similar to: Use one sip account for multiple sipura

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 found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

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

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

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

Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad

Hi all, I was wondering whether anybody here would help me clarify this minor issue please. If I have the following setup; Asterisk ------ Sipura SPA 3000 (fxo) --------- Pstn Line Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does

I know the subject of faxing has been covered in some detail, but I was wondering if anyone has a hardware configuration similar to ours that has faxes working successfully and would be willing to share any settings/insight. We are unable to fax reliably with a Sipura 2100 connected to Asterisk. We do not route calls over the Internet and our network has very low latency. The Asterisk

I can not make a "call pickup" to work with Sipura-3000. I have one SIP phone and one is connected to ATA Sipura-3000 I've in all sip.conf context callgroup=1 pickupgroup=1 in features.conf I've tired: pickupexten = *88 pickupexten = *8 Nothing works. What am I missing? -- #Joseph

Could anyone please help me with that: I have an analog telephone connected to Asterisk using a Sipura 2002 ATA. When calling the extension, the caller ID presented is always the number of that extension rather than the number of the calling one. While I learned that this is the new standard behaviour (?) of Asterisk, I want to show the original caller ID. I tried the options o and f in the

I am wondering if anyone has sample XML config for the Sipura 2100 ATA. We have been autoprovisioning our 2002s with success and the 2100's take the same XML that we have come up with, but I am not sure of the syntax for specific things that I need these boxes to do, such as turning T.38 on. If anyone is willing to share their xml for autoprovisioning Sipura 2100's, it would be much

hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hi All, There are two option in QoS settings of the SIPURA ATA. ( I can't just remember them). please tell me what is better and which one should I choose for my DSL line (128kbps) with a small LAN. thank you kumara

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I have to make a config change to any of these

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

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter