similar to: spa8000 t38 faxing

Hi everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) <----->SPA21000<----->LAN<----->Asterisk<--------> PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it "should"

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, Has anyone compared SPA8000 vs. GXW4008 especially in terms of firmware and hardware stability (the feature sets are apparently similar)? Vieri

Hi, I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to both a local Asterisk server and also with a trunk directly to a VOIP provider. Everything works great, except I'm having a problem setting the outbound caller ID to a value different from the SIP username/authname. The SPA8000 has SIP setting for Display Name, User ID, Password, and Auth ID, as well as a "Use Auth

Hello everyone. I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing. Voice is working great, but I never configured anything using T.38 in Asterisk so I'm kinda lost. On my googling I found out that would be best letting the Linksys SPA8000 (for those that don't know, it's identical to the PAP2 but with 8 ports instead of 2) and the telco negotiate with each

Hi guys, Anyone have experience using the Linksys spc tool to provision devices? I've been fighting with it all afternoon and can not get the tool to parse a xml file properly. I can get it to generate me a .cfg file from a text file, but when I tell the unit to pull down the cfg file it doesn't really look like any of my settings were even set from the file I generated. Thanks for any

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 We are experiencing some difficulties with T38 faxing. I have a Asterisk 11.5.0 with libss7 and Sangoma A104DE digital interface card . The operating system is Centos 6 We are using this server to terminate calls to Telco. So calls are coming to asterisk from sip and we are sending calls to Telco with Dahdi. (It is a one way interconnection only from asterisk to telco ,not from telco to

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 everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) <----->SPA21000<----->LAN<----->Asterisk<--------> PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it "should"

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 call to 404 2.- from 404 don't answer it. 3.- from 402 press *8

Hi, I was wondering if I might be able to ask some advice about doing residual plots for the lmer function in the lme4 package. Our group's aim is to find if the expression staining of a particular gene in a sample (or "core") is related to the pathology of the core. To do this, we used the lmer function to perform a logistic mixed model below. I apologise in advance

Hi all! I am fitting a Poisson model, using the following command: > fit2<-glm(canc~id1+year1+time+lnpa,family=poisson) where 'id1', 'year1' and 'time' are factors. I defined them with: > id1<-C(factor(id1), treatment) and 'lnpa' is a continuous variable. The 'summary' function gives me all the effects estimates, that is, for id1, I

Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354

Hello guys. When I am trying to send fax through T38 to linksys SPA (properly configured etc. - I have tried it with other systems), I'm getting error and fax is not delivered. I'm getting this errors in asterisk.log: WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down:

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to

Hello! I have a boot problem with ISOLinux 2.00, downloaded yesterday. The CD works fine with every PC except on an IBM Thinkpad. If I poweron the TP and try to boot from CD I get back a "isolinux: disk error 01, ax = 4211,8B". Then, if I reboot (ctrl-alt-canc) your isolinux works fine. May you help me? -- Stefano Pettini spettini at inwind.it New ICQ number: 159144497 Notary of

Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been

Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys