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Slightly OT - Freeswitch and Vitelity?
by dave.redmore＠spigotnetworks.com 24 Aug '11

24 Aug '11

Anyone have any experiance with Vitelity as an ITSP? Curious about both interop with Freeswitch and in general. Thanks, Dave Redmore Spigot Networks, Inc.

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FreeSWITCH Conference Call Today
by Michael Collins 24 Aug '11

24 Aug '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_24 Alexandr from the Homer project has been busy updating his software so we're having him back to talk to us again on the subject. We also have a few janitorial items to go over. Thanks! -Michael

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Problem with freeswith and a Digium card
by Caez 24 Aug '11

24 Aug '11

Hello, I have to problems with freeswith and my digium card whith dahdi and freetdm. I'm a french user. - I have no signal when the user hangs up. I have to wait for the timeout before the line is available. If I call a specific phone, I have to wait for the timeout to 60s before using the line - I don't have the phone number for the incoming and outgoing calls. For the incoming, i have 00000000000 For the outgoing, "Anonymous" I'm trying to change my tones.conf files, to add some options on freetdm.conf.xml, but the problem persists Does anyone have a solution? Thank you in advance Sebastien

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V92 modem with freeswitch
by ankIT WALiA 24 Aug '11

24 Aug '11

Hi all, I found a V.92 56k PCI Modem in my old hardware. I want to run some basic functionality test with my PSTN phone line and FS. Can I connect my phone line with FS using this card. If yes how do we configure to use this card. I am new to telephony. I don't know if whatever I said above make sense. Thanks Ankit

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Ignore INFO DTMF
by Mitja Thomas 24 Aug '11

24 Aug '11

Hi, we are experiencing problems with DTMF between Snom Sip Phones and FS. The DMTF Type is set to SIP INFO (sip info only on Snom Phones). We could nail it down to a missing "telephone-event" parameter in sip sdp: INVITE sip:***6@spklw.x;user:phone SIP/2.0 Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-15eyqwai33r1;rport From: "Test" <sip:8000@spklw.x>;tag=qqnbrunjmq To: <sip:***6@spklw.x;user:phone> Call-ID: 3c33e6c729b3-t9yol2613q6v CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:8000@ip:5060;line=9lwqdnrx>;reg-id=1 X-Serialnumber: 000413262C33 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 244 v=0 o=root 334284706 334284706 IN IP4 phoneip s=call c=IN IP4 phoneip t=0 0 m=audio 12610 RTP/AVP 8 9 a=direction:both a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a:ptime:20 a=rtcp-xr:stat-summary=loss,dup,jitt a=sendrecv That resulted in sofia_glue.c disabling DTMF. A slight change in sofia_glue.c (Line 4839 in commit 71964f61ac4817ecfc516edaf4b643f41236868d) did the trick for us: if (tech_pvt->dtmf_type == DTMF_INFO) { //EWETEL } else { //EWETEL switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Disable 2833 dtmf\n"); switch_channel_set_variable(tech_pvt->channel, "dtmf_type", "none"); tech_pvt->dtmf_type = DTMF_NONE; te = tech_pvt->recv_te = 0; } //EWETEL Now the question is: Is this a too restrictive behaviour in FS or a shortcoming from the Snom phones (which would mean we have to submit a bug, wait, wait, wait and then hopefully someday it gets fixed :) )? Regards, Mitja -- Mitja Thomas Vertrieb Geschäftskunden Branchenlösungen / Service Entwicklung Telefon: +49 (0) 441 - 8000-4916 E-Mail: mitja.thomas(a)ewe.de EWE TEL GmbH Cloppenburger Straße 310 26133 Oldenburg E-Mail: info(a)ewe.de Internet: www.ewe.de Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Dr. Werner Brinker Geschäftsführung: Konrad Meier (Vorsitzender), Dirk Brameier, Ulf Heggenberger, Norbert Westfal

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Question about ext-rtp-ip and ext-sip-ip
by Bryan Lemon 24 Aug '11

24 Aug '11

>From what I am seeing, freeswitch is not honoring the ext-*-ip variables in the invite messages. Using the following command entered on fs_cli: originate {origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212 &javascript(somejavascript.js), the invite message is below. Shouldn't the instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*? This is causing the rtp packets to be sent to the incorrect location, and resulting in 1-way audio. send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988: ------------------------------------------------------------------------ INVITE sip:13044150838<https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.* SIP/2.0 Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD Max-Forwards: 69 From: "SomeName" <sip:bryan_bryanlemon_com@209.216. *.*;transport=udp>;tag=HpU27XSQHmX1g To: <sip:15555551212@209.216.*.*> Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a CSeq: 16401016 INVITE Contact: <sip:gw+didforsale@204.111.*.*:5080;transport=udp;gw=didforsale> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11 14-15-26 -0500 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: 203 X-FS-Support: update_display Remote-Party-ID: "SomeName" <sip:15555551212@10.0.10.144 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144 s=FreeSWITCH c=IN IP4 10.0.10.144 t=0 0 m=audio 32712 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.10.144,10.0.10.144 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5060 BIND-URL sip:mod_sofia@10.0.10.144:5060 freeswitch@internal> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.10.144 Ext-RTP-IP 204.111.*.* SIP-IP 10.0.10.144 Ext-SIP-IP 204.111.*.* URL sip:mod_sofia@10.0.10.144:5080 BIND-URL sip:mod_sofia@10.0.10.144:5080 Thank you, Bryan Lemon (302) 648-2747 <https://www.google.com/voice/m/caller?number=+13026482747>

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 62, Issue 178
by Petr Nyklicek 23 Aug '11

23 Aug '11

Hi, thanks for your rep[y. Ca you give me any hint how can do it ? I need (z)rtp routing :( > ....... although is probably possible at least for ZRTP). > Moises Silva Petr Nyklíček

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Hold and BLF - disable flashing or pickup for held call?
by Alex Crow 23 Aug '11

23 Aug '11

Hi. I have some snom 370s, and I noticed that when a monitored extension has held a call then the corresponding BLF lamp flashes exactly as if the monitored extension is ringing, and it is then possible to steal the call from the "holder" by pressing the button. Ideally I'd like neither of these things to happen. Is it possible to disable one or both for a held call? I have a polycom IP 650 with productivity licence and it shows the status of the holding extension as ringing (with the musical notes animation on the status display - not great) but at least it doesn't steal the call. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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Re: [Freeswitch-users] Freeswitch say digits too quick
by Sam 23 Aug '11

23 Aug '11

If I wanted to record my own digit sound files is there a way to set the directory for the say app?

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How to authencticate with mod_rtmp?
by Dmitry Kravchenko 23 Aug '11

23 Aug '11

Hi! It is said in book that user registry is single and centralized for all components (p. 64). But while trying to login via FLEX telephone to default number of 1000 with password of 1234, I get "Authentication failed" message. In freeswitch CLI I see message 2011-07-19 15:46:51.473828 [WARNING] rtmp.c:227 Authentication failed for 1000(a)192.168.10.196 where 192.168.10.196 is true IP of my freeswitch machine. Connection with X-Lite goes OK with message 2011-07-19 15:50:19.473829 [WARNING] sofia_reg.c:1337 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000(a)192.168.10.196] from ip 192.168.10.56 If I put wrong password into X-Lite then I get 2011-07-19 15:50:19.493774 [WARNING] sofia_reg.c:1295 SIP auth failure (REGISTER) on sofia profile 'internal' for [1000(a)192.168.10.196] from ip 192.168.10.56 So my conclusion is that SIP phone is treates as internal profile by default. Is it possible to set RTMP profiles as internal too? Or may be I should better create exteral accounts to authenticate with RTMP? Thanks.

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