ID: c0bd06789074500d CSeq: 498777857 INVITE Contact: <sip:MEETME01@192.168.153.13:5070;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-62c6855 2012-02-27 13-28-27 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 217 Remote-Party-ID: "MEETME01" <sip:MEETME01@192.168.153.13>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1333103289 1333103290 IN IP4 192.168.153.13 s=FreeSWITCH c=IN IP4 192.168.153.13 t=0 0 m=audio 17600 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:10 m=audio 0 RTP/AVP 19 The ACK comes back to 5060, and loops: 15:21:29.338255 IP 192.168.153.13.sip > 192.168.153.13.sip: SIP, length: 599 ....._..ACK sip:192.168.153.13;lr=on SIP/2.0 Via: SIP/2.0/UDP 192.168.153.13;branch=z9hG4bK36e1.59a34dc3.2 Via: SIP/2.0/UDP 192.168.153.28;branch=z9hG4bK01e5c3b7e9a67e0dd Proxy-Authorization: Digest username="D05",realm="192.168.153.13",nonce="4f75cf9700005326420497c8c462fb80c60a857e0d0049ee",uri="sip:MEETME01@192.168.153.13",response="961b9d5a496f610d10fd136fecf0291c" Max-Forwards: 69 From: <sip:D05@192.168.153.13>;tag=8739e6ff4f To: <sip:MEETME01@192.168.153.13>;tag=F5yXe7FeFH6jg Call-ID: c0bd06789074500d CSeq: 498777857 ACK User-Agent: ATC ION M5T SIP Stack/4.1.8.13 Content-Length: 0 Any thoughts? Thanks in advance. Thomas -- Thomas McCarthy-Howe Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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Re: [Freeswitch-users] Setting CallerID for inbound SLA Barge in
by Bote Man 30 Mar '12

30 Mar '12

That originates in the device itself and is contained in the caller_id_name and caller_id_number variables. I just tried the effective_ series of variables, but those are for the outbound B-leg calls and are empty at the time the conference is entered. But if a phone is barging into a conference how does Caller*ID have any impact, other than to test for it in the dialplan? Do you intend to say: the number or name? Bote From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bob McCarthy Sent: Monday, 19 March, 2012 00:37 To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Setting CallerID for inbound SLA Barge in I have read where you can set the outbound Callerid name and number for calling out of a conference, but how do you set the Callerid name and number for phones that barge in on a conference (SLA in particular) ?

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