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  • How can I tell if ZRTP is enabled in a Twinkle SIP call?

    - by komputes
    I recently attended a talk about GNU Telephony. I was informed that Twinkle supports ZRTP for encrypted SIP calls. I went into Edit User Profile Security and made sure that ZRTP was enables and that all boxes were checked. I asked a friend to do the same and then we called each other. There is no immediate indication that I can see that the call is secure. How can I tell if ZRTP is enabled in a Twinkle SIP call?

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  • process of connecting RTP with SIP via SDP & land lines

    - by TacB0sS
    Hello to everyone, I have a problem with starting a media session and to combine it with my SIP client. I've designed a recursive SIP client that reuse the same request template to send the next requests to server, according to the acceptable sequences noted in the RFC's, and examples that I read. as far as I could tell the SIP part is working fine registers to server invites, and authenticates. I didn't complete any calls to clients yet because of the content header needs to be filled up (which I didn't yet so I get a 503 from the server which is OK I guess). for a long time I didn't know where to start with the media session, and slowly learned how to use the JMF and I've constructed an object that handles RTP transmitting, now I'm standing at the cross road, on the one hand I have my SIP signaling but it needs the SDP content header to complete the invite, and on the other I have the RTP which is knows how to p2p. For me to complete my design I require your help with the following questions: Is there an easy//a simple//an implemented way to convert the Audio/Video format from the JMF into SDP media headers? or even a generator that I would input all the parameters for the content header, and it would generate a content header fast, or do I have to implement this myself? Once I've finished constructing the SDK and the SIP is up and running and I get an OK response from the server (after ringing and all), how do I start the media session? do I connect p2p according to caller details I send in the SIP invite? If 2 is correct, then how does a connection to land lines would be? does land lines knows that once they send an OK back to server they listen/start RTP session on a specific port? Or did I get everything wrong? :-/ I really appreciate any help I could I get, I looked every where for answers but they are not clear, they ignore question 2 as if it was an obvious thing, but for me it just isn't. Thank in advance, Adam Zehavi.

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  • Using Google Voice with an internal SIP Server

    - by BHelman
    Let me be upfront and say first that I am new to the entire details of VoIP. My former understanding was just the extent of Skype. Don't worry, I understand a lot more of it now. The situation is this. I have a Google number that is actually very close to the area in which I live. It's convenient as it is not long distance for everyone. I love its features and etc, but I want it to forward to a VoIP phone, which will be my residential phone. Obviously, Google does not allow forwarding calls to domains (yet). So I use SIPGate with a SIPGate number to forward to a softphone for now. I can configure a VoIP phone to interact with my account easily enough. The problem lies with SIPGate itself really. Google Voice gives free unlimited inbound and outbound calling. SIPGate charges you for outbound. So a VoIP phone would work, but I could never make a call on it (for free). So let's say I setup an Asterisk server, or any other SIP server. What is the best way to go about linking my server to Google Voice? I looked into IPKall but it only specifies inbound calling and not outbound. Or is that just assumed? Does an SIP server handle outbound calling by itself?

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  • n900 - SIP - Google Voice - DTMF

    - by Walter White
    Hi all, I've been using Google Voice on my n900 and it works reasonably well, but DTMF tones do not work whatsoever. According to this page, it is fixed and has been for quite some time. https://bugs.maemo.org/show_bug.cgi?id=5505 Has anyone had any luck with SIP on the n900 and DTMF tones? An earlier version of Skype on the n900 didn't support DTMF, but that was fixed. I would have thought this bug would have been fixed for some time now. Thanks, Walter

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  • SIP server ("gateway") for joining accounts

    - by Tomas Srna
    Hello, I have a phone supporting 2 accounts, but I need 4 accounts. Is it possible to install some sort of SIP server/gateway/proxy (on a linux server), that would register those 4 accounts and I would be able to connect to it as if it was 1 account? (With dialing rules, etc.) 3 of the accounts have incoming numbers. Thanks. Tomas.

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  • What is SIP trunking?

    - by hypnocode
    Can someone explain to me in plain English what SIP trunking is, please? I've read about it on Google, but I don't really grasp it yet. Does it allow a VoIP call to be placed outside of the LAN? So if you had Asterisk setup as the PBX, then IP calls could be made outside of the network? Am I close or am I just saying stupid words?

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  • Trixbox: external SIP with no sound

    - by Leandro Vidal
    I have a trixbox server and every works find except the external SIPs. Inside net all sound goes fine, but if I use a SIP phone outside the net, I can connect, I can receive calls but I there is no sound. I have this text in the sip_nat.conf: nat=yes externhost=xxxxx.dyndns.org localnet=192.168.1.0/255.255.255.0 localhost=192.168.1.210 externrefresh=10 qualify=yes And I have the ports from 5036 to 5082, 4569 and from 10000 to 20000 redirected to 192.168.1.210 on TCP and UDP. What's wrong? Thank you very much in advance

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  • email to sip voicemail

    - by alfredwesterveld
    Hi all, I don't know if this is the correct place to answer this question, but here it goes. I have been googling for a cheap email to sip voicemail service. That is because I have got a Linksys/Cisco SPA-941 phone which has a led which will light up when a new message comes in inbox(somebody calls me). So what I want is the following. I want the e-mail(title only is enough) recorded(By computer voice) and sent to my phone which I can playback when the led lights up. Like I said above I was unsuccessful googling for a service like this and I hope somebody knows if this service exists. Many thanks, Alfred

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  • 3CX behind UT7.1 using a callcentric.com SIP account

    - by Corey
    Has anyone had any luck with getting 3CX working behind UT7.1 with a SIP account from callcentric.com? I am willing to reset my current UT box back to defaults, and start from there. I have a static public IP assigned to the external interface. My internal addressing is 192.168.76.0 . My 3CX box has 192.168.76.17 . Would anyone be willing to give me a step by step of changes to make in UT / 3CX. I currently have my UT box unplugged, and have replaced it with a Linksys unit. I have port forwarding setup for… TCP/UDP 5060 to 192.168.76.17 UDP 9000-9049 to 192.168.76.17 … and everything works great. I also have additional external IPs available if that helps.

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  • Asterisk terminating outbound call when picked up, sends 'BYE' message

    - by vo
    I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway the setup does not appear to have issues registering and setting up the call, RTP packets go both ways and you can hear the ringing from the other side. However it appears that when the call is picked up or thereabouts, the incoming RTP packets cease. Upon closer inspection with Wireshark, there are these particular packets that seem to be the cause: trunk->asterisk SIP/SD Status: 200 OK, with session description asterisk->trunk SIP Request: ACK sip:<phone>@trunk:6889 asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 [..about a dozzen RTP packets in/outbound..] trunk->asterisk SIP Status: 200 OK, CSeq: 104 Bye [..outbound RTP continues, phone is silent..] Then the inbound RTP packets cease, however the asterisk logs dont show any activity at this point. The last entry reads 'SIP/ is answered SIP/'. Then when you hangup the extension, you get asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 trunk->asterisk SIP Status: 481 Call Leg/Transaction does not exist My trunk peer settings in FreePBX are: username=<user> fromuser=<user> canreinvite=no type=friend secret=<pass> qualify=no [qualify yes produces 401/forbidden messages] nat=yes insecure=very host=<sip trunk gateway> fromdomain=<sip trunk gateway> disallow=all context=from-pstn allow=ulaw dtmfmode=inband Under sip_general_custom.conf i have stunaddr=stun.xten.com externrefresh=120 localnet=192.168.1.1/255.255.255.0 nat=yes Whats causing Asterisk to prematurely end the call and still think the call is in progress? I have no idea where to look next.

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  • SIP and NAT routers?

    - by OverTheRainbow
    Hello SIP was not built with NAT routers in mind, and I'd like to get to the bottom of this issue to check what needs to be done on all devices so it works with NAT routers, and understand in what context it just can't be used and I should check more NAT-friendly alternatives like IAX. A picture being worth a thousand words, here's the layout I need to use: http://img62.imageshack.us/img62/4077/sipandnatrouters.jpg The PBX server is located in the private LAN behind a NAT router connected to the Internet (I know it'd be easier if it were located in the public network, but this router doesn't support DMZ's so the server has to be in the private network) A couple of (soft|hard)phones are located on the same LAN and connected to the PBX server, along with a PSTN gateway (Linksys 3102 or a Digium PCI card) Remote users using (soft|hard)phones are located somewhere on the Net with dynamic IP's and are also located behind NAT routers I may or may not have control over the local NAT router where the PBX server is located, but I have no control over the remote NAT routers, either because the users don't have the computer knowledge to map ports or because the routers are off-limit (eg. web cafés, hotel LAN's, etc.) Is it possible to configure the PBX server, the (soft|hard)phones, and the PSTN gateway so that the all conversations work fine, no matter the endpoints (POTS caller/local phone, POTS caller/remote phone, local phones, remote phone/local phone)? In which cases may I expect problems, and are there solutions? FWIW, I'm leaning toward using Freeswitch, but I could end up using Asterisk if there are technical advantages to it in this context. Thank you for any info.

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  • Why are SIP calls via my server silent?

    - by Archcode
    I have FreeSWITCH SIP server up and running. It has public IP and sits behind 1-to-1 NAT (it's Amazon EC2 instance actually). I can connect to it, make a call to other endpoint (namely, my android device to my pc and vice versa) and signals are send with no problems (call, answer, hangup, etc). Unfortunately, and what drives me crazy, that's all: no audio gets through, no video either. Server does not throw errors, it reports many retransmission though, looks like this: switch_rtp.c:915 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=15 Codecs are set up correctly (same config worked locally on my LAN). NAT/firewall on client side may be a problem, signals do get through (perhaps due to fixed port, data streaming runs on random one, that is currently my best bet). STUN/TURN/ICE setting on client seem to have no effect. Endpoints sit behind symmetric NAT. On server there are no iptables rules, security group is set as suggested there: http://wiki.freeswitch.org/wiki/Firewall Help, please. How to make it work or at least diagnose what's wrong?

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  • How to stop registration attempts on Asterisk

    - by Travesty3
    The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:50] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:55] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:55] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:57] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device <sip:[email protected]>;tag=cb23fe53 [2012-05-29 15:53:57] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device <sip:[email protected]>;tag=cb23fe53 [2012-05-29 15:54:02] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:54:03] NOTICE[5578] chan_sip.c: Registration from '<sip:[email protected]>' failed for '37.75.210.177' - No matching peer found [2012-05-29 21:20:36] NOTICE[5578] chan_sip.c: Registration from '"55435217"<sip:[email protected]>' failed for '65.218.221.180' - No matching peer found [2012-05-29 21:20:36] NOTICE[5578] chan_sip.c: Registration from '"1731687005"<sip:[email protected]>' failed for '65.218.221.180' - No matching peer found [2012-05-30 01:18:58] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=dEBcOzUysX [2012-05-30 01:18:58] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=9zUari4Mve [2012-05-30 01:19:00] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=sOYgI1ItQn [2012-05-30 01:19:02] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=2EGLTzZSEi [2012-05-30 01:19:04] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=j0JfZoPcur [2012-05-30 01:19:06] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=Ra0DFDKggt [2012-05-30 01:19:08] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=rR7q7aTHEz [2012-05-30 01:19:10] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=VHUMtOpIvU [2012-05-30 01:19:12] NOTICE[5578] chan_sip.c: Sending fake auth rejection for device "unknown" <sip:[email protected]>;tag=JxZUzBnPMW I use Asterisk for an automated phone system. The only thing it does is receives incoming calls and executes a Perl script. No outgoing calls, no incoming calls to an actual phone, no phones registered with Asterisk. It seems like there should be an easy way to block all unauthorized registration attempts, but I have struggled with this for a long time. It seems like there should be a more effective way to prevent these attempts from even getting far enough to reach my Asterisk logs. Some setting I could turn on/off that doesn't allow registration attempts at all or something. Is there any way to do this? Also, am I correct in assuming that the "Registration from ..." messages are likely people attempting to get access to my Asterisk server (probably to make calls on my account)? And what's the difference between those messages and the "Sending fake auth rejection ..." messages? Further detail: I know that the "Registration from ..." lines are intruders attempting to get access to my Asterisk server. With Fail2Ban set up, these IPs are banned after 5 attempts (for some reason, one got 6 attempts, but w/e). But I have no idea what the "Sending fake auth rejection ..." messages mean or how to stop these potential intrusion attempts. As far as I can tell, they have never been successful (haven't seen any weird charges on my bills or anything). Here's what I have done: Set up hardware firewall rules as shown below. Here, xx.xx.xx.xx is the IP address of the server, yy.yy.yy.yy is the IP address of our facility, and aa.aa.aa.aa, bb.bb.bb.bb, and cc.cc.cc.cc are the IP addresses that our VoIP provider uses. Theoretically, ports 10000-20000 should only be accessible by those three IPs.+-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ | Order | Source Ip | Protocol | Direction | Action | Destination Ip | Destination Port | +-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ | 1 | cc.cc.cc.cc/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 2 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 80 | | 3 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2749 | | 4 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 443 | | 5 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 53 | | 6 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1981 | | 7 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1991 | | 8 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2001 | | 9 | yy.yy.yy.yy/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 137-138 | | 10 | yy.yy.yy.yy/255.255.255.255 | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 139 | | 11 | yy.yy.yy.yy/255.255.255.255 | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 445 | | 14 | aa.aa.aa.aa/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 17 | bb.bb.bb.bb/255.255.255.255 | udp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 10000-20000 | | 18 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1971 | | 19 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 2739 | | 20 | any | tcp | inbound | permit | xx.xx.xx.xx/255.255.255.255 | 1023-1050 | | 21 | any | all | inbound | deny | any on server | 1-65535 | +-------+-----------------------------+----------+-----------+--------+-----------------------------+------------------+ Set up Fail2Ban. This is sort of working, but it's reactive instead of proactive, and doesn't seem to be blocking everything (like the "Sending fake auth rejection ..." messages). Set up rules in sip.conf to deny all except for my VoIP provider. Here is my sip.conf with almost all commented lines removed (to save space). Notice at the bottom is my attempt to deny all except for my VoIP provider:[general] context=default allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=g726 allow=ulaw allow=alaw allow=g726aal2 allow=adpcm allow=slin allow=lpc10 allow=speex allow=g726 insecure=invite alwaysauthreject=yes ;registertimeout=20 registerattempts=0 register = user:pass:[email protected]:5060/700 [mysipprovider] type=peer username=user fromuser=user secret=pass host=sip.mysipprovider.com fromdomain=sip.mysipprovider.com nat=no ;canreinvite=yes qualify=yes context=inbound-mysipprovider disallow=all allow=ulaw allow=alaw allow=gsm insecure=port,invite deny=0.0.0.0/0.0.0.0 permit=aa.aa.aa.aa/255.255.255.255 permit=bb.bb.bb.bb/255.255.255.255 permit=cc.cc.cc.cc/255.255.255.255

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  • Unable to call through asterisk

    - by sk
    I want to create a voip service. I have installed asterisk-1.4 on a dedicated remotely hosted debian lenny distro. I made a sip.conf and extensions.conf so as to place a call between two sip phones(i am using xlite 3.0) installed in some other Windows PC. Whenever i switch this phones the asterisk console shows that Registration from '"1000"<sip:[email protected]>' failed for '122.168.10.254' - Peer is not supposed to register Where xx.xx.xx.xx is the server's IP. i.e my sip phones are unable to register with the asterisk server. Please help me to place call between two sip phones #sip show peers Name/username Host Dyn Nat ACL Port Status 2000 (Unspecified) D 0 Unmonitored 1000 (Unspecified) D 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] # sip show registry Host Username Refresh State Reg.Time # sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 0 active SIP channels

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  • problems of making the sip with mingw32-make-3.81

    - by user160542
    This is the problem in the making progress: make[1]: Entering directory L:/1_????/3_Python/python_gui_programe/sip-4.8. 2/sipgen' makefile:29: warning: overriding commands for target .c.o' makefile:26: warning: ignoring old commands for target .c.o' gcc -c -O2 -w -DNDEBUG -DUNICODE -DQT_LARGEFILE_SUPPORT -I. -o main.o main.c process_begin: CreateProcess(NULL, gcc -c -O2 -w -DNDEBUG -DUNICODE -DQT_LARGEFI LE_SUPPORT -I. -o main.o main.c, ...) failed. make (e=2): ??????????? make[1]: *** [main.o] Error 2 make[1]: Leaving directory L:/1_????/3_Python/python_gui_programe/sip-4.8.2 /sipgen' make: *** [all] Error 2 I run the command "make" in the sip-4.8.2 directory followed the install guid(http://www.riverbankcomputing.com/static/Docs/sip4/installation.html#configuring); My platform is Windows Xp! Could somebody help me?

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  • Keep alive using SIP in .net

    - by Ramesh Soni
    I am creating an application where I need to implement SIP protocol in .NET. We have Client-Server setup where client keeps on sending keep alive message to server. We can only use SIP protocol or any other protocol which is support with ICE. Could some one help me in implementing this. I don't have much idea about these protocols but I know .net very well. Some sample code would be of great help.

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  • SIP communicator client

    - by Afro Genius
    I want to build a sip client based on SIP Communicator - the Java VoIP and Instant Messaging client. Basically I need to plug in some how and redirect VoIP to and from my application. Where is a good place to start? If this seems a bit vague, I do apologize.

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  • SIP servlets, chatserver

    - by Senne
    I'm trying to get a SIP servlet chat server working, together with the textclient found here. When I use 2 clients to send messages to eachother (peer to peer), everything goes well. But when I use one or more clients together with my server, I have to wait exactly 32 seconds before the server picks up any new messages in the doMessage() method. I'm using Netbeans together with Sailfin as my SIP server. Is there some kind of limitation or configurable delay or timeout between requests or responses in Sailfin I'm looking over? I can post the server code, if needed. Thanks

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  • How to notify SIP client when there is an incoming call on another phone

    - by fmunkert
    Hi, is it possible to notify a SIP client when there is an incoming call on another phone? I know that there are the SUBSCRIBE and NOTIFY commands but I have found no event package for signaling incoming calls. Background: for a SIP-capable telephony system, I would like to provide an application that displays information about the caller (e.g. name, address, contracts, etc.) when the phone rings. The phones are external to the PC; they are not soft-phones. -Frank

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  • Is there a SIP provider in the UK which provides the P-Asserted-Identity header?

    - by nbolton
    In the US, Flowroute (low cost SIP trunking provider) provides P-Asserted-Identity in the SIP invite request header (example screenshots). It also allows you to set the caller ID for outgoing calls, for example by using the follow in extensions.conf for Asterisk: exten => id,n,Set(CALLERID(all)=123) However, in the UK, I've tried a couple of SIP providers and none of them let me do either of those things (see P-Asserted-Identity or set the caller-ID). Is this because of some sort of restriction in the UK phone networks, or is it only available to really expensive SIP trunking providers?

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  • How can I stop SipVicious ('friendly-scanner') from flooding my SIP server?

    - by a1kmm
    I run an SIP server which listens on UDP port 5060, and needs to accept authenticated requests from the public Internet. The problem is that occasionally it gets picked up by people scanning for SIP servers to exploit, who then sit there all day trying to brute force the server. I use credentials that are long enough that this attack will never feasibly work, but it is annoying because it uses up a lot of bandwidth. I have tried setting up fail2ban to read the Asterisk log and ban IPs that do this with iptables, which stops Asterisk from seeing the incoming SIP REGISTER attempts after 10 failed attempts (which happens in well under a second at the rate of attacks I'm seeing). However, SipVicious derived scripts do not immediately stop sending after getting an ICMP Destination Host Unreachable - they keep hammering the connection with packets. The time until they stop is configurable, but unfortunately it seems that the attackers doing these types of brute force attacks generally set the timeout to be very high (attacks continue at a high rate for hours after fail2ban has stopped them from getting any SIP response back once they have seen initial confirmation of an SIP server). Is there a way to make it stop sending packets at my connection?

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  • How to set up a home SIP Server/Proxy for multi ring?

    - by zio
    I have a sip account which only allows one device to be registered. When i'm at home I want incoming calls to be able to ring on multiple devices. All of these devices are connected to the local network. I'm guessing the way to do this is using a local server/proxy that would allow multiple registrations which then forwards traffic to/from my sip provider. What a simple way to do this on either OS X, Ubuntu or using some low cost SIP router hardware?

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