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

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  • deploying skype for 50 nationwide users with preset usernames

    - by kevyn
    Is there a way to sign up a large group of skype names at once? is there a way to enable the users to be given a skype username based on their own e-mail addresses? What I would like to do is roll out skype in an office in every county in UK with a pre defined username such as 'mycompanyname-warwickshire', 'mycompanyname-bedfordshire' and so on. Our users are only basic computer users, so I would ideally like this done with as least fuss as possible for them! Thanks in advance ps. if anyone has a good way of doing this by using any alternative software, I'm open to suggestions

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  • Gizmo5 mobile with wifi and callin number

    - by Mark
    I want to buy a Nokia 5530 and run Gizmo5 on it in order to receive and make calls. I want to get a call in number based in England and was wondering if receiving calls via the call in number on my mobile via wifi would use any of my callout credit? I ask because the Skype lite version uses your callout credit if you receive a call on it using your online number (it doesn't if you use the desktop version). Thanks!

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  • Checking rtp stream audio quality.

    - by chills42
    We are working in a test environment and need to monitor the audio quality of an rtp stream that is being captured using tshark. Right now we are able to capture the audio and access the file through wireshark, but we would like to find a way to save the audio to a .wav file (or similar) via the command line. Does anyone know of a tool that can do this?

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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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  • NBX v3001 new phone account: all incoming calls directly diverted to voicemail

    - by Pitto
    That's my problem: I've added a new user in our NBX v3001 3com (now hp) and all the incoming calls are diverted, without ringing, to the voicemail. The phone is a Sip Endpoint Terminal and I really don't have a clue why this is happening since I've installed other identical phones without troubles. I've also tried deleting and re-adding the user and changing the extension: nothing. Any hints to solve?

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  • Any good phone system on a Windows Environment ?

    - by Kedare
    Hello, I am looking for a phone system that integrate well with a (almost) 100% Microsoft. I would like something that can integrate with Exchange and Active Directory (in-phone searching/calling by name (is this possible using SIP ?), etc) and if possible something not too expensive (Bye bye Cisco !), what do you recommend me ? I've heard of 3CX as IPBX and Aastra as Phones, are they good for this ? Or do you know something else good at this ? I've also seen some Alcatel IP Phones doing this (search by name, but I think that was a separated address book and not loaded from AD/Exchange) Thank you !

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  • What are some potential uses of Asterisk (PBX) for a "power user"

    - by user144182
    So I read a lot of good things about Asterisk. I am not however looking to run a call center or small business setup. I am still interested what potential uses it has for me as a "power user" and what features I could harness for my communication needs. I'll throw out that I currently use other technologies like Google Voice, Skype, and a cellphone of course. So, what potential uses, if any, could Asterisk PBX have for a user like me?

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  • Report on how many received calls on Cisco unified CM

    - by Robert K.
    I've been struggling with this a couple of days now and I feel like I've seen every webpage about Cisco Unity CM but just can't figure it out... The request I got sounded fairly simple: We want to know how many calls a given number (for example 987) receiver in the month July. Is there anyone who can tell me if this is even possible and if so, how? I've been looking at CDR but I can't seem to extract the information that I,m looking for. System version: 8.5.1.10000-26

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  • Setup wiped Polycom phone without SIP server

    - by Justin
    I'm troubleshooting a Polycom SoundPoint IP 550. I have wiped the hard disk of the phone (via a menu option) and now it's stuck in a reboot cycle. Apparently the only way to setup the firmware of the phone is to use a boot server. Does anyone know another way to setup the phone/firmware?

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  • skype compatible phone

    - by ageis23
    I'm looking for a skype compatible phone I can use to contact my sister when shes logged on sype on her laptop. If possible it should have video conferencing. So a small screen I can she her pretty face on lol.

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  • Call connects through Emapthy, but no voice

    - by Arthur
    I am trying to make calls using Empathy on Debian.The call rings and connects, but I can't hear voice, or answering machines. I had a similar problem a while back that was caused by my PBX not being compatible with a protocol. The protocol issue is fixed now and Linphone works fine on a different pc. I tried using Ekiga, on the problem computer, and it works fine.The problem seems to be with the settings on Empathy. I need to get this going. Any help would be greatly appreciated.

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  • How to auto dial 9+Number on Cisco 7941 redial?

    - by NotDan
    Is it possible to set up a Cisco 7941 phone to dial 9 before the redial number? When I view missed calls, and try to redial one of the numbers, it always fails because it doesn't dial 9 first. I have to write the number down and then manually dial the 9 and then the number.

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  • What are some potential uses of Asterisk (PBX) for a "power user"

    - by mindless.panda
    So I read a lot of good things about Asterisk. I am not however looking to run a call center or small business setup. I am still interested what potential uses it has for me as a "power user" and what features I could harness for my communication needs. I'll throw out that I currently use other technologies like Google Voice, Skype, and a cellphone of course. So, what potential uses, if any, could Asterisk PBX have for a user like me?

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  • Asterisk relay between multiple subnets

    - by immoune
    I wonder what's the best way to go when you have phones on multiple networks which are not directly reachable. I have 3 networks 10.3.x.x 10.6.x.x 10.17.x.x My asterisk server resides on the 10.3.0.5 IP. The machines from the 10.6 and 10.17 networks are routed here through VPN tunnels. At this point we don't talk about NAT anywhere on the network just pure routing. Since the 10.3.0.5 PBX has routes back to all the subnet's it has no problem to communicate with softphones/hardphones from these ranges. The problem comes from that Asterisk (as far as I understand) only responsible for the SIP communication part not the Audio/Video transmission which is in P2P fashion done between the devices. So although a client using sipdroid from 10.6.x.x is able to connect to the pbx (10.3.0.5) and dial a bria client on the 10.17.x.x network once the phone rings out and the call establishes no audio will be transmitted simply because it has no way to directly connect there. For this there are multiple solutions described in this text: http://msdn.microsoft.com/en-us/library/ee480411%28v=winembedded.60%29.aspx What I would prefer is to keep these networks segregated as they are now. What would be the best solution? Is it possible to actually relay through all the audio/video information through the Asterisk server? That would be the best in my case, I using Astlinux there which has a lot of other parts. Thanks

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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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  • Asterisk server firewall script allows 2-way audio from incoming calls, but not on outgoing?

    - by cappie
    I'm running an Asterisk PBX on a virtual machine directly connected to the Internet and I really want to prevent script kiddies, l33t h4x0rz and actual hackers access to my server. The basic way I protect my calling-bill now is by using 32 character passwords, but I would much rather have a way to protect The firewall script I'm currently using is stated below, however, without the established connection firewall rule (mentioned rule #1), I cannot receive incoming audio from the target during outgoing calls: #!/bin/bash # first, clean up! iptables -F iptables -X iptables -t nat -F iptables -t nat -X iptables -t mangle -F iptables -t mangle -X iptables -P INPUT ACCEPT iptables -P FORWARD DROP # we're not a router iptables -P OUTPUT ACCEPT # don't allow invalid connections iptables -A INPUT -m state --state INVALID -j DROP # always allow connections that are already set up (MENTIONED RULE #1) iptables -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT # always accept ICMP iptables -A INPUT -p icmp -j ACCEPT # always accept traffic on these ports #iptables -A INPUT -p tcp --dport 80 -j ACCEPT iptables -A INPUT -p tcp --dport 22 -j ACCEPT # always allow DNS traffic iptables -A INPUT -p udp --sport 53 -j ACCEPT iptables -A OUTPUT -p udp --dport 53 -j ACCEPT # allow return traffic to the PBX iptables -A INPUT -p udp -m udp --dport 50000:65536 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT iptables -A INPUT -p udp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -p tcp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -m multiport -p udp --dports 10000:20000 iptables -A INPUT -m multiport -p tcp --dports 10000:20000 # IP addresses of the office iptables -A INPUT -s 95.XXX.XXX.XXX/32 -j ACCEPT # accept everything from the trunk IP's iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT # accept everything on localhost iptables -A INPUT -i lo -j ACCEPT # accept all outgoing traffic iptables -A OUTPUT -j ACCEPT # DROP everything else #iptables -A INPUT -j DROP I would like to know what firewall rule I'm missing for this all to work.. There is so little documentation on which ports (incoming and outgoing) asterisk actually needs.. (return ports included). Are there any firewall/iptables specialists here that see major problems with this firewall script? It's so frustrating not being able to find a simple firewall solution that enabled me to have a PBX running somewhere on the Internet which is firewalled in such a way that it can ONLY allows connections from and to the office, the DNS servers and the trunk(s) (and only support SSH (port 22) and ICMP traffic for the outside world). Hopefully, using this question, we can solve this problem once and for all.

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  • [python] voice communication for python help!

    - by Eric
    Hello! I'm currently trying to write a voicechat program in python. All tips/trick is welcome to do this. So far I found pyAudio to be a wrapper of PortAudio. So I played around with that and got an input stream from my microphone to be played back to my speakers. Only RAW of course. But I can't send RAW-data over the netowrk (due the size duh), so I'm looking for a way to encode it. And I searched around the 'net and stumbled over this speex-wrapper for python. It seems to good to be true, and believe me, it was. You see in pyAudio you can set the size of the chunks you want to take from your input audiobuffer, and in that sample code on the link, it's set to 320. Then when it's encoded, its like ~40 bytes of data per chunk, which is fairly acceptable I guess. And now for the problem. I start a sample program which just takes the input stream, encodes the chunks, decodes them and play them (not sending over the network due testing). If I just let my computer idle and run this program it works great, but as soon as I do something, i.e start Firefox or something, the audio input buffer gets all clogged up! It just grows and then it all crashes and gives me an overflow error on the buffer.. OK, so why am I just taking 320 bytes of the stream? I could just take like 1024 bytes or something and that will easy the pressure on the buffer. BUT. If I give speex 1024 bytes of data to encode/decode, it either crashes and says that thats too big for its buffer. OR it encodes/decodes it, but the sound is very noisy and "choppy" as if it only encoded a tiny bit of that 1024 chunk and the rest is static noise. So the sound sounds like a helicopter, lol. I did some research and it seems that speex only can convert 320 bytes of data at time, and well, 640 for wide-band. But that's the standard? How can I fix this problem? How should I construct my program to work with speex? I could use a middle-buffer tho that takes all available data to read from the buffer, then chunk this up in 320 bits and encode/decode them. But this takes a bit longer time and seems like a very bad solution of the problem.. Because as far as I know, there's no other encoder for python that encodes the audio so it can be sent over the network in acceptable small packages, or? I've been googling for three days now. Also there is this pyMedia library, I don't know if its good to convert to mp3/ogg for this kind of software. Thank in in advance for reading this, hope anyone can help me! (:

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  • "RFC 2833 RTP Event" Consecutive Events and the E "End" Bit

    - by brian_d
    Hello, I can send out a RFC 2833 dtmf event as outlined at http://www.ietf.org/rfc/rfc2833.txt When I do set the E "End" bit, but leave it as 0, I get the following behaviour: If for example keys 7874556332111111145855885#3 were pressed, then ALL events would be sent and show up in a program like wireshark, however only 87456321458585#3 would sound. So the first key (which I figure could be a separate issue) and any repeats of an event (ie 11111) are failing to sound. In section 3.9, figure 2 of the above linked document, they give a 911 example. Here all but the last event have the E bit set. When I set the bit for all numbers, I never get an event to sound. I have thought of a couple possible thing but do not know if they are the reason: 1) figure 2 shows payload types of 96 and 97 sent. I have not nor know how to exactly. In section 3.8, codes 96 and 97 are described as "the dynamic payload types 96 and 97 have been assigned for the redundancy mechanism and the telephone event payload respectively" 2) In section 3.5, "E:", "A sender MAY delay setting the end bit until retransmitting the last packet for a tone, rather than on its first transmission" Does anyone have an idea of how to actually do this? I have also fiddled around with timestamp intervals and the RTP marker. Any help is greatly appreciated. Here is a sample wireshark event capture of the relevant areas: 6590 31.159045000 xx.x.x.xxx --.--.---.-- RTP EVENT Payload type=RTP Event, DTMF Pound # (end) Real-Time Transport Protocol Stream setup by SDP (frame 6225) Setup frame: 6225 Setup Method: SDP 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: telephone-event (101) Sequence number: 0 Extended sequence number: 65536 Timestamp: 0 Synchronization Source identifier: 0x15f27104 (368210180) RFC 2833 RTP Event Event ID: DTMF Pound # (11) 1... .... = End of Event: True .0.. .... = Reserved: False ..00 0000 = Volume: 0 Event Duration: 2048

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  • Technology behind twilio

    - by John Stewart
    I wanted to discuss the technology behind Twilio. I have been playing around with the service for a few days now and it is simply mind-blowing. While I don't have a direct need for it right now, I am curious to find the back-end of the technology. So can anyone shed some thoughts on how does Twilio do its magic?

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  • Good P2P Flash video conferencing package

    - by Justin Alexander
    Looking for a flash p2p (RTMFP) packaged solution, that includes the following features Time limited sessions: at confrence start, there is a set time limit, when this expires the session ends. Session extension: Sessions can be extended, but require authorization from the server via some sort of REST or Ajaxy response. Generally customizable theme Any suggestions?

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