search for: 20sip

On Thu, 17 Feb 2005 15:04:50 +0100 Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I'm registered with sipgate, a German SIP provider. >Configs works fine so far. Trouble is, after a while, it >seems, my registration is dropped by sipgate. How do I >tell * the interval for * registering with a provider? I >suppose, the re-registration

G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Thanks much. BTY: Does anyone have a How-To on getting the 7960 fully configured

...similar one but I don?t know If mine are hardcoded in some way. This devices are used by MeritCall , MamaKall, Vivophone etc. I found the manual for an exact same type here: http://www.konceptusa.com/downloads/Netphone-KE1020A_KE1021A%20User%20Manual.pdf#search='netphone%20configuration%20for%20sip' Thanks in advance, Francisco

I am trying to make an outgoing call using an iptel account using Asterisk. I have followed a how-to for asterisk and iptel found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER I am getting the following error message: Got SIP response 403 "That is ugly -- use From=id next time (OB)" back from 195.37.77.101 I'm not quite sure what that means. Does anybody know what I might have done wrong? Here is my configuration: sip.conf register =>...

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...meters listed above and it *appears* that 'outgoinglimit' should do what I want, however it also states that this function has been disabled?? "The _outgoinglimit__ is currently disabled in the source code of the SIP channel." http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit My second problem is that when external calls are transferred by our receptionist to other staff members, the CallerID of course changes to her Name instead of the original caller. Is there a way (in the extensions logic or other) to preserve this CallerID information so that sta...

...ChannelStateDesc: Up CallerIDNum: 1234 CallerIDName: <unknown> ConnectedLineNum: <unknown> ConnectedLineName: <unknown> Language: en AccountCode: 11 Context: ABC Exten: 3002 Priority: 8 Uniqueid: 1521724388.197 Linkedid: 1521724388.197 Env: agi_request%3A%20async%0Aagi_channel%3A%20SIP%2F192.168.40.105-000000c5%0Aagi_language%3A%20en%0Aagi_type%3A%20SIP%0Aagi_uniqueid%3A%201521724388.197%0Aagi_version%3A%2013.18.5%0Aagi_callerid%3A%201234%0Aagi_calleridname%3A%20unknown%0Aagi_callingpres%3A%200%0Aagi_callingani2%3A%200%0Aagi_callington%3A%200%0Aagi_callingtns%3A%200%0Aagi_dnid%3A...

I'm using Asterisk to registry several DDI's to a sip proxy (pipecall.com). Everything works fine apart from several times a day my firewall (zywall70) reports a UDP port scan attack from the pipecall sip proxy. I can't seem to work out why this should be. All I could think was that the sip registry was expiring and causing some strange probing from the proxy, is it possible to alter

Hello, I'm trying to do this : http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20mysql%20peers on a FreeBSD (4.9-RELEASE-p11) with mysql 3.23.58 (from the ports) but the only thing i'm getting is : Aug 18 20:22:27 WARNING[135151616]: /usr/lib/asterisk/modules/chan_sip.so: Undefined symbol "mysql_real_escape_string" Aug 18 20:22:27 WARNING[135151616]: Loading mo...

Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian

Hi, I'm using asterisk 1.0.6 and I would let media path be connected directly between the phones without going through Asterisk. I have to it with an AtCom320 (with pa168s chip). I just saw and tryied to do what this page http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%2 0clients%20connect%20directly says. Before going on (with sniffer eth traffic between * and two phones) I'd like to known if it can works. Does anyone just did it? Thanks in advance Gio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.d...

Hi everyone, I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. I have found out something in these links: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels but I don't understand them very well. At first, I tried simply doing this: In SIP Client: -------------------- exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/192.168.0.51:5060,20) (92.168.0.51 is the Asterisk server machine) And nothing on the server side. When calling...

Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there

I have just finished setting up a new asterisk system which is basically the same as our first system. We are using 7960 phones and I used the phone config files the first installation with appropriate changes. The problem is that on the new system I get no message lights, I can't figure this out. One thing I do notice is that when I monitor the sip debug on the second system the sip

Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following command to asterisk via AMI Action: AGI ActionID: C44415 Channel: SIP/192.168.40.105-00001338

Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the

Here is the setup: Phone A (in NYC) on own bandwidth. Phone B (in LA) on own bandwidth. Asterisk box in Houston,TX on own bandwidth. Both phones contact asterisk to register. Not much bandwidth used for this as it is a few packets every hour or so. Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk calls phone B. Both phones are connected and both people are talking.

I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have