January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

PHP ESL timeout?
by Tim St. Pierre 11 Jan '13

11 Jan '13

Hello, I currently have a testing system that opens an ESL connection to freeswitch to launch test calls, and also to make sure the switch is actually running as it should be. If the switch is up, it works. If the switch is totally down, it works. If the switch is partially up, it doesn't work. If the switch is up, but not really responsive, I am able to create a connection, but when I issue an API call, it sits there forever and doesn't return anything. If the API execute timed out and the script continued to execute, it would check the response, see that it is not what it should be, and alert the right people. Anyone know this library that can help? -Tim

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mod_conference auto-record issue
by Usama Zaidi 11 Jan '13

11 Jan '13

Hi, Following in my conference profile: <profile name="intPreConf" description="InterPre Phone Conference">  <param name="auto-record" value="/root/home/test.wav"/> <param name="rate" value="8000"/> <param name="member-flags" value="dist-dtmf"/> <param name="energy-level" value="300"/> <param name="caller-controls" value=""/> </profile> I'm having issues with auto-record, I keep getting these messages on the console: 2013-01-11 06:19:27.082662 [ERR] mod_conference.c:3884 Error Opening File [/root/home/test.wav] 2013-01-11 06:19:27.082662 [ERR] switch_core_timer.c:117 Timer is not properly configured. And if I keep it running for a while, FS segfaults. I was trying recording to a stream initially, but then to narrow down the issue I tried recording to a file and got the same error. Please advise. Regards. -- I'd love to change the world, but they wont gimme the source code to it

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SIP packet exceeds MTU?
by Tim St. Pierre 10 Jan '13

10 Jan '13

Hello, I'm having a problem with BLF in some cases, and I found that the length of the username was a factor in whether the lamps worked or not. When I did some packet captures, I found that problem packets were 1500 bytes or so. Since the endpoint is on a DSL line that has an MTU of 1460, the packet doesn't fit. I'm already using compressed headers. Is there a way to reduce the packet size more, or is there another solution? I will include an example below: Note that my IP address and domain substutions may change the packet size a little, but the point is the same. Thanks! NOTIFY sip:XXXX@YYY.YYY.YYY.YYY:5060;transport=udp SIP/2.0 v:SIP/2.0/UDP zzz.zzz.zzz.zzz:5070;rport;branch=z9hG4bKD5QUr5m2Fm3Dp Route:<sip:aaa.aaa.aaa.aaa:61906>;transport=udp Max-Forwards:70 f:<sip:accountcode-701@ourdomain.com>;tag=TrmHE4FBIrB4 t:"Tim St. Pierre"<sip:XXXX@ourdomain.com>;tag=ecc2498bfd i:3decd12decccafe1 CSeq:448376002 NOTIFY m:<sip:accountcode-701@xxx.xxx.xxx.xxx:5070;transport=udp;fs_nat=yes;fs_path=sip:yyy.yyy.yyy.yyy:61906%3Btransport%3Dudp> User-Agent:Communicate Freely 2.0 Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE k:timer,precondition,path,replaces o:dialog u:talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-session-description,presence.winfo,message-summary,refer Subscription-State:active;expires=1137 c:application/dialog-info+xml l:639 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="477" state="partial" entity="sip:accountcode-701@ourdomain.com"> <dialog id="accountcode-701" direction="initiator"> <state>confirmed</state> <local> <identity display="park">sip:accountcode-701@ourdomain.com;proto=park</identity> <target uri="sip:accountcode-701@ourdomain.com;proto=park"> <param pname="+sip.rendering" pvalue="no"/> </target> </local> <remote> <identity display="park">sip:accountcode-701</identity> <target

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Channel variables for npdi and rn on LRN dipped calls.
by Victor Chukalovskiy 10 Jan '13

10 Jan '13

Hello and Happy New year! A SIP call comes in that was previously dipped. I'd like to be able to have "rn" and "npdi" available as channel variables that I can use in regex expressions. Is there a way to configure FreeSWITCH to put "rn" and "npdi" into separate channel variable? So far, info application shows me the following channel variables that contain LRN: Caller-Destination-Number: [5551234567;npdi=yes;rn=5555550001] variable_sip_req_user: [5551234567;npdi=yes;rn=5555550001] variable_sip_req_uri: [5551234567;npdi=yes;rn=5555550001@1.2.3.4:5060] I can probably dissect those and put "rn" and "npdi" into separate variables. But this means extra steps in dial-plan. In case there is a built-in way to accomplish the same task, it would be great! Could you please let me know. Thank you, Victor

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Re: [Freeswitch-users] freeswitch continously sending sip update-messages: why?
by Michel Brabants 10 Jan '13

10 Jan '13

Hello, I'm testing freeswitch as an sbc (version 1.2.5.3). I think I discovered a bug, but I still have to verify it against git if I can find the time to compile it. Anyway, it all relates to session-timer according to me, but even if I disable session-timers, freeswitch still sends sip-update-messages to the sip-client. Freeswitch is dropping the calls after 30 seconds because of this. There is no problem with natting, firewalling, ... as far as I can tell as I'm tracing the packets on all ends. A packet from freeswitch to the sip-client (phone behind other B2B-server): UPDATE sip:10000@lab2.ddm.priv SIP/2.0 Via: SIP/2.0/UDP 172.16.105.14;rport;branch= z9hG4bKS2pQKQ7SrNgce Max-Forwards: 70 From: <sip:8888@172.16.105.17:5060>;tag=X5FKtKj9ZHaDc To: <sip:10000@lab2.ddm.priv >;tag=849E1A5B-6750-4EDE-B7CF-72DF979FD126-40026 Call-ID: 5AC29923-74EE-46B2-830A-9CB78B9DB47F-39541(a)172.16.105.17 CSeq: 38492344 UPDATE Contact: <sip:8888@172.16.105.14:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121206T050429Z~91eef34d5c Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 249 P-Asserted-Identity: "Outbound Call" <0001> v=0 o=FreeSWITCH 1357636566 1357636567 IN IP4 172.16.105.14 s=FreeSWITCH c=IN IP4 172.16.105.14 t=0 0 m=audio 20250 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 The other end of the sbc/client doesn't support session-timers or update (yate client), and it doesn't have this problem. So, the main question is: --> Why is freeswitch sending these update-messages while the codec, ... remains the same as far as I can tell. It isn't a session-timer, because it is enabled in the profile and it doesn't appear in the message too (which means the disable had some influence ...)? So, is there a way to: 1) fix the above? 2) force session-timers to use reinvite instead of update. As this case may be a way around the bug (freeswitch doing as it is the refresher while they agreed that the client is the session-timer-refresher). 3) (asked before, but not answered) Is there a way to disable sip-"update"-message-support in freeswitch? Thank you and kind regards, Michel

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mod_directory menu-top?
by Phillip Warner 10 Jan '13

10 Jan '13

Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again? For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr Thanks. Phil.

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mod_com_g729 transcoding
by Colin Mason 10 Jan '13

10 Jan '13

I have phone A connected to freeswitch box 1 and phone B connected to freeswitch box 2. Phone A wants to dial phone B using G729. Codec is always G729. The path RTP follows is: Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B (g729) (g729) (g729) My question is, why is it that the freeswitch box receiving the call uses up an encoder/decoder when the codec is G729 along the path? If I reverse the call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license. if I dial the PSTN to a carrier who supports G729, I don't use up a license. Any thoughts? Maybe this is normal. FreeSWITCH Box 1: 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP [sofia/mpls/I888_1(a)172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6 port 16466 codec: 18 ms: 20 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 codec: 18 ms: 20 freeswitch@internal> g729_info Permitted G729 channels: 40 Encoders in use: 0 Decoders in use: 0 FreeSWITCH Box 2: 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP [sofia/transport/3888(a)172.17.17.17] 172.16.16.16 port 17056 -> 172.17.17.17 port 27276 codec: 18 ms: 20 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP [sofia/mpls/sip:C996_1@10.253.200.10:5060] 172.16.16.16 port 18966 -> 10.253.200.10 port 16486 codec: 18 ms: 20 freeswitch@internal> g729_info Permitted G729 channels: 40 Encoders in use: 1 Decoders in use: 1 Thanks in advance guys. Colin

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How to pause and resume play an audio file in conference?
by 王迪 10 Jan '13

10 Jan '13

How to pause and resume play an audio file in conference? For example: I create a conference room with some members, and play an audio file to them. When someone start-talking, I pause the playing. When nobody talking, I resume play the audio file. BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause and resume is for recording, not for my needs.

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Best practices question about SIP registration
by Steven Schoch 10 Jan '13

10 Jan '13

Please excuse me, I'm a VoIP newbie. I found the very helpful comment in conf/directory/default.xml. It says that the default domain is the IP address of the Freeswitch server. It seems to me that this could cause problems if the IP changed for some reason, or if the server has multiple NICs (which one does it choose in that case?) Is it better do just define a well known text string for the default domain, or even use a real DNS domain name? 2nd question is about SIP usernames. At least one phone I have found uses the MAC address as the default SIP user. Would it be better to avoid having a separate TFTP file for each phone, and just use the already unique MAC address as the user? -- Steve

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Issue: Invalid codec mpeg4-generic!
by Muqtheear S 10 Jan '13

10 Jan '13

mod_mp4 module is looking for Hint tracks in the .mp4 file. "Hint tracks are a series of instructions in a mp4 container file that tell a server how to transmit packets." To create hint tracks for a mp4 file is fairly simple with the mp4creator program, we just executed the: mp4creator -hint=<track to hint track number> [-p <payload>] <mp4file> As we know .mp4 has both video+audio data in it. We could also find audio and video encoding information of a .mp4 file as shown below. #mp4info minnale.mp4 mp4info version 1.5.0.1 minnale.mp4: Track Type Info 1 video MPEG-4 Simple @ L1, 97.833 secs, 136 kbps, 176x144 @ 12.000041 fps 2 audio MPEG-4 AAC LC, 97.568 secs, 64 kbps, 32000 Hz 3 hint Payload MP4V-ES for track 1 4 hint Payload mpeg4-generic for track 2 Metadata Tool: mp4creator 1.6 Now we are getting "Invalid codec mpeg4-generic!" error. Please guide me. Do we need to have mpeg4-generic codec in Freeswitch server? Thanks & Regards, Muqtheear.S

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