Re: [asterisk-users] T.38 ATAs
We have probably 40 SPA2100/2102 that use T.38 to connect to a Cisco gateway for fax. Once we get the settings right, it seems to work probably 95% of the time. We had big issues with any of the 5.x versions of software and have stayed on 3.3.6 on both the 2100 and 2102. This isn't exactly what you are doing, but it might be useful info. Peder Steve Underwood wrote: Ian wrote: Hello I am going to try the new Digium Fax for Asterisk product. I'm planning to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has any experience with these devices, or other recommendations, I would be grateful if you could share your experiences. Some Grandstream firmware revisions seem to do T.38 fairly well. Others don't. If you use the latest Linksys firmware (5.2.x) the SPA2102 can do T.38 fairly well with some FAX machines and FAX modems, but its really quirky with others. My impression is the SPA3102 is quirkier than the SPA2102. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA recommendation (wih FTP provisioning)
I used the Patton M-ATA a year or so ago and it was a piece of junk. T.38 didn't work and there was no way to troubleshoot why it didn't work. Also, the web interface was horrible. Mike wrote: Hi, I am looking for a good ATA recommendation, ideally something: 1) with one FXS and one LAN port (so it's as inexpensive as possible) 2) That can be provisioning using _FTP_ (configuration and firmware upon reboot, ideally remote reboot from a sip notify) 3) Supports T.38 Nice to have would be: a) PoE powered and AC powered (my choice) b) Small size-wise I have been recommended the PAP2T in the past, and although I have used it and sort of liked it, it wasn't possible to provision it using FTP (at least as far as I could tell) Any tip is welcomed. I`m looking at the Patton ATA which is small, but it doesn't support FTP provisioning either as far as I can tell. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA recommendation (wih FTP provisioning)
Nope, I gave up and haven't gone back. Not worth the hassle. We use the SPA2100/SPA2102 and they work great for providing analog/fax lines as they support T.38. Olivier wrote: 2009/2/26 pe...@networkoblivion.com mailto:pe...@networkoblivion.com pe...@networkoblivion.com mailto:pe...@networkoblivion.com I used the Patton M-ATA a year or so ago and it was a piece of junk. T.38 didn't work and there was no way to troubleshoot why it didn't work. Have you tried again recently ? I've just tried its T.38 capabilities and I'm not successful yet. Also, the web interface was horrible. Mike wrote: Hi, I am looking for a good ATA recommendation, ideally something: 1) with one FXS and one LAN port (so it's as inexpensive as possible) 2) That can be provisioning using _FTP_ (configuration and firmware upon reboot, ideally remote reboot from a sip notify) 3) Supports T.38 Nice to have would be: a) PoE powered and AC powered (my choice) b) Small size-wise I have been recommended the PAP2T in the past, and although I have used it and sort of liked it, it wasn't possible to provision it using FTP (at least as far as I could tell) Any tip is welcomed. I`m looking at the Patton ATA which is small, but it doesn't support FTP provisioning either as far as I can tell. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phone 7940G.
Windows - http://kin.klever.net/pumpkin/binaries Works great. k4...@bellsouth.net wrote: I'm trying to do the same and have read the mentioned sites. The one item I can't seem to get past is a working TFTP server. What is the easiest method to get one running and what packages in Linux or Windows work best? Thanks for putting up with a Linux newbie. Ronny -- Original message from Alex Balashov abalas...@evaristesys.com: -- Have a look at: http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080 094584.shtml#topic2 On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S. wrote: I understand, but i cannot load the new firmware... is any well know method? On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov wrote: This phone is currently running the SCCP (Skinny) image. Before you will get anywhere you need to load the SIP firmware image onto it. The SEP* configuration files are for SCCP. After doing that, the phone will start requesting the correct files. You may need to upgrade through various SIP images cumulatively. On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S. wrote: Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get any luck here either. I look in the /var/log/messages and I observed that my phone request 4 different files that i don't have it in my tftp directory. Here's my tftp output session with my phone: Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to 192.168.1.3:52178 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml to 192.168.1.3:52180 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf to 192.168.1.3:52181 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to 192.168.1.3:52182 as you see my phone request 4 files that doesn't comes in archive P0S3-08-11-00.zip: SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf, SEPDefault.cnf... while my archive contents is the following: OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads, P0S3-08-11-00.sb2 3.) I want to make this phone to be SIP compatible. A friend of main gave me a .cnf file with an example of configuration for SIP. How may I rename this cnf file to make work with my phone. 4.) On the other side my phone doesn't have ringtone either. Any clue how may I put ringtones on it? I know is a lot of questions for you guys, but I browse on cisco.com web site and google for hours and I don't get it any clue to make work this phone in any way. Thank you for help. Jonson. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775
Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
1. Is it a technical reason that the ISP has restricted the upload to 512 Kbps or is it a Marketing reason that they have restricted the upload ? Marketing most likely. If they have fiber in the building, it would be a symmetric link. I have never seen a fiber connection that isn't the same up and down. They are just trying to sell it like DSL. 2. Can I boot the cisco switch in run level 1 and modify the rate limits on each of the ports ? 2950's don't support rate limiting, they are just semi-dumb layer2 devices. They have to be doing this upstream on a router somewhere. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [ANSWER] Re: Asterisk and CIsco 1760 SIP ?
I'm using an FXS, but it is pretty much the same. What part do you need? It is virtually the same as using a Cisco PRI card. Here is the relevant part on the Cisco. On * you just set it up like a gateway. dial-peer voice 200 voip destination-pattern .T progress_ind setup enable 3 session protocol sipv2 session target ipv4:192.168.1.10 (IP of * box) session transport udp dtmf-relay rtp-nte codec g711ulaw fax rate disable fax nsf 00 no vad ! dial-peer voice 1 pots description DID 1234567890 destination-pattern 1234567890 port 2/0 voice-port 2/0 caller-id enable Phibee Network Operation Center wrote: Anyone use CIsco 1760 with Asterisk Phibee Network Operation Center a écrit : Hi i am search a sample config (for asterisk and for cisco) for connect a cisco 1760 with a FXO card to my asterisk. Thanks for your help Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
You need a router with DSP modules for it to terminate voice, otherwise it is just a data BRI and a router that has SIP. 1600's don't have them and only the 1750-V/1751-V and 1760 support them on the 1700 series. Pretty much every Cisco router from the 1600 and up supports a BRI WIC/NM for data only if that is what you need. I don't think they even sell the BRI cards any more as nobody uses them. If you just need data, then you could also get an 802 or 804 as those are probably $50 total for router and BRI card, of course you can't do SIP/VoIP on them. Eric Chamberlain wrote: On Jan 27, 2009, at 9:49 AM, Michael Higgins wrote: Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have found. What I did find left me with the impression that USA 'BRI', uh, '2B1Q' protocol(?) is not supported by *any* hardware vendor, at all, period, nor is it tested and proved in the software... stack(?), in one related branch or another on the OS side. You might want to look into Cisco hardware, their WIC-1B-U cards work fine in the US, or they did 10 years ago when I last used them for VoIP. Used the WIC-1B-U is going for under $50 on eBay. An old 1600 or 1700 series router with an IOS that supports SIP wouldn't cost much either. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with sip registrations through HP Procurve 7102dl
Is it causing an issue? There are lots of firewalls that do nat and change the source port of packets to some random udp port. In my experience, for outbound registrations, it generally doesn't cause an issue. Robert Augustyn wrote: Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com http://www.linqone.com/ http://www.linqone.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
