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Search found 410 results on 17 pages for 'voip'.

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  • Communicate from PC in internet to remote GSM device

    - by Jakub Jezik
    I have several remote devices that use GSM modem and gather data. I need to communicate with those devices from a PC station located in a office using the dial up connection. Is there a way to do this WITHOUT an additional GSM modem connected to that PC? Is there some way to accomplish PC to remote GSM device communication using e.g. VoIP or similar technology? I want to avoid installing an additional GSM modem in the office and use some software solution instead.

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  • Running multipul lines through a server.

    - by Kevin Roberson
    I am looking to buy numbers in bulk on DIDx.net. After I purchase the numbers in a particular area code, I want to forward those numbers to other numbers that are outside of that area code. This way it will be seen as a local call versus long distance. I have the customers but I don't have the system I need. I have read about Asterisk, VOIP, SIP, and BYOH. But I have no clue what will be the best system for me. Does anyone have any idea what my next step should be when it comes to hardware and software? Or what type of operating system I should use? I basically want to set up a system like GoogleVoice & Phonebooth.

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  • Running multiple lines through a server.

    - by Kevin Roberson
    I am looking to buy numbers in bulk on DIDx.net. After I purchase the numbers in a particular area code, I want to forward those numbers to other numbers that are outside of that area code. This way it will be seen as a local call versus long distance. I have the customers but I don't have the system I need. I have read about Asterisk, VOIP, SIP, and BYOH. But I have no clue what will be the best system for me. Does anyone have any idea what my next step should be when it comes to hardware and software? Or what type of operating system I should use? I basically want to set up a system like GoogleVoice & Phonebooth.

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  • Does anyone know of a inexpensive NAT router that has the ability to limit access to the Internet to

    - by Corey
    Does anyone know of a inexpensive NAT router that has the ability to limit access to the Internet to a specific MAC address? I know the Linksys routers have a MAC filtering feature, but it is the opposite of what I need. It allows you to block access to a specific MAC address. I need something that will block all, but allow an exception. I'm dealing with some VOIP issues in my company's network, and I think the answer is to have a separate router on the network for my PBX to use. I want to make sure that other nodes are not allowed to access the Internet via this second router.

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  • Which handsets work with IP DECT base?

    - by waldrumpus
    My workplace uses an Ascom IP DECT base station (IPBS1), and Ascom handsets as well. We're looking to replace some broken handsets; the ones by Ascom, however, are rather expensive, and we're not altogether satisfied with their quality. I've been looking at handsets from other manufacturers, which are much less expensive - however, since I don't know anything about DECT, VOIP, etc, I don't know if they will work with the base. I've perused the base's manual, but found nothing on handset compatibility. How can I find out what kind of handset works with our base?

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  • Commercial SIP Trunking in mainland China [closed]

    - by Patrick
    Is there any regulation preventing the use/sale of SIP trunks in mainland China? I've set up and used commercial-grade SIP trunks in places where previously we would have used ISDN T1/E1 connections. Here in Shanghai I'm looking for a similar service, and while E1 30B+D services are readily available, every telecoms company we speak with says that SIP trunking is not available in China with re-sellers of both China Telecom and China Unicom. But no one seems to know why. It seems logical to me that SIP trunks are cheaper to operate than ISDN services given that the first mile transit can be run over already-existing Internet infrastructure, and SIP signaling reduces the amount of configuration required by subscribers which is why it appeals to me. As such I've come to expect SIP services to be available in modern markets, and I've used them in quite a few countries. For example, one place I know it's not possible is in India. Government regulations in India make it illegal to provide PSTN service using VoIP. (Citations: 1, 2). However it seems this may be changing. Perhaps China has something similar.

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  • What service do you use for music on hold?

    - by Russ Warren
    This may not be a sysadmin question for some, but it is definitely a hurdle I have to jump as the sysadmin for my company. We recently rolled-out a company wide VoiP system (Switchvox, to be exact) that has come preloaded with some royalty-free music on hold. Our customers have been complaining that the music on hold sounds like "funeral music." This may be the case (although I wouldn't want it played at my funeral), but it is all we have and we aren't willing to be sued over using music that isn't properly licensed. So, that brings me to the question asked in the title -- what and/or how do you provide decent music on hold? I'm assuming many people here use a PBX that allows customized music, so this has to apply to many of you. We've been looking at some sites that allow you to download royalty-free music for a one-time fee, but the music seems...lame. Something like a one-year subscription from ibaudio.com seems to be the best bet so far. Have you been able to discover something a little more mainstream for a decent licensing fee? Thank you. EDIT: Our PBX allows the playback of MP3 and OGG files, but does not allow streaming of a live audio source, Internet-based or otherwise. It also does not allow the use of a "line-in" source such as a CD player or radio. Don't let this stop you from sharing your setup, though. I'm interested in hearing what everyone uses!

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  • How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network

    - by Maharshi Raval
        I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want to use a dedicated broadband connection for the PBX (3CX) box, but the box also needs to be accessible in the local network as we will be using IP Phones and software IP phones. How configure two network cards on PBX box, so that one will be always used to connect to our SIP host over the Internet and the other will be connected to local network accessible from other client pc to connect to the pbx system. It must be noted that currently the Windows SBS 2011 acts as the Primary Domain Controller and gateway for all the client machines.     I cannot use a load balancer as it will conflict and cause issues within the current setup of our SBS2011 as it is also our Exchange Server. Any input is much appreciated. thanks in advance

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  • How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network

    - by Maharshi Raval
        I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want to use a dedicated broadband connection for the PBX (3CX) box, but the box also needs to be accessible in the local network as we will be using IP Phones and software IP phones. How configure two network cards on PBX box, so that one will be always used to connect to our SIP host over the Internet and the other will be connected to local network accessible from other client pc to connect to the pbx system. It must be noted that currently the Windows SBS 2011 acts as the Primary Domain Controller and gateway for all the client machines.     I cannot use a load balancer as it will conflict and cause issues within the current setup of our SBS2011 as it is also our Exchange Server. Any input is much appreciated. thanks in advance

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  • VoIP Tunnel Implementation for SIP Client

    - by Mahendra
    I am planning to provide an option for tunneling in my SIP client. I have tried to search on web for open-source implementation of this, but couldn't find one. My questions are: 1) If I go writing down my own custom code for implementing the feature - What are the different parameters / cases that I should consider & what should be my approach to start it? 2) Is there any open source implementation for SIP Tunneling already available? Any inputs are appreciated. Thanks.

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  • Android, phone call audio stream via wlan

    - by moppel
    I am planning on developing my specific voip app for android. Here's the scenario: when a phone call occurs I want to hear the person who's calling on my local pc speakers and I want to speak to him via my own pc microphone / headset. So I need to send the audio stream of both me and the person I am talking to via the wlan network. Something like this: ... onCallStateChanged(int state, String phoneNumber){ while(state == PhoneListener.CALL_STATE_OFFHOOK){ //while phone call is happaning //send incoming speech via wlan to pc //receive audiostream from pc microphone and direct it to the phone call } } ... Is this possible with the current Android API? (Actually it should be since voip apps are available in the market) I did some research in the Android API and all I found was the AudioManager which has constant named public static final int STREAM_VOICE_CALL; //The audio stream for phone calls But I don't know how to use it our how it should give me access to the actual audiostreams which I can send via network. How do I manage to do this? The connection would be realised by TCP sockets.

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  • Android AudioRecorder object wont read from microphone.

    - by supersk
    I'm trying to build a voip application on a new android device, i use AudioRecorder to read the microphone but I'm getting no valid results, just white noise. This happen only on this new device(other work well) and this is probably because this device has PTT ability. Is there some workaround to avoid using AudioRecoder to receive streaming data from the microphone? Thanks. supersk.

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  • 64-bit Windows 7 softphone to make SIP calls without registering with a SIP proxy?

    - by Dan J
    We have test tools that require us to call SIP addresses like localhost:5061. I used to use SJPhone on Windows XP, and an older version of X-lite, which both worked fine, and didn't require the SIP phone to be registered with a SIP proxy. I have just upgraded to Windows 7 and SJPhone doesn't seem to work any more (see forum here for others with the problem) - it says "No sound input device / No sound output device" at startup. I have tried a range of other softphones (X-lite 3, X-lite 4, Zoiper, 3CX), but I can't seem to find any that will install on Windows 7 and will let me call a SIP address like localhost:5061. It might be that I just don't know how to configure these phones to do it...

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  • Skype performance in IPSEC VPN

    - by dunxd
    I've been challenged to "improve Skype performance" for calls within my organisation. Having read the Skype IT Administrators Guide I am wondering whether we might have a performance issue where the Skype Clients in a call are all on our WAN. The call is initiated by a Skype Client at our head office, and terminated on a Skype Client in a remote office connected via IPSEC VPN. Where this happens, I assume the trafficfrom Client A (encrypted by Skype) goes to our ASA 5510, where it is furtehr encrypted, sent to the remote ASA 5505 decrypted, then passed to Client B which decrypts the Skype encryption. Would the call quality benefit if the traffic didn't go over the VPN, but instead only relied on Skype's encryption? I imagine I could achieve this by setting up a SOCKS5 proxy in our HQ DMZ for Skype traffic. Then the traffic goes from Client A to Proxy, over the Skype relay network, then arrives at Cisco ASA 5505 as any other internet traffic, and then to Client B. Is there likely to be any performance benefit in doing this? If so, is there a way to do it that doesn't require a proxy? Has anyone else tackled this?

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  • AsteriskNow Migration / Shared Extension Space

    - by Aaron C. de Bruyn
    I am testing the possibility of migrating from an old Avaya phone system to AsteriskNow. The migration would cover several hundred phones--but spread out over several years. (Management wants to move buildings to the new phone system one by one as cables get cut or time permits.) Two other directive is that extensions must not change and they want a GUI that other admins (non-Linux geeks) can manage. They currently use 9XXX for all extensions. We linked the Avaya and Asterisk box via PRI card and they both are communicating. From the Avaya side, if we move (for example) extension 9001 to Asterisk, we forward the call over the PRI to the AsteriskNow box and the SIP phone rings. In AsteriskNow we have an outgoing rule '_9XXX' that routes all 4-digit extensions starting with 9 back to Avaya. Here's the trouble. Dialing 9001 (the extension moved over to AsteriskNow) causes the call to be routed out the PRI to the Avaya box, then the Avaya box routes the call back to Asterisk, and Asterisk routes it to the SIP phone. As we get more and more users switched over, it will use up more and more channels over the PRI card. Is there a way I can ask Asterisk to check it's local extensions first--then forward off to the Avaya system if it starts with '_9XXX'? (I know how I can do it when editing the raw config files, I'm just looking for a way to do it in the GUI so other admins can manage it if necessary.) As a last-ditch plan, I know I can specifically add '_9001' as an outgoing call rule and sent it directly to extension 9001--but I'd really hate to do that for several hundred phones

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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they present a digest username value that doesn't match what Asterisk thinks it should. As an example, here are the relevant lines from a successful registration from a soft phone: Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cac3853" Phone responds: Authorization: Digest username="2321", realm="asterisk", nonce="1cac3853", uri="sip:192.168.254.12", algorithm=md5, response="d32df9ec719817282460e7c2625b6120" For the 3com phone, those same lines look like this (and fails): Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Phone responds: Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" Basically, Asterisk wants to see a username in the Digest username field of 2321, but the 3com phone is sending sip:[email protected]. Anyone know how to tell asterisk to accept this format of username in the digest authentication? Here is the sip.conf info for that extension: [2321] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=1234 qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes mailbox=2321@device host=dynamic dtmfmode=rfc2833 dial=SIP/2321 context=from-internal canreinvite=no callerid=device <2321 allow=ulaw, alaw call-limit=50 ... and for those interested in the grit, here is the debug output of the registration attempt: REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (no NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) confbridge*CLI REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 403 Authentication user name does not match account name Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) Thanks for your input!

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  • Bringing people into an Asterisk conference call

    - by Harley
    I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO is trying to do, but it doesn't work quite properly for me. Here's what happens: 1. Internal person A calls person B 2. Person A presses *0, he is given a dial tone and person B is taken to a conference room 3. Person A calls person C and they can talk, and then person A presses **. 4. Person C is brought to the conference room, but person A is disconnected. In the last step, A should be taken to the conference room as well. Here's the relevant logs, where 230 is person A, 231 is person B, 207 is person C, and 282 is the conference room.

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  • Exposing the ipPhone attribute to Communicator and the OCS address book service

    - by Doug Luxem
    I am in the process of integrating OCS with our Cisco phone system using CUCIMOC. After some fiddling with the phone normalization rules, it appears that I can get PSTN numbers to be dialed though the CUCIMOC interface (yay!). However, during this process I came to realize that the ipPhone attribute in Active Directory does not appear to be exposed to Communicator (and CUCIMOC). What is strange though, is that I can see from the OCS address book service "Invalid_AD_Phone_Numbers.txt" that the attribute is processed by the address book service. My question is, how do I expose the ipPhone field in Office Communicator? Currently, Communicator maps like this - Work = telephoneNumber Mobile = mobile Home = homePhone Attributes such as otherHomePhone, ipPhone, otherMobile, otherTelephone, otherIpPhone are ignored.

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  • Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

    - by slashp
    I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic (from Lync) & UDP traffic (to the trixbox, as Asterisk 1.4 does not support SIP over TCP). Our Lync box IP: 10.100.10.41 Our Kamailio box IP: 10.100.10.44 Our trixbox IP: 10.100.10.2 The issue I'm running into is as follows when enabling SIP debugging for the Kamailio box: <--- SIP read from 10.100.10.44:5060 ---> PRACK sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41 SIP/2.0 FROM: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430 TO: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e CSEQ: 24 PRACK CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b MAX-FORWARDS: 70 Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989 CONTACT: <sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41> CONTENT-LENGTH: 0 USER-AGENT: RTCC/4.0.0.0 MediationServer RAck: 1 23 INVITE <-------------> --- (12 headers 0 lines) --- Sending to 10.100.10.44 : 5060 (NAT) <--- Transmitting (NAT) to 10.100.10.44:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d;received=10.100.10.44 Via: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989 From: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430 To: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e Call-ID: 192daae6-00e1-4140-bddd-0394b35d475b CSeq: 24 PRACK User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> trixbox1*CLI> <--- SIP read from 10.100.10.44:5060 ---> ACK sip:[email protected];user=phone SIP/2.0 FROM: "John Jones"<sip:9121;[email protected];user=phone>;tag=4852bab430;epid=CF2380792B TO: <sip:[email protected];user=phone>;tag=3684a6a24e;epid=CF2380792B CSEQ: 23 ACK CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b MAX-FORWARDS: 70 Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK79a21c CONTENT-LENGTH: 0 My SIP trunk on the trixbox looks like this: [from-lync] exten => _+4XXX!,1,Noop(Stripping + from start of number) exten => _+4XXX!,n,Goto(from-internal,${EXTEN:1}) Though I am still having no luck getting the + stripped or the call to go through. Any ideas would be greatly appreciated. Thank you! -slashp

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  • Using an extension to block a caller

    - by Trewq
    I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I typically hang up. I think want to dial some number (like *23 or something) and I'd like the last number that was received to be put in a database. Any future call from that number will be directed to VM or a busy tone. I'd appreciate some pointers on how I'd go about doing this.. I prefer an open source solution.

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  • Call issue with Freeswitch

    - by gbraad
    I am testing the following with Freeswitch and different devices (nokia n900, nokia e60, ekiga) and have similar results between them. On the Freeswitch server (1.0.4 in multi-tenant mode) I have several user profiles for a domain, e.g. 1000, 1001 for host.com The user are authenticated correctly and calls can be placede. When I place a call from a device registered as [email protected] to [email protected] it will show up at the other end (1002) as [email protected] I would expect this call to show up as [email protected]. The IP address is the one of from the Freeswitch server. Because of this, the calls are no correctly recognized by the address book on certain devices. Can the he domain FQDN of the callers domain/acount be used, instead of the IP address of the server in the SIP uri?

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  • Considerations for a business looking to transition from PSTN to IP Telephony

    - by Bryce Thomas
    Full disclosure - This is related to a homework assignment question. I am not asking you to do my work for me, I am merely looking for some pointers and considerations to direct me in my further research. I have an assignment I'm working on where I've been given a scenario where a business wants to look into transitioning to using "Internet Telephone" as opposed to a traditional PSTN/PBX system and I need to write a report on it. I'm after some high level pointers from people, especially anyone that has been involved in a real life transition of this nature, on what some of the most important considerations are. These can be financial considerations, initial setup considerations, ongoing administrative considerations, quality of service considerations or anything else that is pertinent to performing such a transition.

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