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    Linksys SPA 3000

    Thanks very much, to both of you, for the help. The silver bullet was CPC Duration. Setting CPC Duration to 1, the only change I made, resolved the issue and the annoyance is gone.
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    Linksys SPA 3000

    I've got an old installation with a Linksys SPA3000. It's laid out like this: SIP Asterisk ---> SPA3000 ---> Analog phone with answering machine. | | Analog phone line. The analog phone is configured as an Asterisk SIP extension. The...
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    IAX Blind Transfer issue. Yikes!

    If you mean to say that split horizon routing at the IP level is a factor here; no, it should not be. If you mean that there may be something akin to split horizon within Asterisk's call routing functionality(very different from IP routing), I suppose that's possible, though I am unaware of any...
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    Incoming not working on a local extension, getting "unable to create channel of type"

    The extension isn't registered. First I see you are using a nonstandard SIP port(5064) for extension 113. Unless 112 is set to use 5060, make sure that your firewall and fail2ban are not interfering with the non-standard port you have configured. If it is one of the Nortel 1535s, power cycle...
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    Fixed: PAP2T-NA installation

    Why won't it register? Have you looked at the Asterisk log for insight? For this device to work you need only a few things: 1. A configured SIP extension on PiaF. 2. Specify the correct SIP Proxy server IP address(PiaF IP addr.) and port on the PAP2. 3. Correct UserID (PiaF extension number)...
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    Nortel 1535 only one way video

    It depends. When dialing a 10 digit number does the entire call path, every trunk along the way, support h.263 video? This is a requirement. If your call traverses the PSTN or goes through a SIP provider that does not support h.263 codecs then no, you cannot make 10 digit video calls. If you...
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    IAX Blind Transfer issue. Yikes!

    That should work. Can you look at the logs for each server and see what is going on at the time of the transfer? Edit: You're not transferring back to the ring group 778, are you?
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    chan_sip.c: Retransmission timeout reached on transmission

    Glad you got it working but, there are two things. 1. SIP AGLs are an abomination. They are supposed to resolve NAT issues specific to SIP, rewriting addresses within SIP headers. They always seem to be broken, regard less of manufacturer so, turn them off. I find things work better if Asterisk...
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    IAX Blind Transfer issue. Yikes!

    So, what does the routing from Server2 back to Server1 look like? Can you dial extension to extension from Server2 to Server1? You need to be able to do this first.
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    chan_sip.c: Retransmission timeout reached on transmission

    You don't specify the severity of your inconsistent results. Are they bad or are they just inconsistent? Unless your home network tests are bad, your issue stinks of NAT. You've been very thorough in your troubleshooting so far and it all sounds correct. Check the configuration on the Netgear...
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    Problem dialing out through ZAP

    Check the channel's context in /etc/asterisk/chan_dahdi.conf context=from-zaptel
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    Gate Access Phone - FXS or FXO

    Door panels, such as this, are extensions. Think of them as any other phone. This one happens to have a "To Phones" jack which is likely just a pass through, like a common fax machine has. The end that you will need to utilize with PiaF/Asterisk is the "From Telco" port. This FXO port will need...
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    Conference Calls drop unexpectedly

    I think we are onto something. follow randy7376 advice. tsc timing is the least accurate/precise kernel timing source. If I am not mistaken jiffies are the next higher precision, then PIT and finally HPET is the highest precision. The reason that hpet isn't available isn't because your kernel...
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    Conference Calls drop unexpectedly

    Despite the DAHDI card, it sure sounds like timing issues to me. Perhaps this post may help. But, you've expertly described covering most all possibilities.
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    DTMF Volume?

    Verizon would be my guess.
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    Caller ID question/Help

    You are not receiving any caller ID information from AT&T. Not surprising since you are not paying for it. Because there is no CID information, Asterisk is using the locally configured information on the trunk. You can change that trunk information to whatever you like but, it will remain the...
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    Forbidden

    Yes. You should be prompted to enter a user ID and password. Have you cleared your browser's cache/cookies?
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    FIX: Asterisk 1.8 Blind T'fer

    While it might be a big deal to you and many others, the fact that neither Asterisk nor Ward have seen fit to incorporate the fix into 1.8 or 1.8.1 suggests that they do not feel the same way about it. Blind transfers are a very nice convenience feature but, they're absence does not seriously...
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    What is situation with SIP ports these days

    I'm happy for your success! I always prefer a VPN to exposed services(open ports or port forwarding). However, for continued accuracy(and pedantry) you did open ports and likely protocols too, in order to pass the VPN tunnel through. You didn't open SIP ports as SIP and RTP traffic are...
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    TUTORIAL Easy OpenVPN

    I'm not trying to dissuade you from OpenVPN. But, I am curious as to what issues made you "tired of depending on Hamachi".
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