similar to: 2 ip phones and 1 normal, can't neither send nor receive calls at all...

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

Hello, If I'm not mistaken, it is not possible to get T.38 on a SPA3102 FXO port (it is possible with the FXS port). Do you know, by experience preferably, if this is possible with an SPA8800 FXO port ? Regards

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I'm trying to get the spa 8000 used with a fax machine using t38 passthru i have tried with 1.6.2 and 1.8.3 and is still a no go the provider i use is 012 in israel wich supports t38 (i use it with ffa) could anybody give me a clue how to get this working if it should t38pt is set to yes in sip.conf Thanks, Israel -------------- next part -------------- An HTML attachment was

We are evaluating starting a small VoIP consumer based platform. What is the best codec to use with customers using primarily DSL as internet connectivity? I know that g729 is the king-all, but I want to know what the rest of the professional are using out there. g729 has a cost involved, so does the cost really offset the performance? Or is it better to go with g711 to start off? We plan

Hi, I'm having a major problem with random calls dropping. After spending weeks trying to figure it out, i've finally spotted the issue but don't know how to resolve it. I run a sip server that's hosted in a data centre. It has a public IP address with no nat involved. My provider also has a public ip with no nat involved. The sip phones are in a remote office behind a nat

Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110212/ccd9d985/attachment.htm>

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I

Hi, We've got 6 Linksys SPA921s. Does anyone know how to get the "Call Park" soft button to appear? During a call, it is not on any menu. Currently, we have to dial the access code manually. Also, the manual mentions that the user menu has an option labelled "Call Park Status" but it's not in any of the menus.. Many Thanks Dan Journo -------------- next part

Hello, I'm using a dial command to make several phones ring. I use this format : Dial(SIP/4029&SIP/4030,15,tTr) As soon as one of the phone is picked up, all the others should just stop rining. But the fact is that they continue to ring for several seconds (4-5s), and this is quite annoying as all phones are in the same room. Phones are Siemens Gigaset C450IP and Linksys SPA921. Do you

Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir:

Hi all, I have a problem with some asterisk boxes. I have a standard installation with 1.4.14 (I also test with 1.4.4) in core duo Mac Mini on Debian Etch. I use SJphone softphone, Linksys SPA921 or Thomson 2030 for phones. All my phones are in a LAN with good status of 2ms max. Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any

Has anybody tried both and favored one of the other? Or maybe tried both and in the end used neither? I''m having perf (generic searches across 4 fields take upwards of 120 seconds) and index rebuild issues (not sure how long it takes to fully rebuild since the process either gets killed or dies due to a file lock exception) with a dreamhost site that is using AAF, has 65k rows in 1

Hello! On Win9x clients: With dos filetime = no, if the owner change the file the file's modify date will be the owners workstation date, if another user change the file the files modify date will be the samba servers date. On WinNT clients the behavior is uniform: even if the owner change the file, the file's modify date will be the samba servers date. How can I achieve that the Win9x

Package: xen-utils-common Severity: normal When setting (network-script network-bridge) in the xen config, restarting xen will result in error messages like: /etc/xen/scripts/network-bridge: line 219: brctl: command not found /etc/xen/scripts/network-bridge is part of xen-utils-common which doesnt depend or recommend the bridge-utils package which contains brctl.

> Good day, > > I was expecting this to be a FAQ but haven't been able to find an > answer... how would I arrange to, normally, not be presented with either > prompt or menu when booting, yet be still able to call up prompt/menu > when needed? > > If the right answer is "set PROMPT 0, provide a DEFAULT entry and no UI > entry" then that doesn't

Hi, I'm noticing that there are some DSNodes in main() that are neither complete nor global. Specifically, a heap DSNode that is used but not free()d in main() is not marked complete. This is not the only case I'm noticing, but it's the only case I'm noticing for nodes that are actually used by instructions in the function. Is this expected behavior? If so, is there an

Hi, I must say that i'm not very used with customization of FOP. I've a box runing Flash Op.Panel, and i notice that the screen is full of buttons from my sip users, as well as Zapata channels. The problem is that i have more Zapata channels as well as SIP users, is there any way to get a scroll on this to display everything? do i need to resize the buttons? For sure someone now how to