similar to: Turning off CFWD on an SPA112?

Hi all, I've got a provisioning file that I use to configure Cisco SPA112's. I'm wanting to get this file to do 3 things for me. I want to turn T.38 on, Call forwarding off, and Call waiting, off for both lines. but it's not working. This is what I'm using to enable T.38 for line 1. <FAX_Enable_T38_1_>Yes</FAX_Enable_T38_1_>

Hello, I'm preparing a setup before installing it within the next few days. In this setup, I'm using a SPA112 as an ATA for an analog phone. The target phone is a Gigaset A400 DECT handset. In my lab, I've got another A400 handset and an old Matracom 46 handset. When I connect my Matracom 46 handset to my SPA112, I can send and receive calls. When I connect my A400 handset to the

Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's

Hi all, I've got a device that seems to become unreachable for about 2 minutes, every hour. From what I can tell, it isn't due to network or server issues. Any ideas? TIA. -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

Hi all, It seems that the latest ATAs from Cisco/Linksys support SRTP. Did anybody give these features a go with asterisk? Regards Rajil

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater was sent as "maxBitRate". Without capital "M". Are you closer to

Since upgrading to b129, I''ve noticed that my system clock is ahead by five hours, exactly how much I am behind UTC (US Eastern Time). The strange thing is it only happens when I have xen enabled. If I ''svcadm disable milestone/xvm; reboot'' the clock is correct again. If I correct the time manually, ntp knocks it right back ahead again. Oddly enough, even ntp knows

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

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

This is driving me to distraction. I have a switch with multiple clients who are all working fine except for one and I can't figure out what makes them different. I have tried every NAT setting in the ATA (SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different sip ports, different RTP ports and it still fails. I have left the location with it working only to have it fail

Dear Harry, Thx for the explanation. My team manage building's PBX that use Asterisk 13.x. We use Asterisk PBX for this buildings that have apartment and office customer. >From my Asterisk PBX, we connect to IPPhone (yealink) or ATA Converter (cisco SPA112). Others are using PBX like panasonic analog, audiocodes SBC, etc, and we use ATA Converter to convert from SIP to Analog (CO Line)

D'Arcy J.M. Cain If the device is registering and then dropping there are several usual items. The router may be closing the ports on the device. The router may have a AGL SIP helper that is causing issues. Make sure that the device is sending out keep alive packets. Shut down any AGL helpers on the router. Make sure that the site is not double NATing Try using a stun

Are there any standards for setting up pbx dialplans for businesses/offices? What I mean is that, which numbers are reserved for a specific use ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses. If there is such a standard, * and # are used for what ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been banging my head against the wall for weeks now on this one. I have a switch running NetBSD and Asterisk 11.19.0 although I have had this problem on older versions as well. I, and my users, can call out, we can receive calls, quality is excellent but I cannot talk with one user. The different elements are as follows: The switch as described above which is in a server room on the

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensi?n number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What is an easy way to prevent dumb people from creating a loop? -- Telecomunicaciones Abiertas de M?xico S.A. de

I have the very same situation in one of my networks. To solve this you can dial out from the softphone and to move call to the phone you can simply transfer call to the same user (just if you were transferring call to yourself and the other device will ring. While, as you notice, you cannot dial a device, you can surely call your user to tranfer from a device to another. Please note that call

The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting