Re: [Asterisk-Users] spa-3000 review?
Wolfgang S. Rupprecht wrote: Interesting. I'm at -current +/- a day and do see a NAK/retry-with-md5 exchange when I do a sip debug. The md5 authentication is also NAK-ed. Well you got farther than I got when I was having problems. :) My fear was that it was expecting the calling user to use their own username in the validation instead of asterisk using the shared secret with a shared user-id. Asterisk should use whatever credentials you define as HTTP Username/Password in the SPA-3000 configuration. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
Wolfgang S. Rupprecht wrote: Have you gotten asterisk to work for dial-out to the PSTN when using a md5 authentication? What I discovered via tcpdump was that the Asterisk box wasn't responding to the authentication request for whatever reason. I couldn't get it to work until I upgraded to the latest CVS release. Once I did that, I could do it with authentication. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
(Dameon D. Welch-Abernathy) writes: Wolfgang S. Rupprecht wrote: Have you gotten asterisk to work for dial-out to the PSTN when using a md5 authentication? What I discovered via tcpdump was that the Asterisk box wasn't responding to the authentication request for whatever reason. I couldn't get it to work until I upgraded to the latest CVS release. Once I did that, I could do it with authentication. Interesting. I'm at -current +/- a day and do see a NAK/retry-with-md5 exchange when I do a sip debug. The md5 authentication is also NAK-ed. My fear was that it was expecting the calling user to use their own username in the validation instead of asterisk using the shared secret with a shared user-id. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
[EMAIL PROTECTED] (William Suffill) writes: Seems quite interesting. Any suggestions of where to order one and about how much? Mine was $125 from www.voxilla.com. I ordered it on Sunday and had it in my hands on Tuesday. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
On Wed, 2004-07-14 at 14:33, Dameon D. Welch-Abernathy wrote: I experience a some echo. It can be minimized by adjusting the SPA to PSTN Gain and PSTN to SPA Gain values to quieter values. I believe it can range from -12 to 12. 12 is about twice as loud as 6, -12 is twice as quiet as -6. I was incorrect here, it can be anywhere between -15 and 12. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spa-3000 review?
On Wed, 2004-07-14 at 16:56, Kevin Walsh wrote: I'll rush out and buy one for use at home as soon as they support the UK (BT) phone system for Caller*ID and distinctive ringing etc. (as the SPA-2000 does for UK phone handsets). The FXS port on the SPA-3000 supports that stuff, but the FXO port does not. Sipura is looking at supporting that stuff on the FXO port in about 1-2 months, if what they said during beta was correct. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
I've had one for a while now. My primary goal was to be able to insert into into the middle of a phone.. ie, line port to the wall, phone port to the phone at my office. It's seamless. When someone calls my office - SIP/40 is the FXO port, the iquestdesk context dials SIP/30SIP/provider/CELLPHONE or whatever. I have my * box setup so that when I dial _1XXX or _7XXX (our office extensions) it first tries to dial the call out SIP/40, if that fails I dial out to the PSTN and back in. (See below for the sip.conf excerpt.) I also recently purchased a second unit for my parents house. They are complete newbies to VOIP and internet calling. They also have a Bed and Breakfast, so they have the 'public' in their house all the time. They offer free long distance to their guests as an added bonus (Connect.Voicepulse.com for now, so it's not really free.. but close enough since not many people use it.) So, the phone is sitting right out where people walk past it all the time. My biggest concern (and I'm sure a lot of other people on this list..) with this is the public and 911. I don't want to be the one to take the call when they couldn't get help there when they needed it and somebody dies because of it. (I don't feel safe with telling them.. just don't use this phone for 911. In an emergency, that's not an answer.) With the SPA-3000, I was able to modify the dial plan of the SPA-3000 to use gw0 (local PSTN port) in the event someone dials *11. This works even if the network is completely unplugged. Since I have the FXO connected to a POTS line in their house, their address should up on the 911 center. The other reason for this box at my parents if when they start trusting the VOIP/PBX we'll start routing their calls into it as an answering machine/forwarder for them. Another plus side to the SPA-3000, if you pull the power cord it cross connects the phone line. (I've had problems with my office setup where * unfortunately did exactly what I told it to do. :) If you pull the power, it cross connects and the local phone is on the POTS line and completely removes * from the path. I did have a problem originally with caller ID not making it into the box. After talking with them (VERY HELPFUL!) and a firmware upgrade that went away. I had another problem with it not accepting calls. When I posted in the BETA forum, they requested logs. After a short exchange, they found the problem and I had a work around. The first unit I bought was a BETA unit, the second was a production unit. The Beta unit is on 2.0.8 and the production unit is on 2.0.9. However, I've not seen the 2.0.9 firmware up for download yet. The only remaining issue that I have is it doesn't seem to want to receive caller id if it's not 10 digits, this is unrelated to the above caller id problem. Our phone system (Interactive Intelligence CIC) sends caller id out of the extension which is 4 digits. The SPA-3000 seems to miss that? Haven't really dug that far into fixing this one, it's not a major problem for me since I've got a phone client most of the time. Other then that, I've not had any problems with caller id making it through. So far, I'm VERY happy with it and the support folks that I interacted with. IN addition.. I've got a couple of SPA-2000s right now and was already familiar with the interface and the box seems to be about the cheapest I've seen.. so, I'd give it a big thumbs up. ; FXS port - phone at my desk. [30] type=friend host=dynamic context=iquestout secret=NOPE reinvite=no canreinvite=no qualify=300 callerid=IQuest Office FXS 30 nat=1 accountcode=5050 ; FXO port - Line from our office PBX. [40] type=friend host=dynamic port=5061 context=iquestdesk secret=NOPE reinvite=no canreinvite=no qualify=300 ;callerid=IQuest Office FXO 40 nat=1 accountcode=5050 On Jul 14, 2004, at 11:43 AM, Rich Adamson wrote: Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
[EMAIL PROTECTED] (Tom Neville) writes: ; FXO port - Line from our office PBX. [40] ... secret=NOPE Have you gotten asterisk to work for dial-out to the PSTN when using a md5 authentication? I can only dial out when I tell the SPA-3000 to use no authentication. Eg: admin-PSTN Line-VoIP Caller Auth Method-None Changing it to the following doesn't work (adapting the example to use your values from above): VoIP Caller Auth Method: HTTP Digest(their name for MD5 digest) ... VoIP User 1 Auth ID: 40 VoIP User 1 Password: NOPE Turning on sysloging on the sipura wasn't informative at all. (All I got was a bunch of lines like this: Jul 14 16:42:11 hsephone [1:5061]64.142.50.224:5060 Jul 14 16:42:11 hsephone [1:5061]64.142.50.224:5060 Jul 14 16:42:11 hsephone Jul 14 16:42:11 hsephone Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060 Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060 Etherdump also showed quite a few invalid syslog lines coming from the sipura. Mostly they were missing the local0.debug. Some went to local2. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
Seems quite interesting. Any suggestions of where to order one and about how much? On 15 Jul 2004 16:54:03 -0700, Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] (Tom Neville) writes: ; FXO port - Line from our office PBX. [40] ... secret=NOPE Have you gotten asterisk to work for dial-out to the PSTN when using a md5 authentication? I can only dial out when I tell the SPA-3000 to use no authentication. Eg: admin-PSTN Line-VoIP Caller Auth Method-None Changing it to the following doesn't work (adapting the example to use your values from above): VoIP Caller Auth Method: HTTP Digest(their name for MD5 digest) ... VoIP User 1 Auth ID: 40 VoIP User 1 Password: NOPE Turning on sysloging on the sipura wasn't informative at all. (All I got was a bunch of lines like this: Jul 14 16:42:11 hsephone [1:5061]64.142.50.224:5060 Jul 14 16:42:11 hsephone [1:5061]64.142.50.224:5060 Jul 14 16:42:11 hsephone Jul 14 16:42:11 hsephone Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060 Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060 Etherdump also showed quite a few invalid syslog lines coming from the sipura. Mostly they were missing the local0.debug. Some went to local2. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spa-3000 review?
Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
[EMAIL PROTECTED] (Rich Adamson) writes: Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. I got it yesterday afternoon. It is a very cute unit that is surprisingly small. (When I saw the size of the package I was at first afraid they'd mistakenly only sent me a power supply!) The fxo and fxs are indeed separate and show up as two peers and users. bonnet*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 9757/9757192.83.197.10D 255.255.255.255 5061 OK (29 ms) 6003/6003192.83.197.10D 255.255.255.255 5060 OK (22 ms) bonnet*CLI sip show users Username Secret Accountcode Def.Context ACL NAT 9757 XXX from-untrusted- No No 6003 YYY from-trusted-in No No The biggest problem with the unit is that it doesn't come with the slightest scratch of documentation. Not even a URL to download a preliminary manual. Setting it up is apparently meant to be a test that only the true followers of the Polynesian god Sip-Ura will be able to undertake, If one is used to the Grandstream one-page does-it-all http configuration, this baby is going to be a real shock. It goes on for pages and pages and has multiple views where the harder to explain features are not shown, apparently in an attempt to not scare every last person away. It is quite evident that Sipura put quite a bit of work into the code and intent is clearly to provide a mini firmware-based gateway/server that can be used standalone to do much of what we use asterisk for. From paging through the configs it is clear it can do PSTN-VOIP, VOIP-PSTN, VOIP-analog-phone, analog-phone-VOIP, analog-phone-PSTN and PSTN-analog-phone routing, all under the control of touch-tone passwords and/or md5 passwords or RSA certificates. This is all without involving any outside SIP server. I can see that it is going to be a while before I expose this to an outside IP address lest some kiddie that understands the passwords better than I do notices that he can make free PSTN phone calls because I missed filling in filling in one of the dozen or so passwords. Sorry, no detailed HOW-TO's yet. This thing can obviously be made to do what I want of it, but it will be a while figuring it all out. This thing really needs a wiki devoted to it. ;-) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
On Wed, 2004-07-14 at 09:43, Rich Adamson wrote: Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). There's a review of the SPA-3000 on Voxilla. I know of a number of people using the SPA-3000 with Asterisk, as do I. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
On Wed, 2004-07-14 at 14:18, Mike Benoit wrote: Dameon and Wolfgang, Have either of you experienced echo when making a call from the FXS port to the FXO port on the SPA-3000? I experience a some echo. It can be minimized by adjusting the SPA to PSTN Gain and PSTN to SPA Gain values to quieter values. I believe it can range from -12 to 12. 12 is about twice as loud as 6, -12 is twice as quiet as -6. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
Dameon and Wolfgang, Have either of you experienced echo when making a call from the FXS port to the FXO port on the SPA-3000? Thanks -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spa-3000 review?
[EMAIL PROTECTED] (Rich Adamson) writes: Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. I got it yesterday afternoon. It is a very cute unit that is surprisingly small. (When I saw the size of the package I was at first afraid they'd mistakenly only sent me a power supply!) The fxo and fxs are indeed separate and show up as two peers and users. That's good to know. I'll rush out and buy one for use at home as soon as they support the UK (BT) phone system for Caller*ID and distinctive ringing etc. (as the SPA-2000 does for UK phone handsets). My X101P will then be used only to provide timing info etc. As the SPA-3000 appears to contain hardware echo cancellation, I think it'd make a worthy successor to the X101P. The SPA-3000 costs around the same as a X101P (from Digium), and you get a FXS port for free. If I get one and it has the same quality and reliability as the SPA-2000 then I'll have no hesitation in recommending it to anyone who asks - and those who don't. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
Sorry, no detailed HOW-TO's yet. This thing can obviously be made to do what I want of it, but it will be a while figuring it all out. This thing really needs a wiki devoted to it. ;-) Feel free to add as many pages on this topic as you wish to the wiki. Jim [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
[EMAIL PROTECTED] (Mike Benoit) writes: Have either of you experienced echo when making a call from the FXS port to the FXO port on the SPA-3000? There is some echo on LD PSTN calls when the two ends mistakenly talk over each other. I believe they have some VOX that attempts to enforce a ping-pong talk path (eg. the amps in one direction are always set for a gain of 0 while the other direction is a 1.0). Now the first thing that I noticed in making a PSTN call is that the remote side is very hard to hear. The gain between the Sipura-3000/PSTN and a Grandstream BT-100 is much less than between two BT-100's. I'm going to have to bump the gain up a bit in at least the PSTN-VOIP direction. Perhaps I need to do the VOIP-PSTN direction too. This is going to make echo even worse. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
