similar to: Phone always busy after caller hangup

I have a Uniden wireless phone connected into Linksys/Supura 2100. It works well, except I never see any caller ID information displayed on the phone. Is that a setting in the 2100 that I'm missing, or is it an Asterisk setting or isn't it possible? Thanks, Jim.

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

I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 10000 - 30000 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see

Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called pstn_number@SER_ip_address SIP/SER_ip_address-ec75 is

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

Hi Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. Thank you. Regards, Chandramouli

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"

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

I've just made an interesting observation that I'd like to share with you all: the popular Sipura SPA-2100 just doesn't seem to be as great as I'd hoped. I've been trying to get inbound AND outbound faxing working via Asterisk and at least one of my termination services: Voicepulse or Sixtel. In general, inbound has been working flawlessly but outbound has been pretty

I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4. Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo of spa3000, all works ok. Then I call from a sip phone configured for using g729, to the fxo of spa3000, it also works ok. The problem is that after this, when, making again a new call from spa2100 to spa3000, spa2100 receives only white noise. I suspect a

Sorry to bug all of you with this, but I have to assume there are a number of Sipura experts here... I have a Sipura SPA 2000 that I've been using for nearly 2 years now. It's flashed up to firmware 3.1.5. On line 1, I no longer get Caller ID (it used to work, and I can't remember when it stopped). On line 2, I always get Caller ID. To my old eyes, _every_ switch on both lines

Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN<->VoIP settings. Sipura Configuration website form field names are numeral only. :( [1]

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

Well I just need to say I got my phone last week. Here is my quick review of the phone and hope that someone has a possible fix for it or I will be sending it back. First the phone is nice looking in my view and it's heavy so it feels like a real desk phone. But it has these stick, gummy or I really don't know how to describe the bottoms on the phone. There good size but when you press

We are testing the new T38 passthrough code (SVN-oej-t38passthrough-r13347): - we are using a Sipura SPA-2100 as the T.38 user device - we are using a Patton SmartNode 2400 as the T.38/PRI gateway - we are using Asterisk in the middle We have the following in the [general] section of our sip.conf: t38pt_udptl = yes t38pt_rtp = yes When a fax call comes in from the SmartNode to Asterisk