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Can FS work with an external packet relay such as rtpproxy?
by Adino 30 Dec '09

30 Dec '09

What is the current communication protocol with the media proxy? I noticed that MGCP is not on roadmap - can anyone provide a little more info on this please? Thanks

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tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch
by Kristoff Bonne 30 Dec '09

30 Dec '09

Hi all, This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" device for just 15 euro. This is a device which has on one side a USB-connector and on the other side 2 RJ-11 connectors (one FXO and one FSX). Internally, the device seams to contain a tigerjet 560C chipset. (see here: http://www.tjnet.com/chips/tiger560C.htm) What is interesting on this device is that is uses standard USB device-classes that are by default supported by most operating-systems: usb-sound and usb-hid. When I connect it to my server (mac mini 3G running debian), the system automatically recognises these two classes [168391.922479] usbcore: registered new interface driver hiddev [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on usb-0001:10:1b.1-1 [168391.939548] usbcore: registered new interface driver usbhid [168391.943984] usbhid: v2.6:USB HID core driver [168392.154596] usbcore: registered new interface driver snd-usb-audio And -behold- when I connect a handset in one of the port, I even get a dialtone and I can sent out DTMF-dialtone which are somehow partly (But I have no idea what program actually generates this dialtone !!!) Now, the question: Any idea if / how this can incorperated into freeswitch? Is there a way to use this device to connect a phone to freeswitch without having to go throu a SIP-client first. Cheerio! Kr. Bonne. -- jabber/gtalk: kristoff(a)krbonne.net

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Hold is broken in trunk 16055
by Lei Tang 30 Dec '09

30 Dec '09

Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. ====sip trace for trunk 16055 ====re-invite request sent to fs when client hold the line INVITE sip:65960581@10.56.0.189:5060;transport=udp SIP/2.0 From: <sip:37186004616@10.56.0.189:5060>;tag=1c6494 To: <sip:65960581@10.56.0.189:5060>;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: <sip:37186004616@10.56.90.223 <sip%3A37186004616(a)10.56.90.223>> Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: <sip:37186004616@10.56.0.189:5060>;tag=1c6494 To: <sip:65960581@10.56.0.189:5060>;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =====bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: <sip:37186004616@10.56.0.189:5060>;tag=1c6494 To: <sip:65960581@10.56.0.189:5060>;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: <sip:65960581@10.56.0.189:5060;transport=udp> User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ======sip trace for fs 1.0.4 =====re-invite request sent to FS when client want to hold the all INVITE sip:65960581@10.56.0.189:5060;transport=udp SIP/2.0 From: <sip:37186004616@10.56.0.189:5060>;tag=1c8147 To: <sip:65960581@10.56.0.189:5060>;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: <sip:37186004616@10.56.90.223 <sip%3A37186004616(a)10.56.90.223>> Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: <sip:37186004616@10.56.0.189:5060>;tag=1c8147 To: <sip:65960581@10.56.0.189:5060>;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: <sip:37186004616@10.56.0.189:5060>;tag=1c8147 To: <sip:65960581@10.56.0.189:5060>;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: <sip:65960581@10.56.0.189:5060;transport=udp> User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 254 v=0 o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 ACK sip:65960581@10.56.0.189:5060;transport=udp SIP/2.0 Contact: <sip:37186004616@10.56.90.223 <sip%3A37186004616(a)10.56.90.223>> From: <sip:37186004616@10.56.0.189:5060>;tag=1c8147 To: <sip:65960581@10.56.0.189:5060>;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 ACK Date: Tue, 29 Dec 2009 03:20:15 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport Content-Length: 0 INVITE sip:65960581@10.56.0.189:5060;transport=udp SIP/2.0 From: <sip:37186004616@10.56.0.189:5060>;tag=1c8147 To: <sip:65960581@10.56.0.189:5060>;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29658 INVITE Contact: <sip:37186004616@10.56.90.223 <sip%3A37186004616(a)10.56.90.223>> Content-Type: application/sdp Content-Length: 473 Date: Tue, 29 Dec 2009 03:20:18 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport v=0 o=sipX 5 35 IN IP4 10.56.90.223 s=call c=IN IP4 10.56.90.223 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 From: <sip:37186004616@10.56.0.189:5060>;tag=1c8147 To: <sip:65960581@10.56.0.189:5060>;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29658 INVITE User-Agent: PowerIVR Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 From: <sip:37186004616@10.56.0.189:5060>;tag=1c8147 To: <sip:65960581@10.56.0.189:5060>;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29658 INVITE Contact: <sip:65960581@10.56.0.189:5060;transport=udp> User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 242 v=0 o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 ACK sip:65960581@10.56.0.189:5060;transport=udp SIP/2.0 Contact: <sip:37186004616@10.56.90.223 <sip%3A37186004616(a)10.56.90.223>> From: <sip:37186004616@10.56.0.189:5060>;tag=1c8147 To: <sip:65960581@10.56.0.189:5060>;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29658 ACK Date: Tue, 29 Dec 2009 03:20:19 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport Content-Length: 0 -- Lei.Tang lei.tlfly(a)gmail.com

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freeswitch init
by Neil Patel 30 Dec '09

30 Dec '09

Hi All, On the Freeswitch init page (Debian) the setup you to create a user "freeswitch" to run the FS process. When I did this, freeswitch started up but wasn't able to find/open channels to my sangoma/wanpipe hardware. Is this because: 1. The sangoma hardware was installed and running through root, which is a different user than the one running FS? 2. Wanrouter is not getting started before freeswitch on system boot? Is there a disadvantage to setting up the init script for FS to be run by root? Will that solve this problem? Thanks in advance from a Linux/FS newbie, Neil

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[Dialplan?] Call either sent to VM or phone doesn't ring at all
by Fred-145 30 Dec '09

30 Dec '09

Hello I'm running "1.0.trunk (15841)" on Linux CentOS with a the default settings. After succesfully connecting a Windows PC running XLite (EyeBeam, really) and a GrandStream IP phone to Freeswitch, I try to make calls, but am having the following issues: 1. When calling XLite from GS, XLite rings, but when I pick up the call, the caller is sent to voice-mail right away ("the person on extension 1001 is not available") 2. When calling GS from XLite, the GS phone doesn't even ring. FWIW, the phones seem to have registered OK: freeswitch@internal> sofia status profile internal Registrations: ======================================================== Call-ID: Yzc2MzFiMjVhNGQwNjE5YWU1OGZjNGMxMTg0NDIwNDA. User: 1001(a)192.168.0.7 Contact: "Freeswitch" <sip:1001@192.168.0.1:41380;rinstance=0516dddfe24deef4> Agent: eyeBeam release 1104a stamp 54437 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:34:00) Host: centos.workgroup IP: 192.168.0.1 Port: 41380 Auth-User: 1001 Auth-Realm: 192.168.0.7 MWI-Account: 1001(a)192.168.0.7 Call-ID: 3f6d4ebebd5e829f(a)192.168.0.9 User: 1003(a)192.168.0.7 Contact: "user" <sip:1003@82.237.75.54> Agent: Grandstream BT120 1.1.0.3 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:44:02) Host: centos.workgroup IP: 192.168.0.9 Port: 5060 Auth-User: 1003 Auth-Realm: 192.168.0.7 MWI-Account: 1003(a)192.168.0.7 ======================================================== Has someone seen this type of behavior? Thanks for any hint. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Port item in "sofia status" output?
by Fred-145 30 Dec '09

30 Dec '09

Hello I was wondering: What is the meaning of the Port item that shows up when I run "sofia status"? > sofia status profile internal reg [...] IP: 192.168.0.1 Port: 59724 Is is the UDP port on which the phone listens for incoming SIP messages, or is it the RTP port on which it expects to receive voice packets from the remote phone to which is it will be connected? BTW, does RTP use a single port to TX/RX voice, or does it use the RTP port and the one that immediately follows? Thank you. -- View this message in context: http://old.nabble.com/Port-item-in-%22sofia-status%22-output--tp26965939p26… Sent from the Freeswitch-users mailing list archive at Nabble.com.

1 0

Cisco 501's
by Brian West 30 Dec '09

30 Dec '09

Anyone have access to these phones? Two of them if possible and provisioning information? Thanks, Brian

2 1

Server Configuration for 50 concurrent sessions
by Sharad 30 Dec '09

30 Dec '09

Hi I just want to know what should be the approx configuration of the server for 50 concurrent call sessions having 3000-4000 users. Regards -- View this message in context: http://n2.nabble.com/Server-Configuration-for-50-concurrent-sessions-tp4231… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Message Wait Lamp for an unregistered user
by Sharad 29 Dec '09

29 Dec '09

Hi, I am trying to integrate Freeswitch as a Media Server with our own SIP Server. For this, initially, I am using freeswitch for Auto Attendant & Voicemail Application. In this case, all the SIP users are registered with my SIP Server & whenever Auto Att or Voicemail application is required, my SIP Server just forward the call to Freeswitch. Everything seems ok & working fine except some small small points. One of the point is - whenever there is a voice message in the mailbox of a user, Freeswitch is not generating the MWI (Notify) to my SIP Server. So just want to know , is there any way so that freeswitch can light up / light -off the message wait lamp for the users of my SIP Server. Thanks in advance for your answers. Regards Sharad -- View this message in context: http://n2.nabble.com/Message-Wait-Lamp-for-an-unregistered-user-tp4226726p4… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

Cant able to make call through X-lite
by Scott Fernandez 29 Dec '09

29 Dec '09

Hello , I have a VOIP account which I configured through X-lite and it works fine. However, when I configure the same account in Freeswitch, the status shows as UP. If I call through freeswitch extension (ex. 1001) via X-lite client it says that USER_BUSY. But when I dial the same number through API command, I am able to make a call and bridge it. What could be the issue and Can any one assist me on this? I have pasted the logs in this URL http://pastebin.freeswitch.org/11635 Regards, Scott

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