December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Video codec negotiation
by peely 30 Dec '10

30 Dec '10

Hi, I have a profile which supports video codecs H264, H263, H261. With even the latest GIT version it seems that FS doesn't make any attempt to limit the codecs offered between endpoints. For example, if User Agent A has H.264 first priority then H.263 and H.261, but User Agent B has only H.263 and H.261 it seems that FS 200's with H.264 as the codec for UA A and H.263 for UA B, as a result, no video exchanged between UAs. I have tried greedy codec negotiation but still the same. If I remove H.264 from User Agent A then both negotiation H.263. Given that FS only supports passthrough video, shouldn't it enforce the same video codec end to end or no video codec at all? Regards, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-codec-negotiation-tp587… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_callcenter features
by rajkumar 30 Dec '10

30 Dec '10

Hi, I am developing an application with mod_callcenter. I need to know the following about mod_callcenter. * Is it possible to add/update/delete the queue configurations dynamically without using static xml configurations. * How can I control the call flow (for playback message and recording) before and after bridging the call. Thanks in advance regards rajkumar k -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-callcenter-features-tp586… Sent from the freeswitch-users mailing list archive at Nabble.com.

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ERR : NORMAL CIRCUIT CONGESTION
by Ashish Chavan // Viva 30 Dec '10

30 Dec '10

Dear Team, Here I am having 2 port CTI card on FreeSWITCH server. Here I am having Asterisk server with pep2t. whenever I am executing originate command on fs_cli like #> originate openzap/smg_prid/a/1001@g1 &echo getting -- 2010-12-29 14:28:00.530435 [WARNING] ozmod_sangoma_boost.c:341 TX EVENT: CALL_START:(80) [w1g1] CSid=28 Seq=54 Cn=[N/A] Cd=[1001] Ci=[N/A] Rdnis=[] 2010-12-29 14:28:00.531591 [WARNING] ozmod_sangoma_boost.c:1473 RX EVENT (N): CALL_START_NACK:(82) [w256g256] Rc=34 CSid=28 Seq=28 2010-12-29 14:28:00.531591 [WARNING] sangoma_boost_client.c:221 TX EVENT (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=28 Seq=55 2010-12-29 14:28:00.532579 *[ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]* 2010-12-29 14:28:00.532579 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] whereas on /var/log/sangoma_mgd.log getting---- Dec 29 10:12:05 freeswitchtest2 sangoma_prid: HeartBeat timeout (current:10:12:05 last:10:11:55 grace:0) Dec 29 10:12:15 freeswitchtest2 sangoma_prid: HeartBeat timeout (current:10:12:15 last:10:12:05 grace:0) Dec 29 10:12:15 freeswitchtest2 sangoma_prid: *Assuming application is dead* Dec 29 10:12:25 freeswitchtest2 sangoma_prid: HeartBeat timeout (current:10:12:25 last:10:12:15 grace:0) Incase of any further queries, Please feel free to mail me or contact me on the numbers provided below. Thanks & Regards, Ashish Chavan Server Administrator Viva Infomedia Pvt. Ltd. 242, Oshiwara Industrial Centre, Opp. Oshiwara Bus Depot, New Link Road, Goregaon West, Mumbai 400104. Direct: +91.22.4293 0162 Board: +91.22.4293 0100 Email : ashish(a)vivaconnect.in Web : www.vivaconnect.in Viva Infomedia: Awarded as Best SME (E-Commerce) at CNBC Emerging India Awards 2009 Disclaimer: This e-mail and any files transmitted with it are for the sole use of the intended recipient(s) and may contain confidential and privileged information. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies and the original message. Any unauthorized review, use, disclosure,dissemination, forwarding, printing or copying of this email or any action taken in reliance on this e-mail is strictly prohibited and may be unlawful. The recipient acknowledges that Viva Infomedia Pvt.Ltd. or its subsidiaries and associated companies are unable to exercise control or ensure or guarantee the integrity of/over the contents of the information contained in e-mail transmissions and further acknowledges that any views expressed in this message are those of the individual sender and no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of Viva Infomedia Pvt.Ltd. Before opening any attachments please check them for viruses and defects.

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mutilple destination phone ringing on incoming call, skypopen dial plan from Sip X-lite
by Norman Lam 30 Dec '10

30 Dec '10

Hi Everyone 1. Could anybody tell me how to have multiple destination ringing from an incoming skypopen, Currently is set to <param name="destination" value="1000"/> But I want it to ring on 1001, 1002,1003, and so on. 2. Does anybody have a outgoing dial plan from sip X-lite to skypopen ( skype network), I try looking at the Mod skypopen Skype Endpoint and Trunk wiki site but couldn't figure it out. Please advise , Thank in advance Norman

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DTMF missing
by Sam 30 Dec '10

30 Dec '10

Hi,, I have installed the latest ver of freeswitch and i have configured the conference. now when i punch in the digits for password , i could see that the DTMF digits are missed on fs_cli. it only happens when i dial it from polycom or cisco phones. I have tried with and without these values below:- <param name="dtmf-type" value="rfc2833"/> <param name="pass-rfc2833" value="true"/> traces fetched: 192.168.2.49:5060 -> 192.168.2.190:5060 INVITE sip:7050@192.168.2.190 <sip%3A7050(a)192.168.2.190> SIP/2.0..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" < sip:7028@192.168.2.190 <sip%3A7028(a)192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To: <sip:7050@192.168.2.190 <sip%3A7050(a)192.168.2.190>>..Call-ID: 0017592a-eb33001a- 63da3294-1a7bfdfa@192.168.2.49..Max-Forwards: 70..Date: Fri, 24 Dec 2010 10:13:52 GMT..CSeq: 102 INVITE..User-Agent: Cisco-CP7940G/8.0..Contact: <sip:7028@192.168.2.49:5060;transport=udp>..Proxy-Authorization: Digest username="7028" ,realm="192.168.2.190",uri="sip:7050@192.168.2.190<sip%3A7050(a)192.168.2.190>",response="a668f5c480285b35e7ff6bcd446879f0",nonce="d2c540f2-8487-4d87-bdab-871585253eb8",cnonce="0a6c4176",qop=auth,nc=00000001,algorithm=MD5..Expires: 180..Accept: application/sdp ..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub..Content-Length: 220..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 16102 0 IN IP4 192.168.2.49..s=SIP Call..t=0 0..m=audio 17298 RTP/AVP 0 8 18..c=IN IP4 192.168.2.49..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=sendrecv.. 192.168.2.190:5060 -> 192.168.2.49:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.49:5060;branch=z9hG4bK1531f395..From: "7028" <sip:7028@192.168.2.190 <sip%3A7028(a)192.168.2.190>>;tag=0017592aeb3305185b4a37ba-615f498d..To: <sip:7050@192.168.2.190 <sip%3A7050(a)192.168.2.190>>;tag=2XXUZpgr1rvgc..Call-ID: 0017592a-eb33001a-63da3 294-1a7bfdfa@192.168.2.49..CSeq: 102 INVITE..Contact: <sip:7050@192.168.2.190:5060;transport=udp>..User-Agent: NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIF Y, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Session-Expires: 180 0;refresher=uas..Min-SE: 120..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 249..Remote-Party-ID: "7050" <sip:7050@192.168.2.190<sip%3A7050(a)192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1293163579 129 3163580 IN IP4 192.168.2.190..s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 22050 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20.. But the dtmf are not missed when punched on eyebeam softphone. And all the phones have RFC 2833. traces fetched for softphone:- 192.168.2.17:6182 -> 192.168.2.190:5060 INVITE sip:7050@192.168.2.190 <sip%3A7050(a)192.168.2.190> SIP/2.0..To: < sip:7050@192.168.2.190 <sip%3A7050(a)192.168.2.190>>..From: 7001< sip:7001@192.168.2.190 <sip%3A7001(a)192.168.2.190>>;tag=6c557c1e..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport..Call-ID: d32ffe546570a77e..CS eq: 2 INVITE..Contact: <sip:7001@192.168.2.17:6182>..Max-Forwards: 70..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..Proxy-Authorization: Digest username="7001",rea lm="192.168.2.190",nonce="d0b4db6e-bd76-447f-9076-2e6b7809cb54",uri=" sip:7050@192.168.2.190 <sip%3A7050(a)192.168.2.190>",response="45fb9eb4f0e0e4fffd87a22769a007ba",cnonce="1c27f3687059b16d",nc=00000001,qop=auth,algorithm=MD5..User-Agent: eyeBeam release 3007n stamp 17816..Content-Length: 233....v=0..o=- 27833664 27833670 IN IP4 192.168.2.17..s=eyeBeam..c=IN IP4 192.168.2.17..t=0 0..m=audio 6398 RTP/AVP 0 18 101..a=alt:1 1 : 2C830AD9 0000004F 192.168.2.17 6398..a=fmtp:101 0-15..a=rtpmap: 101 telephone-event/8000..a=sendrecv.. 192.168.2.190:5060 -> 192.168.2.17:6182 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.17:6182;branch=z9hG4bK-d87543-879697683-1--d87543-;rport=6182..From: 7001 <sip:7001@192.168.2.190 <sip%3A7001(a)192.168.2.190>>;tag=6c557c1e..To: < sip:7050@192.168.2.190 <sip%3A7050(a)192.168.2.190>>;tag=H0ctQv7KNgU2j..Call-ID: d32ffe546570a77e..C Seq: 2 INVITE..Contact: <sip:7050@192.168.2.190:5060;transport=udp>..User-Agent: NOVANET..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Support ed: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Session-Expires: 1800;refresher=uas..Min-SE: 120.. Content-Type: application/sdp..Content-Disposition: session..Content-Length: 249..Remote-Party-ID: "7050" < sip:7050@192.168.2.190 <sip%3A7050(a)192.168.2.190>>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1293161640 1293161641 IN IP4 192.168.2.190.. s=FreeSWITCH..c=IN IP4 192.168.2.190..t=0 0..m=audio 24852 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20.. Any thing you can think how it can happen? Regards Sam

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ftmod_sangoma_isdn: dialtone in overlap dialing
by Nagalenoj H. 30 Dec '10

30 Dec '10

Dear friends, I would want to know how to configure ftmod_sangoma_isdn to send dialtone. My setup is as follows, Telcom <---> Freeswitch <---> PBX <----> Extensions 1..n My pbx has got overlap dial enabled and it is configured to seize the trunk when it receives '0' from extensions. When I connect the PRI to my pbx, and when '0' is dialled from an extension, my pbx seizes the trunk and I'm able to hear dialtone. But, when I connect the PRI to freeswitch and connect freeswitch to pbx and when '0' is dialled from an extension, I'm not hearing dialtone. Whether some configuration is available to ask ftmod_sangoma_isdn to send dialtone before started to collect digits in overlap mode? or any other way to give dialtone?? -- Regards, Nagalenoj H.

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unregister extension from cli
by Madovsky 30 Dec '10

30 Dec '10

is there any way to unregister an extension from CLI ? for example I register my phone with 123456 and from CLI I disconnect it.. Thanks

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Briding two sip-trunks
by Ralph Eckhard 30 Dec '10

30 Dec '10

Hi there, I'm trying to use FreeSwitch to connect my MS Lync (previously OCS) to a SIP-trunk. Problem is that Lync only works using TCP-based trunks, while my SIP-provider uses UDP. I understand that by using FreeSwitch I can bridge these two protocols. I installed FreeSwitch on my Lync server and configured both the external (to sip-provider) and internal (to Lync) sip profiles. These both work; when i dial in from remote to a DDI on my trunk, i can see the incoming call in the FS CLI. Same goes for outgoing calls; when dialing a number from my lync-client, the FS CLI logs the call. Now, what do i config to bridge the two trunks? I don't need any extra extensions or otherwise, all call handling is done by Lync. All calls coming in from my external sip trunk need to be forwarded to my internal sip, and all calls originating from my internal sip need to be forwarded to my external sip trunk. Can someone point me in the right direction? Thanks in advance ;) Ralph

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New FS sounds testing
by Michael Collins 30 Dec '10

30 Dec '10

FreeSWITCHers, We have the new sounds rolled and I'd like to do some expanded testing before I update in git. If anyone wants to get the latest sounds then go to your src directory and edit build/sounds_version.txt. Change en-us-callie from 1.0.13 to 1.0.14 and then do "make cd-sounds-install". The latest sounds should get installed. Please try this on your non-production systems first! :) Please reply to me off-list and let me know if you run into any problems or if the sounds are okay. I would like feedback either way. If we don't have any issues then I'll bump the sounds version in git and everyone will be good to go. Thanks! -Michael P.S. - If you would like to review which sound prompts are new to 1.0.14 then check this commit: http://fisheye.freeswitch.org/browse/freeswitch.git/docs/phrase/phrase_en.x…

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Re: [Freeswitch-users] multiple-registration for same extension
by Madovsky 29 Dec '10

29 Dec '10

ok I think the problem is not FS... thanks ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Wednesday, December 29, 2010 2:00 PM Subject: Re: [Freeswitch-users] multiple-registration for same extension I have this in directory/default.xml <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> tried this in my dialplan : <action application="bridge" data="user/${dialed_extension}"/> <action application="bridge" data="${sofia_contact(${dialed_user}@${dialed_domain})}"/> this in internal.xml : <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="dbname" value="share_presence"/> <param name="presence-hosts" value="$${domain}"/> so when an outbound call arrives : - if one phone with one extension is registered it works well, it receives the call no matter from whatever node - if 2 phones with same extension are registered on 2 different nodes it works well now - if 2 phones with same extension are registered on the same node only 1 phone rings (the latest registered) tell if I need to pastebin the logs (don't remember the link) Thanks ----- Original Message ----- From: Avi Marcus To: FreeSWITCH Users Help Sent: Wednesday, December 29, 2010 12:26 PM Subject: Re: [Freeswitch-users] multiple-registration for same extension On Wed, Dec 29, 2010 at 7:11 PM, Madovsky <infos(a)madovsky.org> wrote: yes, <param name="multiple-registrations" value="true"/> and restart FS. also is it need to put this line in internal.xml or external.xml knowing that registrations are effective only in internal.xml ? > It's worked for me several times, however, I usually bridge to "user/$extension". but if you use sofia_contact in the dialplan bridge it gives to you sofia/$${domain}/extension@$${domain} so how to use user/extension in mutliple-registrations ? as I use a cluster with SRV NAPTR for outbound calls maybe it's the problem. I noticed that if the call arrives in the same node as the one where phones are registered it works. if not, only 1 phone or no phone rings. weirdly there is nothing abnormal on debug log I guess you're not clustering the DB and replicating the internal registrations across the servers? ----- Original Message ----- From: Avi Marcus To: FreeSWITCH Users Help Sent: Wednesday, December 29, 2010 11:44 AM Subject: Re: [Freeswitch-users] multiple-registration for same extension You set: <param name="multiple-registrations" value="true"/> and did a reloadxml, and waited for both to register again? It's worked for me several times, however, I usually bridge to "user/$extension". -Avi On Wed, Dec 29, 2010 at 6:18 PM, Madovsky <infos(a)madovsky.org> wrote: I enabled multiple-registrations to true in internal.xml (which is the difference between true and contact ?) added this line to my dialplan <action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/> but only the phone registered lastly rings... thanks Franck _______________________________________________ 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 ------------------------------------------------------------------------ _______________________________________________ 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 _______________________________________________ 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 ---------------------------------------------------------------------------- _______________________________________________ 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

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