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

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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

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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

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  • SIP clients not working on Ubuntu 12.04

    - by Xlearner
    I am trying to find a SIP client that works on Ubuntu 12.04. I have an account on voipdiscount (www.voipdiscount.com) and 12voip (www.12voip.com) Earlier on Ubuntu 11.10, I was able to use SIP clients: Twinkle, SFL Phone. Using these clients, I was able to access my account and make the phone calls to different destinations. But after I

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  • Configuring 12voip with a sip softphone in Ubuntu

    - by Joey
    I want to change my work PC to Ubuntu but one thing is in my way... We use 12voip since we didn't find lower calling rates in Europe. The thing is 12voip doesn't have a Linux program :-(. I tried to set it up with instructions from this page (under 'Software configuration') with almost all clients that the software center of Ubuntu offers,

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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

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  • Implementing the transport layer for a SIP UAC

    - by Jonathan Henson
    I have a somewhat simple, but specific, question about implementing the transport layer for a SIP UAC. Do I expect the response to a request on the same socket that I sent the request on, or do I let the UDP or TCP listener pick up the response and then route it to the correct transaction from there? The RFC does not seem to say

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  • Cloud services, Public IPs and SIP

    - by Guido N
    I'm trying to run a custom SIP software (which uses JAIN SIP 1.2) on a cloud box. What I'd really like is to have a real public IP aka which is listed by "ifconfig -a" command. This is because atm I don't want to write additional SIP code / add a SIP proxy in order to manage private IP addresses / address translation. I gave

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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

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  • SIP UAS asks for OPTIONS

    - by TacB0sS
    Hey, I have UAC that registers to a UAS, after registration the UAS sends me an OPTIONS request, what should I answer it? only the audio media streams? Update I: Allow me to explain myself better... if I want to invite someone to a session I USE the INVITE method and negotiate the media then, for that specific

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  • PSTN Trunk TDM400P Install on Asterisk / Trixbox

    - by Jona
    Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I've run setup-pstn which produces the

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  • SIP Service to record all calls?

    - by TK Kocheran
    I read an article that I can't find at the moment which detailed a way to have Google Voice point to a SIP phone number which forwards to your phone in order to take advantage of the SIP service in order to Have all calls use a data connection = no usage of cell-phone plan minutes. Record

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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 /

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  • maemo - n900 - SIP call quality

    - by Walter White
    Hi all, I have been using SIP / VoIP on my n900 to make calls and my problem is after about 15 minutes of talk time, more recently 18 minutes exactly, my connection dies and I can no longer hear them or them me. I have tested this with various VoIP providers to confirm that it is not

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  • Unregister SIP UAC message

    - by TacB0sS
    Hi, I've looked so much on the internet, but I could not find a any SIP Unregister example, and when I search RFC 3261,3665 the word does not even appear, perhaps I'm searching for the wrong phrase. I manage to understand the part of setting the expires to zero, but it still does

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