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  • Skype Connect as SIP/Trunk for Asterisk

    - by Kaurin
    First off: I'm not sure if this should be on superuser or here. I have recently built a few Asterisk boxes with OpenVOX FXO/FXS ports little or no trouble. My current project is building an Asterisk box with SIP trunks. My current employer insisted on getting Skype Business/Skype connect for that purpose. After reviewing the Skype Connect plan, I agreed, because I thought it is going to be straightforward: Purchase G729 licences and setup SIP trunk/trunks. Boy was i wrong :) Here is the setup: The setup is for calling US numbers only via skype (we got skype US minute bundles in skype connect) AsteriskNOW - Asterisk 1.4 + asterisk-gui Trunks: SIP Trunk configured with Skype Connect - shows as registered Users: 2 test extensions. Both work fine when calling each other, voicemail etc works fine too The asterisk box is behind a Mikrotik router which i configured to forward all relevant ports: 5060-5090 UDP, 10000-20000 UDP. When trying out an extension outside of my LAN, it worked. I could make calls to the other extension. Outgoing rule: _NXXXXXXXXX Strip:0 Prepend:+1 Use skype trunk Inbound rule: Trunk: Skype Pattern: s Destination: Extension1 (6210) Here is the output of asterisk CLI (-rvvvvv) with outgoing calls: http://pastebin.com/eWVpL72e you can see the circuit-busy response when using trunk1 (skype) When calling my Skype Connect number from the outside, I get nothing in the logs. Can anyone with Skype Connect / Asterisk experience help out? :)

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  • Reliable applicance for routing IT emergency calls (SIP and ISDN)

    - by chiborg
    We have a fairly big IT installation and our IT staff needs to be reachable 24/7. At the moment we have the following setup for "emergency" calls to our IT staff on our main Asterisk box: An incoming emergency number (connected via SIP trunk and a BRI card in case the SIP trunk goes down). When the number is called during the office hours, all the SIP phones of the IT staff are called simultaneously. When the number is called out of office hours interface, a list of mobile phone numbers is called, one after another until someone picks up. The list can be changed by the IT staff via command line script. The setup works well, but the Asterisk is heavily used in a call center, has experienced some outages and misconfigurations, each of them bringing down the IT emergency number. So we'd like to put the IT emergency call functionality on a separate device. This does not need to be a big server, it even does not need to be Asterisk, it only has one purpose and should do it reliably. It should be very low-maintenance. Any suggestions for hard- and software?

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  • Test-service on Internet for testing incoming INVITE

    - by leiflundgren
    I am trying to set up Asterisk at home. I think I'm having trouble configuring my firewall, so that inbound traffic is accepted, but I am not sure. I got the idea that, perhaps, there is a service out on the Internet, where I can, though a web-browser, initiate an incoming call, an INVITE. And then see the SIP-trace that the remote-part experience. Anyone know of such a service? Note. I have a SIP-PSTN provider so I can generate inbound calls. But I cannot see the SIP-logs from my provider...

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  • How to install PyQt on Mac OS X 10.6

    - by Albert
    I want to install PyQt. This seems kind of complicated to install on OS X. I haven't found any precompiled packages of it (are there any? I would really prefer those). So I downloaded PyQt. And SIP, because it depends on that. These files: http://www.riverbankcomputing.co.uk/static/Downloads/PyQt4/PyQt-mac-gpl-4.7.3.tar.gz http://www.riverbankcomputing.co.uk/static/Downloads/sip4/sip-4.10.2.tar.gz Did a python configure.py && make && sudo make install on SIP -- installed without any problems. Tried the same on PyQt -- and failed of course: /Library/Frameworks/QtCore.framework/Headers/qglobal.h:288:2: error: #error "You are building a 64-bit application, but using a 32-bit version of Qt. Check your build configuration." Ok, so I tried with python configure.py --use-arch=i386. Same error. Any idea?

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  • How do I activate my gizmo5 phone number in Google Voice? [closed]

    - by Sorin Sbarnea
    I wasn't able to activate my gizmo5 number because Google Voice activation(verification) requires you to enter two dial tones (DTMF) and they did not work at least not with these two variants: Using gizmo5 PC client using fring from Iphone as gizmo5 SIP client Redirecting gizmo5 to a US mobile number None of the above methods worked for me. Any ideas? More info: http://www.google.com/support/forum/p/voice/thread?tid=1d8c1d99721e3509&hl=en http://googlevoices.blogspot.com/2009/04/forwarding-sip-calls-to-google-voice.html

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  • asterisk extensions.conf & sip.conf

    - by Josh
    I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below. ; PSTN on sip.conf [pstn] type=friend host=dynamic context=pstn username=pstn secret=password nat=yes canreinvite=no dtmfmode=rfc2833 qualify=yes insecure=port,invite disallow=all allow=ulaw ; PSTN on extensions.conf [pstn] exten => s,1,Answer exten => s,2,Wait,2 exten => s,4,DigitTimeout,5 exten => s,5,ResponseTimeout,10 exten => s,6,Background(thanks-calling) exten => 0,1,Goto(incoming,123,1) ; (Member Services) [incoming] exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console exten => 123,n,Ringing() exten => 123,n,Answer() exten => 123,n,Playback(silence/1) exten => 123,n,Playback(connecting1) exten => 123,n,Wait(3) exten => 123,n,Dial(SIP/line1,60) exten => 123,n,Congestion

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  • asterisk extensions.conf & sip.conf

    - by Josh
    I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below. ; PSTN on sip.conf [pstn] type=friend host=dynamic context=pstn username=pstn secret=password nat=yes canreinvite=no dtmfmode=rfc2833 qualify=yes insecure=port,invite disallow=all allow=ulaw ; PSTN on extensions.conf [pstn] exten => s,1,Answer exten => s,2,Wait,2 exten => s,4,DigitTimeout,5 exten => s,5,ResponseTimeout,10 exten => s,6,Background(thanks-calling) exten => 0,1,Goto(incoming,123,1) ; (Member Services) [incoming] exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console exten => 123,n,Ringing() exten => 123,n,Answer() exten => 123,n,Playback(silence/1) exten => 123,n,Playback(connecting1) exten => 123,n,Wait(3) exten => 123,n,Dial(SIP/line1,60) exten => 123,n,Congestion

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  • Can i have a Asterisk IP PBX Server Behind ISA 2000

    - by garyb32234234
    Hello Is it a simple procedure to configure ISA Server 2000 to allow an Asterisk IPPBX connect to SIP provider. On asterisk forums they say the ISA has difficulties handling SIP, softphones that i have installed behind the firewall work fine with the provider when the firewall client is installed on the workstation. With asterisk being a linux based system this will not be an option. Is the config a matter setting up port forwarding, is this a more complicated task on ISA server than just selecting the ports i need and then the ip of the internal machine i want to forward them to? UPDATE: I dont think this is possible from what ive researched Regards Gary

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  • VoIP - need setup ip network with our SIP operator

    - by evgeniy.labusnkiy
    Need to make next one: for ex i'm now in UAE, but i need to make a call to my girlfriend who is in Ukraine. I need to find the way how can i make the connection to my home router and make the call from VoIP gateway using my standard phone network in Ukraine. I have some imagination about this, to do like this: Connect to router or VoIP gateway (how? soft?) using inet - Gateway make a call using standard phone line in my country. Any ideas? Best practice? What devices i need to make this? Pay attention that i don't want to use any SIP providers. Thats a lot!

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  • Linux QoS (Skype / BitTorent / SIP / HTTP priority)

    - by Andre
    We are configuring a linux box that will act as internet gateway for an office of 30-50 computers. We are using iptables/HTB for traffic shaping. Is there a way to match traffic on L7 level? It's easy to identify traffic by TCP/UDP ports (like SIP and HTTP). But what if we are dealing with Skype & BitTorent? It was surprise for me that there is no powerful and matured sulution for tasks like this. I found only l7-filter (http://l7-filter.clearfoundation.com/) patch for the Linux kernel, but it's no longer supported (it seems to). Moreover it couldn't be compiled with modern Linux kernels. The only option I found was to use a Cisco router. Are there other ways to identify and shape Skype and Bittorent traffic?

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  • DrayTek 2820 configuration using public IP addresses

    - by Kev
    I have a /29 range of public IP addresses assigned to me by my ISP. I'm trying to configure a SIP VOIP handset to register with my VOIP provider who recommend using public IP addresses rather than NAT. I have a DrayTek 2820 router flashed with the latest firmware and have configured my router as per DrayTek's FAQ at: How do I use a public subnet on the LAN (non-NAT operation ) ? My IP range is: xx.xx.94.16 -> xx.xx.94.23 This gives a usable range of: xx.xx.94.17 -> xx.xx.94.22 My router's public IP address is: xx.xx.94.17, the SIP VOIP handset is allocated xx.xx.94.18. I have a second internet connection and via that I can ping the handset. However for some reason I can't seem to get it to register with the provider. I tried adding a new Firewall filter to pass through from WAN to LAN: Source: ANY, Destination: xx.xx.94.18, UDP - Ports 1024 -> 65535 Out of interest I also tried opening port 80 to see if I could browse to the phone's admin web interface but no joy. I know that my ISP aren't blocking inbound service ports because I NAT Port Forwarded port 80 to one of my internal web servers and it rendered a test page I had set up. All the NAT settings are reset to factory defaults, i.e. there are no Port Redirection, DMZ Host, Open Ports or Address Mappings configured. The handset I'm using is a GrandStream GXP-2000. Is there anything else I should be doing?

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  • local RTP port unreachable when using mjsip/jmf

    - by brian_d
    Hello, I create a sip session with mjsip to an external voip provider. Then I transmit a test wav file over rtp to the provider using RtpManager. The program runs with no errors and I answer the sip call. However, no audio is transmitted. When I diagnose the network traffic with wireshark, I see a bunch of RTP traffic from my localhost (behind some kind of nat) to the voip provider and nothing back. After a while I get the ICMP error "Destination unreachable (Port unreachable)" from the provider to my localhost. The software linphone works using the same localhost and voip provider - though it is using a different sip stack. Any suggestions? Thanks

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  • Direct VoIP call from one iOS device to another

    - by user1682856
    Could you please give some advises. I'am going to develop peer-to-peer VoIP iOS application. And want do it without any SIP proxy, SIP providers and other servers. Just VoIP calls frpm iOSdevice-to-iOSdevice. Both iOSdevice could be somewhere in Internet. Is it real in VoIP (with PJSIP for example and general with SIP)? Could you please point me to main keys that I need for development. I'am already read these topics. Is it real solve problems with addressing in my configuration. Anybody know is PJSIP could help with correcting addresing.

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  • Can I set up an "involuntary" conference call with Freeswitch?

    - by Atilla Filiz
    I am trying to set up a SIP/RTP public announcement infrastructure. Basically there are several slave user agents that are configured to answer automatically, and a master UA which should be able to call all of them and make announcements. A way to work around seems creating a conference and making all UAs to join via some RPC mechanism but I don't want to go that direction unless I have to. The slave UAs are linphone and I haven't decided on the master agent yet.

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  • Audio/video streaming on Windows platform

    - by bushtucker
    I'm building an interactive language learning application to be used in a classroom environment. The idea is that a teacher should be able to talk to the students (=audio stream to all students), let students talk to each other (= audio P2P) in groups of two or more, let students watch a video coming from a the DVD player or coming from a media server. It should be possible to save the audio/video streams. The teacher should also be able to monitor, take-over or block the desktop of the students. The platform is Windows and it's a desktop application, no web application. The audio delay should be as minimal as poosible. Optionally a student sitting at home should be supported, but it's not a high priority. I am now finished with the classroom control part of the application (login, monitor, block, ...) and want to start the audio and video part. I've been evaluating several options like DirectX, GStreamer and SIP but now I have to make a decision. DirectX seems an obvious choice for the Windows platform, but it only lets me capture and playback audio and video. The encoding/decoding/network part I should do myself. GStreamer contains all kinds of options to capture/encode/stream/save audio and video streams. I've experimented a bit with it (ossbuild) and it does seem to involve a lot of trial and error to make something work: - microphone capture (via directsoundsrc) produces cracking noises on some computers - rtpL16 payloader didn't work well - streaming raw audio over the network only working at a sampling rate of 8000, no higher - there are a lot of errors when receiving mpeg4 video (bad I-frame), on some computers worse than others It is my impression that gstreamer is primary targetted at linux platforms. Development and support for the Windows platform seems to be a little behind. Nevertheless it's a powerful framework that could save me months and years of work. SIP seems to be able to do everything I want, but it is targeted towards telephony and IM. I don't know how flexible SIP is. It seems to me that the SIP layer would just be overhead as I already have a central (teacher) application that can control and setup all the streams. The interesting parts of frameworks like opalvoip and freeswitch are the actual audio/video capture, the encoding and transmission. Does anyone know how these interesting parts relate a framework like gstreamer? Are they easy to integrate into a custom application? Are they flexible enough? Does anyone have experience with all or one of these technologies? Maybe there are even other options I can look at? Many thanks for your advice

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  • Location of global libraries for Python on Mac ?

    - by xTrol
    Hi, Im fighting with installation SIP for Python on Mac OS X. Finally after compilation and installation when I run console form folder of SIP (locally) I can import sipconfig, but when Im in other folder I cant - there is no module called sipconfig. My question is - Where is folder to which I have to copy modules if I want to have them available globally (like "import os"), or how I can check it, because location "/Library/Python/2.6/site-packages/" doesn`t work.

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  • Asterisk, IAXModem & Hylafax how-tos?

    - by Brian Postow
    I'm trying to set up Asterisk and IAXModem to send faxes via T38 (Yes, I know I'm swatting a fly with a Buick...) However, since I'm trying to do something so small with a product so large, I'm having trouble finding samples or how-tos that show me how to set this up. I've got all three installed, and I THINK I have my IAXModem config correct. I'm pretty sure that I have Hylafax correct (I've used it with T38Modem) so, I need to know which of the Asterisk samples I need to use, and how to use them. I think I want to use some combination of iax.conf, iaxprov.conf, sip.conf and sip_notify.conf. But I'm not sure where to put them, or what to change... I'm sure that the answer is RTFM, but I'm not sure WHICH M, or where in it to R... thanks.

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  • How does the caller ID works on voip?

    - by Cawas
    Maybe this depends on the protocol, if so, I'm wondering mostly about SIP. That's all, just focus on the title please, but for a little background... I am using voipraider and I was just trying to set it up to have a caller id as my phone number, since I can't have this voip of mine on all the time (thus using DID or being able to receive calls through voip wouldn't be a solution here). I could actually make it work, but only using voipraider software. From other places, the caller ID doesn't show properly. So, I am wondering how it actually works to be able to go and look for a fix for this. I want details about the protocol, if that's relevant.

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