[Asterisk-Users] ATA's and faxing
Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's and faxing
Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config. Hans Garth van Sittert schrieb: Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's and faxing
I am using alaw and I have already enabled the pass through. Does alaw and ulaw work? I can fax out, but not receive faxes. Garth Johann Steinwendtner wrote: Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config. Hans Garth van Sittert schrieb: Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's and faxing
Garth, this is my sip-configuration for a fax machine at a AT386 ; SIP Accounts Analog devices like Faxmachines [analogdefaults](!) type=friend host=dynamic dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw [222](analogdefaults) context=sip-ol callerid=Fax 222 username=222 secret=123 ; The ATA adapter itself is configured as follows: Fax Mode: (x) T.38 (Auto Detect)Pass-Through so, I don't even have configured Pass-Through Bernd Garth van Sittert wrote: I am using alaw and I have already enabled the pass through. Does alaw and ulaw work? I can fax out, but not receive faxes. Garth Johann Steinwendtner wrote: Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config. Hans Garth van Sittert schrieb: Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's and faxing
ulaw was neccessary when pass through was disabled. What does a sip debug tell you ? Hans Garth van Sittert schrieb: I am using alaw and I have already enabled the pass through. Does alaw and ulaw work? I can fax out, but not receive faxes. Garth Johann Steinwendtner wrote: Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config. Hans Garth van Sittert schrieb: Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA's ???
Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Thanks in advance. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's ???
Hi Phil, if you want to use ATAs take a look at grandstream site...they are better than digium but you could use a card, TDM400 is excellent for analog lines and devices. Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Thanks in advance. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's ???
[EMAIL PROTECTED] wrote: Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Yes, that should work fine - I have a fax machine here connected to a Grandstream Handytone ATA-286 which (with recent firmware) performs faultlessly using G.711 aLaw. I'm not sure how well it will support higer speed modems, though... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA's ???
Phil I have very good experience with the vegasteam ATAs devices.(you might also want to look @ sipura ATAs, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, January 27, 2006 12:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ATA's ??? Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Thanks in advance. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
[...] In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying one to test. Street price around US$ 90. Another one with dual g729 channels is MTA V102. Street price US$ 100. Also will test this one. I'm still looking for other units with dual g729 channels... yoda.com.tw has single, dual and quad channel ATAs, and AFAIK they support all channel codecs individually. roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
On Wed, 16 Feb 2005 07:08:08 +0800, Leo Ann Boon [EMAIL PROTECTED] wrote: See my comments in line From my experience, the ATA is a very solid, dependable piece of hardware. I was told by a source in the company that OEMs for Cisco, the units are expensive because of the high quality parts being used. The web config looks crappy but otherwise where else do you find a $100 device that does SCCP/MGCP/SIP/H323? None of the competitors even come close to that level of protocol support. For developers who have to work on various protocols, the ATA is really cool. I agree. I'm using the ATA 186 and think it's great. The latest firmware also changes the web interface - it's similar to the 7905G/7912G phones now. I've also tried the Sipura SPA-2000, but had some problems with it, so the Cisco ATA is my ATA of choice now. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
Shaun Ewing wrote: I'm using the ATA 186 and think it's great. The latest firmware also changes the web interface - it's similar to the 7905G/7912G phones now. Did you have much trouble getting the ATA 186 working? I have one running Version: v3.1.0 atasip (Build 040211A) I have it setup and it does poll the * server but does not work to use and errors in sip. Followed the instructions on the wiki page for them and it still wants to be a pain :( Other problem is that it is in Denmark and I am in AUS :) so timming is an issue. Any advice would be appreciated. David Uzzell This is the sip debug from * end. Sip read: REGISTER sip:203.29.98.221 SIP/2.0 Via: SIP/2.0/UDP 62.79.110.156:5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: Test901 sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;expires=120 User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 62.79.110.156 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.79.110.156:5060;received=62.79.110.156;rport=5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone;tag=as560861ed Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 62.79.110.156:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 62.79.110.156:5060;received=62.79.110.156;rport=5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone;tag=as560861ed Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=pbx.unifiedau.net, nonce=4a523e7e Content-Length: 0 to 62.79.110.156:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
David Uzzell wrote: Shaun Ewing wrote: I'm using the ATA 186 and think it's great. The latest firmware also changes the web interface - it's similar to the 7905G/7912G phones now. Did you have much trouble getting the ATA 186 working? No. I have one running Version: v3.1.0 atasip (Build 040211A) I'm using Version: v3.0.0 atasip (Build 031210A). I have it setup and it does poll the * server but does not work to use and errors in sip. Followed the instructions on the wiki page for them and it still wants to be a pain :( Other problem is that it is in Denmark and I am in AUS :) so timming is an issue. Any advice would be appreciated. Looks like you have a NAT problem. You need to be more specific about your NAT setup. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
On Feb 15, 2005, at 6:08 PM, Leo Ann Boon wrote: From my experience, the ATA is a very solid, dependable piece of hardware. I was told by a source in the company that OEMs for Cisco, the units are expensive because of the high quality parts being used. The web config looks crappy but otherwise where else do you find a $100 device that does SCCP/MGCP/SIP/H323? None of the competitors even come close to that level of protocol support. For developers who have to work on various protocols, the ATA is really cool. Guess I never really looked at it that way. Perhaps when if I cancel my Vonage account I'll just hang on to the ATA [he casually comments as the collection of VOIP adapters steadily grows in the basement...]. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs Excuse me I have just bought a PAP2 ,, is it true that only one g729, one of the Damn things Cisco had in the ATA186? at the same time. DAMN , its just a Sipura inside I really dont know why is this, offer a 1 port version or a optional second port g729 is really a pain this. regards HA On Tue, 15 Feb 2005 08:52:21 +0100, Nicolas Bougues [EMAIL PROTECTED] wrote: On Mon, Feb 14, 2005 at 10:47:23PM +0900, Hermann Wecke wrote: Matthew Boehm wrote: [...] In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying one to test. Street price around US$ 90. Another one with dual g729 channels is MTA V102. Street price US$ 100. Also will test this one. I'm still looking for other units with dual g729 channels... Back in december, the Uniden was supposed to do 2xG729 at a later time. Not sure if the current firmware allows it. BTW, I've been fairly disappointed with Uniden firmware and their release cycle : their hardware is great, but they take months to release new firmwares, even when phone crashing bugs are discovered. If you want 2xG.729 now, working reliably, for under $90, you can't go wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is a bridge mode, where the LAN and WAN ports would act just like a switch, so that you can easily chain devices without routing/NAT. Just like most IP phones do. -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] ATA's
On Feb 15, 2005, at 3:17 AM, Voip Business wrote: hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs I can't speak so fondly of the Azatel which I had sitting around after a canceling a VOIP service. Maybe I just need a new firmware rev (but they don't exactly make those available at the Azatel site). Plus, the web interface is excruciatingly limited. I mean, you can't even configure echo cancellation. I think the ATA186-L2 is kind of pointless at this stage. It's old hardware...although Cisco did end up issuing a firmware update last year. Still, there's got to be some reason why Cisco as switched to using a Sipura produce (the PAP2)BTW the ATA186 was designed by some of the Sipura folks as well. My choice is still Sipura-branded equipment. There's no way of knowing how often firmware will be released for the Linksys-branded stuff or what level of support there will be. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
what about the digium S100i, haven't used but any comments? i know it's only one fxs one lan port does g711 also. no g729. On Tue, 15 Feb 2005 10:29:35 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 15, 2005, at 3:17 AM, Voip Business wrote: hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs I can't speak so fondly of the Azatel which I had sitting around after a canceling a VOIP service. Maybe I just need a new firmware rev (but they don't exactly make those available at the Azatel site). Plus, the web interface is excruciatingly limited. I mean, you can't even configure echo cancellation. I think the ATA186-L2 is kind of pointless at this stage. It's old hardware...although Cisco did end up issuing a firmware update last year. Still, there's got to be some reason why Cisco as switched to using a Sipura produce (the PAP2)BTW the ATA186 was designed by some of the Sipura folks as well. My choice is still Sipura-branded equipment. There's no way of knowing how often firmware will be released for the Linksys-branded stuff or what level of support there will be. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
Every member of this list is deeply indebted to Digium for making Asterisk freely available. However, do quick search of this list for IAXy, and you'll probably run the other way from this little device. It *appears* to be very immature. -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 15, 2005 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ATA's what about the digium S100i, haven't used but any comments? i know it's only one fxs one lan port does g711 also. no g729. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is a bridge mode, where the LAN and WAN ports would act just like a switch, so that you can easily chain devices without routing/NAT. Just like most IP phones do. So what happens if you try and chain a bunch of SPA-2100s? Does the 2100 act more like a router? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
There are tons of comments about the S100i, or IAXy, as it's called. Can't say I like it myself, we have 3 of them right now, and they get really hot and for some reason don't have the MAC address labeled on them, and also we couldn't get them to actually take an IP from a Microsoft DHCP server (worked fine with Linux/Linksys). Also, the IAXy provisioning is annoying. But this is just my thoughts based on my experience. -- Dana On Tue, 15 Feb 2005 12:29:42 -0500, Erick Perez [EMAIL PROTECTED] wrote: what about the digium S100i, haven't used but any comments? i know it's only one fxs one lan port does g711 also. no g729. On Tue, 15 Feb 2005 10:29:35 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 15, 2005, at 3:17 AM, Voip Business wrote: hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs I can't speak so fondly of the Azatel which I had sitting around after a canceling a VOIP service. Maybe I just need a new firmware rev (but they don't exactly make those available at the Azatel site). Plus, the web interface is excruciatingly limited. I mean, you can't even configure echo cancellation. I think the ATA186-L2 is kind of pointless at this stage. It's old hardware...although Cisco did end up issuing a firmware update last year. Still, there's got to be some reason why Cisco as switched to using a Sipura produce (the PAP2)BTW the ATA186 was designed by some of the Sipura folks as well. My choice is still Sipura-branded equipment. There's no way of knowing how often firmware will be released for the Linksys-branded stuff or what level of support there will be. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] ATA's
See my comments in line Mark Eissler wrote: On Feb 15, 2005, at 3:17 AM, Voip Business wrote: hello, my experience 1.-Azatel Azacall 200 GREAT PIECE OF HARDWARE 2.- MTA-V102 3.- Sipura spa 2000 4.- Granstream ATA186 SUXs I can't speak so fondly of the Azatel which I had sitting around after a canceling a VOIP service. Maybe I just need a new firmware rev (but they don't exactly make those available at the Azatel site). Plus, the web interface is excruciatingly limited. I mean, you can't even configure echo cancellation. I think the ATA186-L2 is kind of pointless at this stage. It's old hardware...although Cisco did end up issuing a firmware update last year. Still, there's got to be some reason why Cisco as switched to using a Sipura produce (the PAP2)BTW the ATA186 was designed by some of the Sipura folks as well. From my experience, the ATA is a very solid, dependable piece of hardware. I was told by a source in the company that OEMs for Cisco, the units are expensive because of the high quality parts being used. The web config looks crappy but otherwise where else do you find a $100 device that does SCCP/MGCP/SIP/H323? None of the competitors even come close to that level of protocol support. For developers who have to work on various protocols, the ATA is really cool. And, the ATA is relatively easily to port forward on the firewall. Just 1 SIP + 8 RTP ports, compared to the recommended 2 SIP + 1 (ten thousand) RTP ports + 2 misc ports for the Sipura. Documentation for the ATA is really good - the usual Cisco standards. Most of the competitors ship with simple manuals that come nowhere close. My choice is still Sipura-branded equipment. There's no way of knowing how often firmware will be released for the Linksys-branded stuff or what level of support there will be. I have an SPA3000. The configuration is very comprehensive but the documentation (the 90+ page version) is not as well written as the ATA. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
On Tue, Feb 15, 2005 at 01:40:24PM -0600, Matthew Boehm wrote: wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is a bridge mode, where the LAN and WAN ports would act just like a switch, so that you can easily chain devices without routing/NAT. Just like most IP phones do. So what happens if you try and chain a bunch of SPA-2100s? Does the 2100 act more like a router? The SPA-2100 will perform NAT between the LAN and WAN ports. It means that you won't be able to connect to devices behind SPA-2100 from the WAN side (except by configuring DMZ, but it's gets awful then). It also means that you can't chain them with their basic config, because it won't like having the same network address (192.168.1.x) on both its interfaces. So right now, if you want to chain them, you have to play with IP addresses, DHCP settings, etc. Not fun, particularly when you consider that this device is really plug and play, it remotes configure everything. Hopefully a bridge mode will appear in a later firmware upgrade (which, for Sipuras, are frequent and readily available on their website). -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote: We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN port. Only downside is that only 1 call can be using 729 at a time. This has been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100. My 2100s have 3 LEDs, plus 2 for each RJ-45 port. Instead of just 2 for the SPA-2000. -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
On Sun, Feb 13, 2005 at 10:39:36AM -0800, Luki wrote: The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) then you can be of great benefit. Otherwise they'll be worthless. I particularly miss the dial-plan, distinctive ring and audio gain options on the Grandstreams. Remote syslog can also be useful for debugging. It all depends what you need, I guess. Further, the Sipuras have a more detailed status, that is accessible WHILE you are engaged in a conversation. I think you're paying a bit more for the 1000 (1 line version) as compared to the Grandstream 286, but if you need/want two independent lines, then the Spa 2000 is more economical (as Peter said). The Sipuras are really a dream to manage, particularly in an international environment. You can customize the tones, the rings, the voltages, the dialplan, the features... well, everything. They are (securely) remote manageable and upgradeable. They are rock solid. Sipura support is helpful in case you need them for complex issues. Voice quality is top notch. The Grandstreams are less manageable, have less parameters, have only american tones, no dialplan support, no auto-upgrade (well, they recently added some kind of support). Voice quality is OK. -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
On Feb 14, 2005, at 5:39 AM, Nicolas Bougues wrote: On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote: We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN port. Only downside is that only 1 call can be using 729 at a time. This has been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100. My 2100s have 3 LEDs, plus 2 for each RJ-45 port. Instead of just 2 for the SPA-2000. I think Matthew was referring to the lack of leds on the front of the Sipura. I can't seem to figure out why these manufacturers insist on building these boxes like you're going to stick them on your desk next to your phone. I want something that's more suitable for a phone closet. Too bad the PAP2-NA can't be purchased retail anymore. Then again, you're probably better off with a Sipura-branded unit anyhow. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
Matthew Boehm wrote: [...] In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying one to test. Street price around US$ 90. Another one with dual g729 channels is MTA V102. Street price US$ 100. Also will test this one. I'm still looking for other units with dual g729 channels... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
On Mon, Feb 14, 2005 at 10:47:23PM +0900, Hermann Wecke wrote: Matthew Boehm wrote: [...] In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying one to test. Street price around US$ 90. Another one with dual g729 channels is MTA V102. Street price US$ 100. Also will test this one. I'm still looking for other units with dual g729 channels... Back in december, the Uniden was supposed to do 2xG729 at a later time. Not sure if the current firmware allows it. BTW, I've been fairly disappointed with Uniden firmware and their release cycle : their hardware is great, but they take months to release new firmwares, even when phone crashing bugs are discovered. If you want 2xG.729 now, working reliably, for under $90, you can't go wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is a bridge mode, where the LAN and WAN ports would act just like a switch, so that you can easily chain devices without routing/NAT. Just like most IP phones do. -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA's
Guys.. which ATA is better for connecting analog phones (features, stability, experiences, etc)? Sipura 2000 or Handy Tone 286, etc? What are you experiences? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
Anton Krall wrote: Guys.. which ATA is better for connecting analog phones (features, stability, experiences, etc)? Sipura 2000 or Handy Tone 286, etc? What are you experiences? In my experience the Sipura 2000 has three hardware advantages: * 2 independent phone ports * Mounting holes * The price for a single Sipura 2000 is less than the price for two Grandstreams. As far as software and compatibility with * goes, I only have experience in a LAN environment, where both worked (with the right firmware) without a problem. The Sipuras seem a little bit louder (or so the users tell me). -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) then you can be of great benefit. Otherwise they'll be worthless. I particularly miss the dial-plan, distinctive ring and audio gain options on the Grandstreams. Remote syslog can also be useful for debugging. It all depends what you need, I guess. Further, the Sipuras have a more detailed status, that is accessible WHILE you are engaged in a conversation. I think you're paying a bit more for the 1000 (1 line version) as compared to the Grandstream 286, but if you need/want two independent lines, then the Spa 2000 is more economical (as Peter said). --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
Good evening, allow me to join in right here. Which ATA/TA would you suggest for connecting analogue fax machines to Asterisk? One of the ones named before or e.g. a ATA-186 made by Cisco? Cheers Sascha The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) then you can be of great benefit. Otherwise they'll be worthless. I particularly miss the dial-plan, distinctive ring and audio gain options on the Grandstreams. Remote syslog can also be useful for debugging. It all depends what you need, I guess. Further, the Sipuras have a more detailed status, that is accessible WHILE you are engaged in a conversation. I think you're paying a bit more for the 1000 (1 line version) as compared to the Grandstream 286, but if you need/want two independent lines, then the Spa 2000 is more economical (as Peter said). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
Sascha E. Pollok wrote: Good evening, allow me to join in right here. Which ATA/TA would you suggest for connecting analogue fax machines to Asterisk? One of the ones named before or e.g. a ATA-186 made by Cisco? At the moment I am deploying Grandstream ATAs for faxing machines with out a problem so far. -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
-Original Message- From: Luki [mailto:[EMAIL PROTECTED] The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) Really? 87 pages aren't enough for you? http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ATA's
We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN port. Only downside is that only 1 call can be using 729 at a time. This has been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100. -Matthew - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, February 13, 2005 12:39 PM Subject: Re: [Asterisk-Users] ATA's The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) then you can be of great benefit. Otherwise they'll be worthless. I particularly miss the dial-plan, distinctive ring and audio gain options on the Grandstreams. Remote syslog can also be useful for debugging. It all depends what you need, I guess. Further, the Sipuras have a more detailed status, that is accessible WHILE you are engaged in a conversation. I think you're paying a bit more for the 1000 (1 line version) as compared to the Grandstream 286, but if you need/want two independent lines, then the Spa 2000 is more economical (as Peter said). --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
