similar to: Fwd: Re: Asterisk T.38 Pass-Through doesn't work

Hello, We're trying to receive faxes on the Asterisk server, but for the time being T.38 negotiation fails. The SDP that the Asterisk reINVITE sends contains these lines: ---------------------- m=image 4968 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400

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?

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

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 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 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 List, I think I've discovered a little bug in t.38 bug in 1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes. Asterisk always responds with a=T38MaxBitRate:2400. I've tried with Patton and Grandstream devices and the result is always the same. Patton ignores the parameter and sends the fax at 9600.

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

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)

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

Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18

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

Hi: If you want to use icecast 2 for streaming vorbis audio then don't get it from the CVS repository at icecast.org. That's ancient developer stuff in there. Instead get it from the xiph.org CVS repository (see http://www.xiph.org/cvs.html which I see now lists the icecast stuff (yay!). You'll need the icecast module plus the avl, httpp, log, net, thread and timing modules (check

Hi, I'm experimenting with managed mode encoding with specifying maximum bitrate. I call: vorbis_encode_init( &vorbisInfo, 2, 44100, -1, 96000, 96000); to initialize the encoding. To my surprise, it seems the maxbitrate value of 96000 doesn't have an effect, the bitrate of the

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), I have several extensions that can register 2 separate devices (chan_sip) ( FreePBX calls this Devices & Users mode : Users are extension/internal number, devices are the 'SIP Accounts' for the internal 'endpoints' ) (this

Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT -> ASTERISK): """ . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288.

Hi, I have a Portech MV-374 GSM Gateway and I'd like to send SMS from a web page to confirm the subscriptions. How can I achieve it? Is Asterisk of any use to send SMS with the Portech? I really have no idea because I know nothing about the whole SMS thing... Thanks, Niccol? -- http://www.linuxsystems.it

Hi All, This is my first post to centos-docs. I've read several months of archives, but, this may be too trivial. Re: http://wiki.centos.org/TipsAndTricks/EncryptedFilesystem/Scripts This line: FREE_SPACE=$(df -m $SECRET_PATH |grep / | awk '{ print $4 }') gave me a problem due to line wrapping in df. For example: $ df -m /home/ja/verify/ Filesystem 1M-blocks Used

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things