similar to: Sipura 2000 not answering em_w calls

I have a couple of live T1s sitting around and they are not ISDN(like most of the people that are using Asterisk seem to be using), they are regular old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits. Can I get these T1s to work with a T100P Digium card and asterisk? Searching through the lists and the documentation I haven't seen any examples of how to configure this kind

Hello, We have one production server that is identically configured to another production server except for the fact that they each use a different type of digium quad T1 card(one has T400P and one has TE405P). The server with the 405 has been developing delay problems with Sipura SPA-2000 phone adapters after a call has gone on for 15-30 minutes (up to 1 second audio delay). I asked Sipura and

When I send a call from my TE410P using E&M, the legacy PBX answers the call but doesn't route it. Any idea what this could be? I assume the digits aren't being delivered properly to the legacy pbx. Any suggestions on what config settings to muck with? Asterisk SVN-branch-1.2-r40901 built by root @ pbx04 on a i686 running Linux on 2007-01-14 14:05:02 UTC zaptel.conf

I am trying to turn up DID trunks with our local phone company but do not know the correct format of extensions.conf to do this.

Hello, I have a big old Fax machine that will only pick up a ring at 24volts and 20mA(minimum) ring[according to the technical specs manual]. None of my SIP -> Analog phone adapters supply this: Cisco - 50 volts SIPURA - 70 volts Handytone - who knows, but it doesn't work can anyone tell me if there are any analog adapters or channel banks that can send a 24V 20mA ring signal? Also,

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear

I have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys

I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse

I'm setting up Asterisk for the first time. I purchased a SIPURA 2002 ATA to connect with the Asterisk server. In the /var/log/asterisk/messages log I keep getting an error indicating wrong password. Below is the error I am receiving. Note that the IP address and username has been modified for security. Sep 10 15:56:22 NOTICE[24099] chan_sip.c: Registration from 'John Doe

At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: >>[snip] >Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk >can handled the NAT traversal all by itself with Qualify (as John points >out) disabling the NOTIFY will not change anything. > >The NOTIFY will in no way affect the status - unreachable/reachable. > >Another problem with the SIPURA is

With sipura (I tried this with both the 3000 and 841) set to prefer the g726-32 codec, a call from the sipura to asterisk will use g726. Asterisk sip.conf has: disallow=all allow=g726 allow=gsm allow=alaw When the call is from asterisk to the sipura, asterisk will not use g726. It ends up using alaw. I usually use stable but I tried this with head too, and same thing happens. Anybody know how

Hi list, Well I really didn't want to see things get to this point, but Sherman at Sipura along with their President Jan F. leave me no other choice. SIPURA has been provided a letter from our attorney for Breach of Contract and damages. They have yet to respond. A quick background. 1. Sherman (SIPURA's Director of Marketing), stated that we would do a join press release for the Oct

Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just