search for: spa112

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 same SPA112 port, I...

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...

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_> <FAX_T38_R...

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, 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 no...

Hi all, I'm not sure how this happened, but one of my customers managed to turn call forwarding on on his spa112. I thought I had that turned off in the provisioning file. I have this in the provisioning file: <Cfwd_All_Serv_1_>No</Cfwd_All_Serv_1_> <Cfwd_Busy_Serv_1_>No</Cfwd_Busy_Serv_1_> And I have a similar entry for line 2. When I dial the device, I use this line to dial it:...

...t > > available from Cisco? > Oh! I didn't guess to check. The firmware was not fresh, but upgrading > doesn't help. > > Looking at your SIP information when your ITSP initiated a T.38 > > session it did not indicate a maxmimum bitrate, it would appear your > > spa112 attempted to negotiate a connection at 2400bps. > Whether there is a way to force my provider to indicate maximum bitrate? > > Do you have a sip debug session when you sent a fax from your Asterisk > > box to the PSTN, it would be interesting to see if it sends it as a > > t.38...

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

Hi all, I have a user with an old Mitel PBX connected to a couple of SPA112's. The user is reporting that their phones ring several times a day and when they answer the "call," all they hear is dial tone or busy signal. Their PBX guy says that the SPA112's aren't providing line supervision and the PBX requires it. Does anyone know how to fix this?...

...th the T.38 gateway patch applied (I know I know, we should move to asterisk 11. I'm trying that tonight after business hours). The issue we're seeing is that faxes incoming from some specific fax machines consistenly fail. I have indentified 2 types of failures: - Incoming call is sent to SPA112, reinvite to T.38 is initiated and accepted. One T.38 packet is sent from the SPA112 to asterisk (t30ind: no-signal, see pcap output below) followed by 20 seconds of nothingness and then the call is hung up. - Incoming call is sent to SPA112, reinvite to T.38 is initiated and accepted. A number of...

Hi all, I've got a few devices, SPA112's and SPA8000's, that are giving me problems. Each device has a separate SIP credential for each port, but sometimes, only a few of the ports register. So, the device will be running fine for a while, then suddenly one or more of the ports will become Unreachable. These ports will st...

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) Now, we have a new customer (tenant) that have Panasonic TDA600. If we use FXS or ATA Converter, its going to have a lot of that, because this tenant going to use abo...

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 later. He always gets registered but when a call is sent it doesn't respond so the caller hears no ring and the phone...

...cannot talk with one user. The different elements are as follows: The switch as described above which is in a server room on the Internet backbone with a public IP address. My home system which is behind a bridged modem through a Linksys WRT54GS with priority given to my ATA. The ATA is a Cisco SPA112. I also have an actual SIP phone. The problem happens with both. Obviously I am using NAT but both devices work just fine if I am going to the PSTN. My user who is also going through a bridged modem to a Linksys SPA-2102 which is doing the PPPOE so it has a public IP address and no NAT involved...

...asterisk-users at lists.digium.com Subject: [asterisk-users] Unstable phone connection 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 later. He always gets registered but when a call is sent it doesn't respond so the caller hears no ring and the phone do...

...evices to register on the same user/secret, so the u/d mode would not make sense any more; however this creates another interesting problem, pls read on) Some endpoints are grouped in pairs so that calling an extension, rings on both devices. (One 'device' is a real handset, usually dumb: SPA112 or SPA301, the other is a softphone (CSipSimple or WebRTC or both) used to bring the incoming CID to users' eye level and to perform some client-side CRM integration ) On Incoming call, as expected, the softphone shows me the CID [as intended] and I can pick up the handset, then the softphon...

...evices to register on the same user/secret, so the u/d mode would not make sense any more; however this creates another interesting problem, pls read on) Some endpoints are grouped in pairs so that calling an extension, rings on both devices. (One 'device' is a real handset, usually dumb: SPA112 or SPA301, the other is a softphone (CSipSimple or WebRTC or both) used to bring the incoming CID to users' eye level and to perform some client-side CRM integration ) On Incoming call, as expected, the softphone shows me the CID [as intended] and I can pick up the handset, then the softphon...

Hi, Is there anyone here who has experience connecting Asterisk (ver 13.8) with PBX Panasonic KX-TDA600 ? The architecture more less like this : Telco Sip Trunk ---> Asterisk 13 ---> Panasonic KX-TDA600 ---> Phone / Fax Thanks in advance, Regards, Ikka - Jakarta, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL: