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Search found 316 results on 13 pages for 'sip'.

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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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  • Android as SIP to GSM gateway

    - by user346034
    Soon I could use a SIP to GSM gateway, because I'll need to make phone calls from Germany to a mobile phone in Czech Republik. Hence, I thought about implementing one. Now, the questions are: Does such a solution already exist (for a reasonable price)? Is it possible to redirect a (voice) stream to a GSM connection with the available Android APIs (SDK or NDK)? Ideas, suggestions, comments are highly welcome.

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  • Looking for good SIP Book

    - by Dave
    Hey guys I am looking for a SIP book similar to this one on XMPP - Professional XMPP Programming with Javascript and Jquery (http://www.amazon.com/Professional-Programming-JavaScript-jQuery-Programmer/dp/0470540710) I am new to the area and any resources would be appreciated, thanks

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  • Videoconference using Flash and SIP

    - by Júlio Santos
    The front-end will be Flash, to run in a browser and have access to the camera. I must use SIP to control the sessions. How could I do this? Will a Red5 server and a MjSip sever do the trick? As in i'd use MjSip to setup the session and warn users about calls, and Red5 to stream the video and audio? Any suggestions? Note: only 1-on-1 conference is required.

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  • router gets disconnected once I terminate my SIP application

    - by TacB0sS
    Hey, Here is an interesting one, I have a SIP VoIP application which is able to register to the PBX server, and I can invite and see the user call on the callee end receiving an Invite, and on the caller end I see the Ringing response... now here is interesting part, if I close my application with out any notification to the server my router disconnects and restart, after a short while (30 - 150 sec). I could fix that if I would complete the ACK BYE process, but I'm just wondering why does my router hangs up? any ideas?

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  • How to establish SIP connection, when SIP-proxy is required?

    - by LA_
    I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. I should use the following data to connect to the server: Auth name – 7499952XXXX User name/User ID/Display Name – nickname Authorization user name - [email protected] Domain - sip.beeline.ru SIP proxy server - msk.sip.beeline.ru I've also found the following string: [email protected]:password:[email protected]@msk.sip.beeline.ru:5060/7499952XXXX I've tested the parameters on my PC thru X-Lite and it works well (so, assume there is no any problem with the router, no need to do anything with router's NAS settings). But since I am quite new to Asterisk, I can not understand where to input all these data. Asterisk GUI doesn't have fields for proxy: Can somebody please help me with step-by-step instruction? Thank you in advance!

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  • How to establish SIP connection, when SIP-proxy is required?

    - by LA_
    I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. I should use the following data to connect to the server: Auth name – 7499952XXXX User name/User ID/Display Name – nickname Authorization user name - [email protected] Domain - sip.beeline.ru SIP proxy server - msk.sip.beeline.ru I've also found the following string: [email protected]:password:[email protected]@msk.sip.beeline.ru:5060/7499952XXXX I've tested the parameters on my PC thru X-Lite and it works well (so, assume there is no any problem with the router, no need to do anything with router's NAS settings). But since I am quite new to Asterisk, I can not understand where to input all these data. Asterisk GUI doesn't have fields for proxy: Can somebody please help me with step-by-step instruction? Thank you in advance!

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  • How compliant is SIP VoIP software on the net?

    - by rusbi
    I developed a SIP stack for my company. It's far from perfect, a it's lacking a lot of things from the RFCs, but it's functional and work well with a lot of tested softphones and other SIP hardware and software. My question is: How much of SIP software can truly say that they are entirely SIP compliant? (Of the softphones you can find on the internet...)

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  • odd response after INVITE request, SIP

    - by supersk
    After sending an invite request i receive a trying answer, and immidietly after that i receive error 407 proxy authentication required. After sending ack & another invite with the proxy header i receive session progress about 1/4 of the time!! other times it just sends 407 error again & again. Any ideas?

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  • How to split registration and media?

    - by Stackfan
    I have a SIP project. Where i will have SIP server running. Server will do following: will only do routing and receive incoming calls But the audio/video will be peer 2 peer Can this be done with Asterisk? Only the media i have to split but the registration will be with Server. Tools: A) server with SIP B) One PC with SIP client C) Anoher PC with SIP client My goal is: B and C gets connected via A and audio/video packets are not via A

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  • Oracle Coherence?UCOM?IP???????SIP?????????????

    - by Norihito Yachita
    ?????????Oracle Coherence???????????????????????UCOM?IP???????SIP(Session Initiation Protocol)?????????????????????????????????? UCOM?????????????????????????????????????????IP????????????????????????????????????ISP?????????????????????????????????????????????????????????????????????????????????????????SOHO??????????????????????? UCOM??IP???????????????????SIP??????????????????????????????IP???????????????????????????Oracle Coherence??2011?2????????????????????????????????????????????????????????????????????????????????????????·????????2????????? 11?30?(?)??????????·??????·????? 2011?(??:??????)??UCOM??Oracle Coherence?????????????????????:?UCOM ????????????????????????

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  • How to install SIP+PyQt with apt-get + pip + virtualenv?

    - by kjo
    [I originally posted this question, under a different title, in StackOverflow (here), but later I realized that my problem is very specific to apt-get, hence I am re-posting it here. Sorry for the duplication.] I'm trying to install PyQt on Ubuntu (and within a virtualenv). The list of obstacles I'm dealing with is far too long to include here, but the one I'm currently trying to get past is this: % workon myvenv (myvenv)% cd ~/.virtualenvs/myvenv/build/pyqt (myvenv)% python ./configure.py Traceback (most recent call last): File "./configure.py", line 32, in <module> import sipconfig OK, so let's install sipconfig... (myvenv)% pip install SIP Downloading/unpacking SIP Downloading sip-4.14.8-snapshot-02bdf6cc32c1.zip (848Kb): 848Kb downloaded Running setup.py egg_info for package SIP Traceback (most recent call last): File "<string>", line 14, in <module> IOError: [Errno 2] No such file or directory: '/home/yt/.virtualenvs/myvenv/build/SIP/setup.py' Complete output from command python setup.py egg_info: Traceback (most recent call last): File "<string>", line 14, in <module> IOError: [Errno 2] No such file or directory: '/home/yt/.virtualenvs/myvenv/build/SIP/setup.py' ---------------------------------------- Command python setup.py egg_info failed with error code 1 in /home/yt/.virtualenvs/myvenv/build/SIP Storing complete log in /home/yt/.pip/pip.log The only recipe I've found so far installing SIP is this % python configure.py % make % sudo make install ...but this recipe goes against my policy of doing all my Ubuntu installations either through apt-get (or through pip in the case of Python modules). Is there some way that I can install SIP with apt-get (and possibly pip)?

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  • What's the most advanced SIP client for Linux these days?

    - by Stefan Armbruster
    I'm currently struggling which SIP client software to use with respect to Ubuntu / Gnome. Some clients I've looked so far: Blink, seems promising but the Linux variant lacks a lot of features Twinkle Latest release is ~2 years old. AFAIK the only one capable of encrypting calls using zrtp. Empathy: default tool for IM on ubuntu Ekiga Some features I'd like to see: availablity of buddies conference calls call log chat desktop sharing (Blink seems to do that for Mac) So my question is: what client software do you prefer and for what reason?

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  • What's the most advanced SIP client for Linux these days?

    - by Stefan
    I'm currently struggling which SIP client software to use with respect to Ubuntu / Gnome. Some clients I've looked so far: Blink, seems promising but the Linux variant lacks a lot of features Twinkle Latest release is ~2 years old. AFAIK the only one capable of encrypting calls using zrtp. Empathy: default tool for IM on ubuntu Ekiga Some features I'd like to see: availablity of buddies conference calls call log chat desktop sharing (Blink seems to do that for Mac) So my question is: what client software do you prefer and for what reason?

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  • SIP and Java, where to start and with what?

    - by Senne
    I want to implement the SIP protocol in java and would want to be able to create different clients (5 or more) and make them connect to a proxy server. This is all for testing purposes so I would like to be able to see well what's happening on a rather low level. The clients should first be able to communicate trough text and later on maybe also by audio. (If I ever get that far) I already read a bit about the JAIN libraries and what I understood from that is that they are not really well suited for the server side? I also didn't really find any proxy server examples, tutorials, using JAIN. I also found this SIP Servlet Tutorial book, I used HTTP servlets in the past but should I prefer servlets or JAIN or ...? I'm quite new to SIP so I don't really know where to start or what to choose in combination with java.

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  • How wrong is it to modify the SDP body of a SIP message?

    - by rusbi
    A requirement for the SIP PBX I created for my company was to record all calls passing through it. I solved it by forcing all SIP message to pass through the PBX and to modify the SDP body so the stream passes through it and gets recorded. It works well. I recently found out that this is not allowed. Is there any other way to implement call recording and how "wrong" is this in regard to the protocol?

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  • asterisk/freeswitch in nat/no-nat setup

    - by pQd
    hi, my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party voip providers [ since it's in different countries - those are different providers, but that's irrelevant ]. i was thinking about terminating sip calls on something like asterisk/freeswitch server and having all sip-devices log on just once to such server[s] - mostly to provide things like voicemail, groupcalls, redirections etc. it seems perfectly doable but there is one problem - i cannot find examples how to prepare for nat/no nat. for calls routed to from/to 3rd party voip operator - i'll need handling for nat/stun etc, but for handling of internal calls - i do not want any nat, all traffic should go via vpns to different branches. can you provide me some hints how to configure it? any tutorials? thanks!

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  • T38modem manuals?

    - by Brian Postow
    Are there any T38modem users here? I'm trying to figure out how to call T38modem with SIP. I've got everything except the --route option for receiving. I know my own phone number, but I am not sure how to set it up. Currently, I have: --route "modem:0.*"="sip:<dn>@64.136.174.30" --route "modem:1.*=sip:[email protected]" I've also tried: --route "modem:*.*"="sip:<dn>@64.136.174.30" --route"sip:*@74.94.184.154"="modem:<dn>@127.0.0.1" what amI doing wrong? And where (other than t38modem --help) can I find some documentation on how to use it? thanks.

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  • Fortigate restrict traffic through one external IP

    - by Tom O'Connor
    I've got a fortigate 400A at a client's site. They've got a /26 from British Telecom, and we're using 4 of those IPs as a NAT Pool. Is there a way to say that traffic from 172.18.4.40-45 can only ever come out of (and hence go back into) x.x.x.140 as the external IP? We're having some problems with SIP which looks like it's coming out of one, and trying to go back into another. I tried enabling asymmetric routing, didn't work. I tried setting a VIP, but even when I did that, it didn't appear to do anything. Any ideas? I can probably post some firewall snippets if need be.. Tell me what you want to see. SIP ALG config system settings set sip-helper disable set sip-nat-trace disable set sip-tcp-port 5061 set sip-udp-port 5061 set multicast-forward enable end Interesting Sidenote VoIP phones, with no special configuration can register fine to proxy.sipgate.co.uk, which has an IP address of 217.10.79.16. Which is cool. Two phones are using a different provider, whose proxy IP address is 178.255.x.x. These phones can register for outbound, but inbound INVITEs never make it to the phone. Is it possible that the Fortigate is having trouble with 178.255.x.x as it's got a 255 in it? Or am I just imagining things?

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  • not able to register sip user on red5server, using red5phone

    - by sunil221
    I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = ** ip = asteriskserverip and i got --- Registering contact -- sip:[email protected]:5072 the right contact could be --- sip :99999@asteriskserverip this is the log: + SipUserAgent - listen - Init... Red5SIP register [SIPUser] register RegisterAgent: Registering contact (it expires in 3600 secs) RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout RegisterAgent: Failed Registration stop try. Red5SIP Client leaving app 1 Red5SIP Client closing client 35C1B495-E084-1651-0C40-559437CAC7E1 Release ports: sip port 5072 audio port 3002 Release port number:5072 Release port number:3002 [SIPUser] close1 [SIPUser] hangup [SIPUser] closeStreams RTMPUser stopStream [SIPUser] unregister RegisterAgent: Unregistering contact SipUserAgent - hangup - Init... SipUserAgent - closeMediaApplication - Init... [SIPUser] provider.halt RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout

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  • can't register a soft phone to asterisk11

    - by Tom
    I have a VM (on oracle vbox) running Fedora17. I've installed asterisk 11 on it from sources. I've followed the wiki for installation (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) to the letter. The ip on the VM machine running fedora is 192.168.1.7 and I can ping it from the host machine (Ubuntu 12.04), which is at 192.168.1.2 I've tried registering with ekiga with the following settings: user: [email protected]. Password: verysecretpassword registar: 192.168.1.7 but I'm getting an error "transport fail". Also, while trying to register I'm logged in to the asterisk CLI with verbose level 3 and debug level 4 and nothing appears. some more relevant data: I've added the following code to the end of my sip.conf.sample file: [demo-alice] type=friend host=dynamic secret=verysecretpassword context=users deny=0.0.0.0/0 permit=192.168.1.0/255.255.255.0 [demo-bob] type=friend host=dynamic secret=othersecretpassword context=users deny=0.0.0.0/0 permit=192.168.1.0/255.255.255.0 After I changed the sip.conf.sample file, I've created a copy of it and named it sip.conf. then I logged in to the asterisk CLI and typed sip reload. Then I'm trying to register and ekiga client from my host machine at 192.168.1.2 but it doesn't work and nothing appears on the asterisk CLI while in verbose mode level 3. BTW, If there is missing information about my question, please don't close it. comment about what you need to know and I'll edit it in to the question. tnx.

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