- FreeSWITCH-users - FreeSWITCH Mailing Lists

Connecting HP 8500 All in One FAX Machine to FS
by Nice Voip 06 Apr '11

06 Apr '11

Hi All, I have got HP 8500 All in One machine, including FAX, now i want to use it to send faxes to my VoIP provider through FS, how do i connect it to FS? it has wireless card, but i don't see any VoIP settings. Many thanks. - NV

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FS and Phone are in the same IP
by dunchan 06 Apr '11

06 Apr '11

Hi! My voip phone doesn't get the SIP bye request from the called party (via PSTN gateway) Phone is in the same machine with FS, IP addr is same too, Contact header is filled corretly. If phone is in other machine everything works fine. Should I change the NAT handling somewhere, or what? thanks, Viktor

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Need to enable NDLB_force_port to work properly with Windows PC Internet Gateway
by Vitalii Colosov 06 Apr '11

06 Apr '11

Hi list, I am testing the following configuration: Sip ATA adapter (Linksys PAP2) -> Windows PC (Internet Gateway) -> ... Internet ... -> FreeSWITCH Sip adapter sends REGISTER to the FS, from port 5060, however Windows Gateway transfers this packet to FS from a different port (for example 62000). FS replies to the port 5060 (it looks like it takes it from the SIP text information). So Windows Gateway receives the answer to the port 5060, but since it expects it to be received in 62000, it looks like it drops the packets. The only way to fix this, is to set NDLB_force_port= true in the internal profile configuration file. In this case, FreeSWITCH replies exactly to the port 62000, and windows forwards it to the Sip adapter on port 5060. Do you know whether it is possible to configure Windows to route properly? Or maybe there is a way to configure the Sip adapter? (Like it will say to windows PC - hey, don't change the port, use the same as I am using... :)) Or it is a fantastic wish) Otherwise I will have to create 2 internal sip profiles, one with NDLB_force_port = true (for those who uses Windows PC as the gateway), and another with NDLB_force_port = false (for those who use normal routers). And I need 2 default dialplans, etc... so it complicates the solution. Thank you, Vitalie

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Re: [Freeswitch-users] FS and Phone are in the same IP
by Viktor 06 Apr '11

06 Apr '11

There is no nat, preveously i got a tipp to check this section. Phone is a softphone. I think the problem is the FS doesn't forward the bye message to softphone which is listen on same ip but different port. FS suppuse the bye message sent to him. How can i avoid that? Thanks, Viktor Sent from Samsung Mobile Meftah Tayeb <tayeb.meftah(a)gmail.com> wrote: >where's nat in this path? >you say phone and fs have same ip >is this a hardphone or a softphone? >if this is a hardphone, how come a pc and a phone can fill in the same >network with same ip? >thanks >On 06/04/2011 07:27, dunchan wrote: >> Hi! >> >> My voip phone doesn't get the SIP bye request from the called party (via >> PSTN gateway) >> >> Phone is in the same machine with FS, IP addr is same too, Contact >> header is filled corretly. >> >> If phone is in other machine everything works fine. >> >> Should I change the NAT handling somewhere, or what? >> >> thanks, >> Viktor >> >> _______________________________________________ >> 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 >> > > >-- >Meftah Tayeb >inum: +883510001288000 >phone: +13477595883 > >

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SIP Proxy and DNS SRV options
by mazilo 06 Apr '11

06 Apr '11

Under the SIP Settings and Proxy and Registration sections in Line 1 and Line 2 TABs of a Linksys WRTP54G VoIP router with firmware v3.1.27, I have noticed the following options (see the attached image file below): 1. SIP Proxy-Require option in the SIP Settings section and Proxy option in the Proxy and Registration section. 2. Use DNS SRV, DNS SRV Auto Prefix, and Proxy Redundancy Method options in the Proxy and Registration section. The question I have is what are the equivalent settings in FS for the above mentioned options. Any examples will be a plus. http://freeswitch-users.2379917.n2.nabble.com/file/n6245865/SIP_DNS_SRV.png ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-Proxy-and-DNS-SRV-options… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Gateway ReInvite Problem with a=inactive in SDP
by Lars Bobka 05 Apr '11

05 Apr '11

Hi, I have a problem with ReInvites and bypass-media option. A call over a gateway comes in. The gateway is a shared line from a broadsoft application server. The Invite over the gateway shows an a=inactive in the SDP. v=0 o=BroadWorks 66603944 1 IN IP4 xxx.xxx.xxx.xxx s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 21568 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:30 a=fmtp:101 0-15 a=bsoft: 1 image udptl t38 a=inactive After sending the 200OK over the gateway ang getting an ACK, the broadworks server sends a ReInvite, because of the shared line. Now the FS should answer the ReInvite with a 200Ok and sendrecv, but the FS sends inactive to the broadworks server and I have no audio in the call. Only when I hold hold the call and unhold the call, the inactive messages is not given and I have 2way audio. I try <param name="NDLB-sendrecv-in-session" value="true"/> everywhere (external.xml, internal.xml, user.xml), but I had no luck. So could you please help me. Regards Lars ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

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simultaneous voice conference question
by Madovsky 05 Apr '11

05 Apr '11

When at least 3 persons are in conference I noticed that we can't hear the 3 voices in same time, only one voice at a time. is it normal ? thanks

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How to get remote gateway IP
by M.Emran 05 Apr '11

05 Apr '11

Hi, What is the way to read variable for termination gateway ip while it was ringing/answer and stop in lua script? -- Regards ---------- M Emran

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Sane Dialplan Regex
by Max Clark 05 Apr '11

05 Apr '11

Hello, I am trying to clean up the number of invalid calls being sent upstream to PSTN gateways. I've been researching numbering formats and could use some guidance/clarification. Currently I'm matching NADP calls looking by looking for a destination number matching 1 + 10 digits: <condition field="destination_number" expression="^(1\d{10})$"> >From what I've found NPA should be 201-999 and NXX should be 200-999 (ugly regex): <condition field="destination_number" expression="^(1[2-9][0-9]{2}[2-9]\d{2}\d{4})$"> Are there any cases where the NPA & NXX would not conform to this list? Thanks, Max

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video problem in conference with H264
by Seven Du 05 Apr '11

05 Apr '11

I tested with default 3000 conference and it just OK. But I have problem on H264. I tested with one Bria 3.1 on Mac and two XTP8886 hardware phones. http://www.gvscusa.com/xtp8886.html Bria 1003 XTP 1011/1012 call from 1003 to 1011 and from 1011 to 1003 both ok with videos. http://pastebin.freeswitch.org/15910 http://pastebin.freeswitch.org/15911 When 3 phones calling into 3000(conference), Everyone call see Bria(1003), but no one can say 1011 and 1012. Even when I muted 1003. http://pastebin.freeswitch.org/15913 As I said there's no problem with similar test with h263. Can anyone help take a look, thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow

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