similar to: Free World Dialup and Asterisk

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a "480 - user is not online" message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

Hi all, I'm currently trying to install a APC Back-UPS ES 700 on FreeBSD. This UPS has a usb connection to the PC and is recognized as: ugen0: APC Back-UPS ES 700 FW:829.D2 .I USB FWD2 rev 1.10/1.06, addr 2 my ups.conf looks like: [back-ups] driver = usbhid-ups port = auto desc = "server" When I now run /usr/local/libexec/nut/upsdrvctl start the output is: Network UPS Tools

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

Dear all Recently, i did some work to test small files performance for gnfsv3 transport. Following is my scenario. #####environment##### ==2 cluster nodes(nodeA/nodeB)== each is equipped with E5-2650*2, 128G memory and 10GB*2 netcard nodeA: 10.254.3.77 10.128.3.77 nodeB: 10.254.3.78 10.128.3.78 ==2 stress nodes(clientA/clientB)== each is equipped with E5-2650*2, 128G memory and 10GB*2

Which router wichi support FXo ports do you recommend for -ASterisk SIP PBx? The smallest and cheapest? --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/6ec0ad0b/attachment.htm

Hi guys, I know this isn't strictly about Asterisk, but it is related... I am looking to buy a few IP phones, but I don't have a huge budged (hence why I love Asterisk, its amazing and free !), so I was wondering if anyone knew where I could get some cheap IP Phones ? Ideally they should be no more then about ?50 ($90). Thanks, Paul.

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

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Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

Hi all, I have seen a lot of messages in the last time about * If you put a section in sip.conf as: > [fwd1] > reinvite=no > canreinvite=no > nat=yes > type=friend > secret=dunk13 > username=33537 > host=fwd.pulver.com > ;host=192.246.69.247 > context=inbound this does make sense??? What is [fwd1]? For me it means that a SIP user with the name fwd1 is defined in

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

I added those lines to my configuration, and i just see with ethereal that my client dial and the 1204 led turn on and they started to interchange packets, im newbie with asterisk i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up? could u send me all the configuration i need step by step? ----- Original Message ----- From: "Wojciech

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called 2217008@Mediatrix Sep 6 16:54:54

I had my asterisk server working fine with FWD as a SIP provider, so I now added a second SIP provider (voctel). The addition to my sip.conf file is almost identical to FWD, however, asterisk now generates lots of debug messages for some strange reason! In particular, the line "##### Testing 127.0.0.1 with 172.31.0.0" shows up every two seconds! (See my log below). If I comment out

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>