Re: [Asterisk-Users] free sun boxes
Northern California, bay area. Tom Lynn wrote: Whare are they located? On 6/17/06, *Bob Knight* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, bk... -- Bob Knight [-w] the work option [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 925-449-9163 ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --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] free sun boxes
Northern California, bay area. Dovid Bender wrote: Where are they locater ? Dovid */Bob Knight [EMAIL PROTECTED]/* wrote: I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, bk... -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --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 Do you Yahoo!? Get on board. You're invited http://us.rd.yahoo.com/evt=40791/*http://advision.webevents.yahoo.com/handraisers to try the new Yahoo! Mail Beta. ___ --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 -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --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] free sun boxes
I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, bk... -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --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] logrotate and logger reload
I have one system that went totally crazy on me.
It went into an infinite loop rotating * message and log files.
From the asterisk console I kept seeing the message about re-loading
logger.conf over and over and it just kept creating more and more files.
I baby set many different * boxes all running the same script without
this problem.
Here is my cron script:
/var/log/asterisk/cdr-csv/*csv {
missingok
rotate 12
monthly
create 0640 root root
}
/var/log/asterisk/*log /var/log/asterisk/messages {
missingok
rotate 5
weekly
create 0640 root root
sharedscripts
postrotate
/usr/sbin/asterisk -rx 'logger reload' /dev/null 2 /dev/null
endscript
}
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
___
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[Asterisk-Users] Re: 4-port external sip fxo which doesnt suck?
For a box that has very poor reviews, it sure is great to use a box that you can throw in the closet and just forget about it. They just always work and sound great. The first time you configure one is a bit of a pain, but after that it is cruz time. I use a linux mib browser (mbrowse) because I work in an usoft free environment. I can drop ship a unit and have them plug it into the pbx lan and then configure it remotely. I find snmp more convenient than a browser interface. I have deployed quite a few Mediatrix 1204 and have never gone back and looked at any of them again. They just work. I'm looking for a 4-port external sip fxo which doesn't suck. o) Clipcomm CG-410. Poor reviews. o) Mediatrix 1204. Very poor reviews. o) Audiocodes MP104. Poor reviews. o) DLink DVG-3004S. Doesnt seem to exist yet. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --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] Polycom IP50X Park Softkey
I am now running sip 1.6.2 with a 2.6.1 bootrom. After moving from a 1.5 I now only see 2 softkeys at the main window: New Call and Forward. How do I get a Park softkey? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] v1-2 install mkdep loop
Just pulled a v1-2 onto a system that was running a v1-0. Zaptel and libpri, build and install just fine. Building asterisk is fine. But when I try to do a make install on asterisk, it goes into an infinite loop doing on .depend doing: build_tools/mkdep I did the same thing on another box the other day with a different pull and did not have any problems. Do you think this is something related to this box? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] polycom ip500 mwi, quite please
Does anyone know how to silence the audible mwi on a soundpoint ip500 or ip501 running sip 1.4.1? I tried changing just about all the se.pat.callProg.11 vars and nothing seems to change. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] dhcp vars, mediatrix 1204's
I have been deploying a bunch of sip gateways that I configure via snmp. I have noticed that a lot of the variables I need to set, can be set via dhcp. I like to just put common entries into my dhcpd.conf file, like: option some-variable-name some-variable-value example: option sip-server 192.168.0.1 option sip-port 5060 How do I know what some-variable-name should be in my dhcpd.conf file that will map to some snmp mib variable? I have peeked at the mediatrix mibs and docs and can not seem to find what I am looking for. I am guessing the dhcp client in the gateway is parsing dhcp packets, looking for option names. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Mediatrix 1204 Help
Message: 16 Date: Tue, 3 May 2005 09:12:13 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Mediatrix 1204 Help To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from. The firmware is not openly available. Mediatrix approach is to charge customers for every release they generate, and they only do that through approved resellers. If you know a company that resells their products, you might be able to twist their arm, but I'd guess they aren't going to give it away. (That's probably why it was being sold on eBay in the first place.) You will need the firmware that runs on the box (be sure to get the sip version), and you'll need the Windows-only snmp management software to configure the thing. Each firmware version has a specific snmp management package intended to be used with the firmware. You'll need both (matching) to accomplish anything as there is no telnet or web interface. No no no. Screw windows. All you need is the mib files and mbrowse. SNMP makes remote admin of these boxes a piece of cake. Much faster then a web browser. Once you figure out what you are doing, then you can just config and admin it with simple shell scrips, or if your a hack like me, c code. You can even use SNMP to monitor the PSTN line status. Way cool stuff and these boxes just run forever. If you are ready to give up on the boxes and want to dump them at a good price, just let me know. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] L2 QoS switch
I am looking for a switch that I can set up priority queues either on a per port bases or mac address. I really don't want to screw around with anything above L2 or routing. Something small (just a few ports) and cheap that I slam in just before the dsl or cable modem. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] small qos switch
I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from the voice lan and one from the data lan. I have found quite a few 24 or 48 port switches that will do this, but I really do not need anything that big. There are already switches in place. Any recommendations please? thanks, bk... -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Re: PSTN to VoIP FXO gateways?
-- Message: 3 Date: Mon, 3 Jan 2005 07:50:28 -0700 From: Damon Estep [EMAIL PROTECTED] Subject: [Asterisk-Users] PSTN to VoIP FXO gateways? To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99% reliable it will be a great product and I will gladly buy them. Now, what DOES work? Channel banks are a little pricey when you consider the need for PRI interfaces to * to use them. The solution seems to be inexpensive FXO VoIP Media Gateways, but there are only a few out there and fewer reports of using them with *. If you use one please share your experience! mediatrix 1204. rock solid. I have deployed several of these units and have never had to touch any of them. They just keep running. I would strongly suggest getting mbrowse running so you can do your initial set up and config via linux. This took longer than the initial mediatrix config. You can also crank up it's syslog to debug level to 5 if you run into any problems. This was helpful for me when I was first playing around with them. -- sipura 3k this unit has been pretty solid, but I must admit I still do not have it working the way I want. There are folks that are running their own code on this unit. It would be nice if someone would come out with an IAX version. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Re: Ethernet Channel Bank idea
On Wed, 15 Dec 2004, Matt Klein wrote: 3) good luck getting the firmware source is the firmware source freely available, -- I've been asked by others. All the other (excellent, thought provoking) conversation aside, Jake Messenger from Portmasters.com has been granted a license by Lucent for ComOS. http://www.portmasters.com/pipermail/comos/2004-August/41.html That contains a link to the license the source is under. It isn't free as in GNU, but I don't think that really matters much. I had to give up following this list too closely, because it just sucks up too much time. But I did just stumble onto this thread about portmasters. I worked at Livingston and wrote the drivers on the portmasters. That source code is easy to find and even compiles on a linux box these days (we used to use SunOS). If you come up with anything interesting to do with the boxes, please let me know I may be able to help. Contact me off list is best. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Asterisk on Solaris
Jongsuk Lee wrote: I am waiting for solaris 10 for x86. You can download 32 bit versions now. I just downloaded the sparc version. On Wed, 17 Nov 2004 09:53:31 -0900, Rich Allen [EMAIL PROTECTED] wrote: according to Sun, all Linux apps run under Solaris 10 ... would be interested in anyone who has actually done it You are going to have to wait for that. http://www.eweek.com/article2/0,1759,1724923,00.asp -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] IAX2 trunking - timing - ztdummy??
Håkan Källberg wrote: Hello! Perhaps someone can spread i little bit light on this: I want to trunk two Asterisk systems with each other. System A, behind a NAT-Firewall and System B with a real IP address. aix.conf on B: [mytrunk] host=dynamic username=mytrunk auth=md5 secret=yyy trunk=yes iax.conf on A: register = mytrunk:[EMAIL PROTECTED] When I make a reload an B I get the following: Nov 15 16:32:32 WARNING[-1244329040]: chan_iax2.c:6427 build_peer: Unable to support trunking on peer 'mytrunk' without zaptel timing I have downloaded the zaptel package, compiled it ( including ztdummy, which may be what I need ) and installed it. The kernel modules load: ztdummy 3492 0 zaptel228996 1 ztdummy crc_ccitt 2176 1 zaptel I don't know how to configure zaptel ( /etc/zaptel.conf ) to get this to work. I have no hardware, I only want timing for the IAX2 trunk ( and later on for Conference calls ). I have also read about the rtc package but have not tried it. I may have overseen very basic things... Please enlighten me! Regards: Håkan Try running zttest. Once zttest is working you should be OK. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Manager API Call Origination Variables
Peter Osborne wrote: Hi all, I am using the Asterisk Manager API to originate calls and it is working well, when a call is placed the local phone rings, once you pick it up you can here the call ringing the other end. Now, I am using Polycom IP 300 and I have them setup to auto-answer if I set the ALERT_INFO variable to Ring Answer. This works fine from my dial plan but I can't figure out how to set ALERT_INFO from the Manager API. Basically I want calls that are originated from the Manager API to automatically take place on the speaker phone. I have tried Action: SetVar Channel: sip/pete_desk Variable: ALERT_INFO Value: Ring Answer but it gives me about no such channel but this is the same channel I use to place the call immediately after attempting to set the variable. Any ideas? I have 2 extension entries for all my auto-answer phones. If you dial just the normal extension (like 1234) it does the normal answer thing. If you dial an * before the extension (like *1234) it does the auto answer thing. So you could just use: Channel: sip/*1234 or Channel: sip/*pete_dest -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] [OT] Old Building Needs a New Telephone System
Joe Greco wrote: Michael Welter wrote: We have a 100 year old building here in Colorado that needs a new telephone system. The building (five floors) is steel frame with lath and plaster walls. There is no crawl space above the ceilings or under the floors. The building is historic, and nothing can be done to the exterior. The current system uses existing Cat3 (two pair) to get to the digital telephone set in each office. Some offices have an additional pair which is used for fax (and DSL). I belive this fax line is a POTS line from the telco. The owners would like to replace the existing telephone system, but they are adamant that the exsiting wiring be reused. They would like to provide a LAN connection to each office for both data and voice. (They would also like to install cable TV in each office, but cable install costs would be $80,000+.) The owners are concerned about frequent power failures and keeping the telephones operational. Whatever equipemnt and telephone sets we put in the offices will have to be powered from a central UPS (PoE). So how can I do this? Can I use RS485 adapters to get ethernet to each office via the two pair? What kind of data rate can I get with RS485, and would it be half- or full-duplex? Would wireless work in a steel building? Is there some other technology that can be used? Ideas, anyone? It is real easy. EoV (ethernet over vdsl). I have done this and it works great. For every 24 ports I used a 1u EoV, 1u splitter, 1u fxs gateway. The little termination modems have ethernet and fxs. Just add an * box, done. I was under the impression that none of that stuff ran at 10Mbps or faster speeds. If he's got two pair and Cat3, he can just run 10Mbps Ethernet (and full duplex at that, if it's done right). Or has the short-range DSL stuff (which I know at least one local telco uses for in-house network extension purposes) finally beaten that speed? ... JG 15Mbps symmetrical -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] [OT] Old Building Needs a New Telephone System
Michael Welter wrote: We have a 100 year old building here in Colorado that needs a new telephone system. The building (five floors) is steel frame with lath and plaster walls. There is no crawl space above the ceilings or under the floors. The building is historic, and nothing can be done to the exterior. The current system uses existing Cat3 (two pair) to get to the digital telephone set in each office. Some offices have an additional pair which is used for fax (and DSL). I belive this fax line is a POTS line from the telco. The owners would like to replace the existing telephone system, but they are adamant that the exsiting wiring be reused. They would like to provide a LAN connection to each office for both data and voice. (They would also like to install cable TV in each office, but cable install costs would be $80,000+.) The owners are concerned about frequent power failures and keeping the telephones operational. Whatever equipemnt and telephone sets we put in the offices will have to be powered from a central UPS (PoE). So how can I do this? Can I use RS485 adapters to get ethernet to each office via the two pair? What kind of data rate can I get with RS485, and would it be half- or full-duplex? Would wireless work in a steel building? Is there some other technology that can be used? Ideas, anyone? It is real easy. EoV (ethernet over vdsl). I have done this and it works great. For every 24 ports I used a 1u EoV, 1u splitter, 1u fxs gateway. The little termination modems have ethernet and fxs. Just add an * box, done. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] polycom IP 500/600
Kristian Kielhofner wrote: Richard wrote: Hi Kristian, I'd like to use ftp because of several advantages it has. For example, ability to change the time stamp and reload the phone. But the default password is a big issue. I'd like to change it but don't want to go to each phone and reset it. Any way to change it? Thanks, I understand why you would want to use FTP (no filename changes). Why is the default password such a big issue? As a polycom user, it is the default username that is the issue. It is mixed case, something like Polycom. I think the good old tty drivers still support upper case only terminals, so as soon as it sees the capital P, it will turn on folding. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] meetme latency
I am pretty sure that I had used meetme in the past (many months ago) with great results. Small number of users, mixed connections, IAX2 and SIP. For the past month or so, meetme has been a real pain due to very large latency. I can take 2 phones on the local lan and still get many seconds of latency. This makes it really hard to carry on a conversation. If I try to have folks join in over the net, we end up with 4 to 5 second latency. Is this normal, or do I have a problem. I am running 2.6.8ish kernel with no zap hardware. I am using the 2.6ish ztdummy. zttest looks ok. Echo test and phone calls are great. I think it is only when I get into the pseudo zap driver that I start having problems. Is it time for me to check out app_conference? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Sending broadcasts to all phones?
Henry Devito wrote: I am writing this in C, well trying to write this in C. I will let you know when it is ready for testing. I found the solution in the WIKI to be clunky for the install I am proposing to a company that will have 250 phones and want to page through the phones with no overhead paging. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Monday, October 18, 2004 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sending broadcasts to all phones? Henry Devito wrote: I am in the process of writing an app to do this with Cisco phones7940/60. The feature on most PBX's is Page Groups, This allows paging through the speaker phones. This sounds interesting. Can I help in testing? Are you writing it in C or is it an agi script? I am also interested. I can help in coding and testing when you are ready to share. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] E3 PCI Cards
Steve Underwood wrote: Arinze Izukanne wrote: Hi Guys, Does anyone know of E3 PCI cards that work with Asterisk? Arinze No, but if you find an E3 PCI card with nice Linux support there might be people interested in helping to get it working with *. SBE (side band engineering). -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Jamie Carl wrote: Bob Knight wrote: There is a linux package called mbrowse that you can use with your mediatrix mibs. I can get and walk everything in my 1204's. For some reason I have not had any success with writes, but I have not spent that much time on it. I don't even have the MIBs which is half the problem. I can do certain things using windoze SNMP software, but not exactly being a guru on SNMP i'm guessing that without the MIBs i'm pretty much stuffed. Anyone with MIBs they can send me? hehe Please? :) I have MIBs for whatever version I am running that I am more than happy to share. Anyone know where I can place these for public access. Sort of like the freedomphones site for Polycom. We could then put pointers on the wiki. Thanks for the info tho. If mbrowse is console based it will be very useful. :) It has gui (X, gtk I think) if that is what you mean by console based. I can ssh into a remote * server and do get walks on my 1204's. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Jamie Carl wrote: Thanks to everyone for their help and comments on this. You've all been very helpful. I've actually got outbound calls working on it fine right now without having to change the configuration on the Mediatrix box at all, as I don't have the Unit Manager Software at the moment. Outbount seems to work well but without inbound it means I can't put it in place for general use. I have my 'reseller' tracking down the software for me right now so hopefully he'll be able to find it for me. :) Asterisk doesn't seem to have any issues working with the APA III-4FXO at all as yet. Thanks again guys. There is a linux package called mbrowse that you can use with your mediatrix mibs. I can get and walk everything in my 1204's. For some reason I have not had any success with writes, but I have not spent that much time on it. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Dell PowerEdge 750 rackmount
Scott Stingel wrote: Hi- I have an upcoming order for a bunch of asterisk boxes, and I'm considering using an assembled package for the server, instead of building them from components as I usually do. Does anyone have experience with the Dell PowerEdge 750 server, or any other 1U rackmount server for use with asterisk? Hey Scott, that is the exact box I am running * on in my office. But I have not been brave enough to plug in any PCI cards yet. I am still doing it the expensive way, with external gateways. I can't wait for someone like you to come out and say these PCI cards are way solid and ready for prime time commercial deployment. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Mediatrix 1204 - Error: Operation not permitted
[EMAIL PROTECTED] wrote: When I try to make a call to PSTN via Mediatrix 1204 I received the error below: Aug 7 21:01:48 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x482082ec (len 430) to 192.168.199.5 returned -1: Operation not permitted There are here anyone that knows what I can do to correct it ? Crank up the syslog debug level to 5 on the 1204. Even if you do not have a syslogd running (but you should) you can still read all the ascii messages with ethereal. This will provide pretty good debug messages. When you are done debugging, I would suggest dropping the level back down to 4. It gets a little verbose. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] HELP! With Postresql
Martin Keding wrote: I am having some real problems with getting CDR records to go to a Postresql database. I think I have followed every post and instruction available and Asterisk still happily writes to a text file. Postresql is installed and working on a Redhat 9.0 box, the same one as Asterisk. I have created the CDR table in a database called Asterisk. Conf files etc are set. I even recompiled Asterisk. Any pointers would be greatly appreciated. In file /var/lib/pgsql/data/pg_hba.conf uncomment the line: hostall all 127.0.0.1 255.255.255.255 trust In file /var/lib/pgsql/data/postgresql.conf change line to: tcpip_socket = true Add -i to option in your /etc/init.d/postgresql export PGOPTS=-i Not sure if all that is needed, but it did get my FC2 linux 2.6.6 running. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Cisco ATA 186
Gonzalo Gasca wrote: Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ labs softphone, i have the most recent Asterisk version, but when connecting to the PSTN i have choppy voice problems, not internally just when connecting with my Mediatrix gateway and ATA, my SJLabs softphone works ok with Mediatrix any ideas? Any working configuration? Turn VAD off on the 1204. * can not clock itself. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] PRI dead in USA?
brian wrote: Well they fail to realize that ISDN is used for more than data. I just wanna scream at them and say IT DOES VOICE TO YOU NINNY!.. Rates are far from reasonable. 167/mth here is what I would have to pay for ISDN-BRI. SBC is lame. Back in the day, Pacbell was pretty lame also. I worked at a place that made isdn routers. We had a cheat sheet we used to give customers so they could tell pacbell how to provision their line. I had several BRI lines at just $28 per month. I would stack up the B channels and run MLPPP. We allowed users to cheat and make data calls look like voice calls. I think the speed went down from 64 to 56 when you did this, but you saved some per minute phone charges. The good old days. The phone company never seemed to really want to deal with isdn back when it was cool. Now with dsl, they must really ignore it. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Bounty! For help with echo cancellation code.
[EMAIL PROTECTED] wrote: From the CLI and during a call I want to be able to: *** Pulse the outgoing line and record at least 50 ms of the incoming line. The pulse waveform must be specifiable as a series of amplitudes for each 1/8000 sec time slot. It would be best of these values could be read from a file specified on the CLI command line. Timing should be synced between the pulse and the echo so that the delay from the pulse to the echo can be accurately determined. Echo cancellation should be disabled during this operation. This would operate similar to the echo-training code that operates at the initiation of a call except that this could be done at any time. The initial pulse and any echoes can be combined and saved in a single channel. Output should go to a file and should be in a simple format that a program such as Audacity can read, display and play. *** Pulse the outgoing line and record at least 50 ms of the incoming line. Same as above EXCEPT echo cancellation would not be disabled during this test and the results of the echo cancellation operations should be recorded and saved in a separate channel. *** Change variables used to control echo cancellation. Only the code in mec2.h is of interest. I will help identify the variables and modify the mec2.h code as needed to accomplish this goal. There are a lot of parameters in mec2.h that may affect the quality of the echo cancellation. I want to be able to adjust them 'on the fly' and be able to immediately hear the results. I am open to alternative proposals which would accomplish the same goals. Name your price. How about being able to see the results real time? I use a package called SMAART from siasoft.com. It is a dual channel spectrum analyzer. Run the output line as your reference channel and the input line as your measurement channel. You can get great info from the impulse response and transfer function. You could also use this to compare different codecs. The impulse function will tell you how long it takes. The transfer function will tell you just how good a job it did at reconstruction the original audio. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Bounty! For help with echo cancellation code.
At least one of us that have worked a fair amount with the echo problem tends to believe the issue is system related as opposed to pstn line issues. Off-list, we found that swapping motherboards does have a very noticable impact, and processor speed does not appear to be a consideration. (Kind of thinking the echo (or feedback loop) is actually internal to the system.) Would the SMAART package help if this is the case? It probably would not help, but it sure is fun to play with. SMAART compares any 2 signals. If you pump a signal into a black box and then compare the output to the input, it can show you what the black box did to the signal in both time and frequency domain. It will show you phase response, impulse response and transfer func. Bad news, it is not open source and does not run on linux. They do have a free 30 day demo version you can download and play with. It can make real pretty pictures on the screen. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] mediatrix 1204 hysteria
Jair Martinez wrote: I read that some of you installed mediatrix devices with a SIP server and it worked OK. Could you please tell me which SIP server you used, and how did you configure it on the 1204? The SIP server is called something like asterisk. The only problem I had with 1204's was having to use a damn windows box for config. But now that I have it working with mbrowse on linux, the universe is in balance again. My office is back to a totally microsoft free environment. I did have to use the windows box to grab the mib files off the 1024's cd. For some reason I can not read that cd on a linux box. Anyone know why? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Wake Up Call AP
Stuart Baggs wrote: Can someone please tell me what sound files to record to get wakeup.agi to work? I'd recommend William Hung's version of She Bangs. If that does not wake up up, nothing will. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Do people actually answer questions here?
Leif Madsen wrote: On Tue, 29 Jun 2004 11:08:43 +1000, Jean-Yves Avenard [EMAIL PROTECTED] wrote: I have to admit I'm rather disappointed with Asterisk, information is probably available but very hard to find ; it seems to be limited to a few privileged people for whom their job is setting up VoIP system Based on your statement, I would presume that you have never even attempted to search for documentation. I can think of at least 3 excellent resources: http://www.voip-info.org http://www.fnords.org/~eric/asterisk/ http://www.asteriskdocs.org Plus using site:lists.digium.com and site:voip-info.org in Google is an excellent resource. Don't forget the most important link of all at the bottom of every email. The unsubscribe link! -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] DSP Coding
My thoughts on a DS3 * box: Forget PCI. Forget x86. There are very good bsd and linux ports for the powerpc. There are ppc's with very good TDM interfaces. All these framers and dsps speak TDM. Very simple clean design. If you do not want to build any hardware, you can probably find something off the self. You can always use an eval board from IBM or Moto. Any expensive but easy way to start. The only pain would be the * port. Yet more ifdef's. OK, that is a different rant. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] polycom soundpoint ip500 help
I just received a shipment of ip500's. They came with no documentation and a cd with a bunch of windows stuff on it. I could not find any config or load files on the cd. No problem. I found a pointer to config and load files via the wiki. Fired up the phone, gave it a static ip and watched it asking for tftp files. Copied the files in to place. It successfully download new boot and sip. Now the only thing it will do is just send out CDP packets. No display (other than initial polycom logo) or keyboard response. The folks at polycom explained to me that the reseller should be helping me, but they tried anyway. We were never able to bring it back to life or a factory default. Some how I seem to have turned on the cisco switch and do not know how to get it back. We tried all the magic multi button pushing and hand shakes. Anyone have an suggestions? Can anyone suggest a good polycom reseller that will provide boot and sip load images? I sure like the way these phones look and feel. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Re: Fedora Core 2 and Kernel 2.6
Maron Kristófersson wrote: Running asterisk on gentoo 2004.0, kernel 2.6.5, 2.8 Ghz hyperthreading CPU 1G RAM. I decided to use kernel 2.6 after reading about problems with hyperthreading and asterisk in 2.4 on this list. So far I've only connected to VOIP service providers and everything has been working very well. I will however connect a PRI line in the next 3-4 weeks so I'm interested in hearing from experienced kernel 2.6 users as well. I'm also interested in getting in contact with people using asterisk as a hotel pbx, which is my setup (100 rooms in 3 locations, 1 asterisk box). If you hit a wall trying to get intel based boxes to do the job, let me know. I am working on a SunOS port. It would be fun to see this running on a Sun Fire server. Should be able to scale it to 1000+ rooms. Only problem, servers run from about 50k to a million. That's like real money. But it would still be fun. btw: this is not a very pretty port. The current state of the * source tree does not lend itself very well to other OS's. Quite a bit of hacking involved. Something that I would never want to see checked into cvs. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] ATA devices
Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, mediatrix 1124 -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Spring VON Wrap Up
Bob Klepfer wrote: Mark Messmore, Technical Support, University Telcom Inc. wrote: K...maybe this was stated earlier in the conversation...but what's the deal with the phone? Or was this phone just being carried around by everyone and ripped apart? Old Bell Phone + IAXy + 802.11b card + Batteries = Homemade WiSIP, IIRC After a peek under the hood, I would guess we could have these manufactured over seas for around $1000 USD per unit. It would not be the same to modify the design in any way. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Informal Astricon at the VON Show in Santa Clara...
Steven Sokol wrote: Anybody out there either going to the Spring VON show in Santa Clara or live in the Bay area? I'm trying to put together an informal Astricon as an after-hours event for Asterisk users. Mark (and presumably Greg and Malcolm) from Digium will (tentatively) be there. I was hoping I could get a head-count so I can find a venue of the proper proportion. I _think_ the event will be either Monday night or Wednesday night, since Jeff Pulver has a huge party scheduled for Tuesday. If you would be interested in getting together, please let me know. No obligations, just a rough estimate so we don't wind up packed into a tiny bar or something. 2 more bay area * nerds ready and willing to participate and help in any way: Bob Knight in Livermore Todd Taylor in Tracy -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Music on Hold sound goes off if environment is silent
Ernest W. Lessenger wrote: At 08:37 AM 3/11/2004, you wrote: Music on hold works if the environment is noisy. But in case of silence the sound goes off. If I scratch continuously on the mikrofone, then the replay works without any interruption. Q: is there a parameter which influences this behaviour? Whatever phone or softphone you are using, you need to disable silence suppression. Why? Dunno exactly. In the newest version of Xten, the feature is Advanced System Settings - Audio Settings - Silence Settings - Transmit Silence - Should be Yes. Why? Because the * community is just a little on the lazy side. * can not self clock RTP packets. Instead of clocking itself and just locking on to received packets, it totally relies on received packets for it's timing. No packets coming in for timing, no packets going out. This would be something fun to work on, but who has time when there are work arounds. I am unemployed and I do not have the time. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Hotel wake-up
Nicholas Bachmann wrote: Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? It seems like it could be accomplished with an AGI and a script that wrote call files. Have the AGI prompt for the wakeup time (or have a web interface for a front-desk person do it) and write a file to a directory indicating when the wakeup call should occur. Then, have a Perl script that goes through those files and generates a call file in /var/spool/asterisk/outgoing at the right time. Call files make retries simple as well, allowing you to space them and choose how many you want. If you wanted to get fancy, you could use a database (perhaps with triggers?), voice recognition, or mp3s for the user to wake up to. Good old at job may be able to help with this (man at). -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Anybody going to the Spring VON converence [ OT]
Not sure if I will attend VON, but myself and a friend would be way into an * nerd fest. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Anybody going to the Spring VON converence [OT]
Roy wrote: Here's the web site for the convention http://www.pulver.com/von/ The convention center has conference rooms and breakout rooms. I bet if you asked nicely, you could get one for an asterisk BOF Yeh, but what kind of beer do they have on tap? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Stuck TE410P cards
Scott Stingel wrote: Hello all- I have 3 TE410P cards in service in the field. Two of them have an regular problem that they get stuck during a system reboot. What I mean is that they display no LED's during any part of the restart, and they are not seen by the drivers during or after the reboot. The only thing that brings them back to life is to power down and restart the box they are in. Even pressing the reset button on the processor does not clear their state. This sounds very much like a hardware problem with the cards, since one would assume normally that a front panel reset would clear a stuck card. Has anyone else experienced these symptoms? This happens fairly regularly on two of the three TE410P cards. It does not happen with older cards such as the E400P, of which I have several. Do pci read cycles show anything in the slot? Does pci id come back as all 1's or 0's or just some invalid number? Gee, the price on those sip gateways don't seem quite so high now. have fun, bk. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Snom 100 Code Recommendation
Jason Ross wrote: G'Day, I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm having DMTF problems no matter what configuration I try. And as yet I haven't downgraded it to see if an earlier release makes a difference Just wondering if anyone can provide some guidance as to what the best release of code for this phone may be. I also have DTMF problems with Snom 200 running 2.03o, but haven't had the time or desire to dig too deep into it. I am running p2p with a sip gateway, so * is not in the picture and I have never changed code or reconfigured my gateway. I guess I have just been waiting for 2.03x release of the day to see if it gets better. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] s/asterisk mailinglists/asterisk forum/g ?
Good thing I am unemployed, so I have time to read this list. Every morning when I suck down my 200 emails from the list, I say to myself, I am going to implement some filters to help sort all these emails. But after blasting my way through the email, I am out of time and energy. Anyone have any filters they use on this list that may help me out. I have never set up any email filters. I run on a sun/sparc solaris 9 and use mozilla to read my email. A linux solution should be easy to get working on solaris. I know I should just learn how to do this myself, but I am too busy reading email. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Sip flow diagram?
Rich Adamson wrote: Does anyone have a high level flow diagram showing acceptable sip messages exchanges? For exampe: Source Dest Invite - -Trying Ok - I'm specifically trying to debug an issue with various hangups, prior to call completion, after call completion, calling vs called party hold, etc, and getting rather confused watching the various packets flowing between sip devices with a sniffer (and no reference document). Rich It may be a little verbose, but you can find it in the rfc 3261 as a start. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] OT:Linux(or *BSD) SNMP tools (Was: Re: rtp sound quality?)
Chris Craft wrote: On Saturday 31 January 2004 21:31, you wrote: CHOP I am just a low level c hack. Before I go out and write any thing to do this snmp admin stuff, are there any linux tools I could use to do this? Net-SNMP (http://freshmeat.net/projects/net-snmp/ , formerly UCB-SNMP or something) is very handy for this. Perfect. Thank you very much. bk... -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Mediatrix 1204 SIP FXO 4-port gateway review
Rich Adamson wrote: Product Review Mediatrix 1204 4-Port SIP FXO Gateway Firmware: v2.4.10.69 - US Version US Retail: ~$750, Street Price: ~$450. Trouble shooting is limited to the SNMP manager only. The manager can be used to view configuration data, however needed dynamic operational statistics are limited to mib2 definitions only. For example, when trying to determine the souce of choppy MOH sound, I wanted to check the Ethernet port speed. There was no mib variable defined for this purpose. I found the syslog feature pretty niffty. You crank the syslog up to level 5 and get a lot of info. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP gateway question
Rich Adamson wrote: The 1204 then sends one more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? That would be RTCP (RTP + 1) 3. Has anyone played with this box and found any unusual problems, weird config's, etc? I have several of these boxes in use at a few different sites. Once installed, I have never gone back in and looked at any of them. They just work. I have it running in canreinvite mode and all sip phones running p2p. The poor * box has really no work to do. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
Rich Adamson wrote:
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9NXX,2,Congestion
[trunk-toll]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Congestion
--
Bob Knight
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Re: [Asterisk-Users] rtp sound quality?
Rich Adamson wrote: Thanks Bob, that fixed it. Any other hints/issues/default values that I should muck with, or is that about it? Seems like it works pretty good; excellent echo cancellation, etc. I haven't done anything with the box as yet for dialing outbound. Anything to be concerned with, special parameters, etc? I can't think of anything off the top of my head. It has been a while since I set mine up. My one and only complaint so far with this box is the snmp config stuff. They only give you a windows version. I have no windows boxes in my office. I just thought some day I would have to slam together a few little snmp scripts or gui code that drives off their MIB files. But I never had to go back into the box to do anything, so this has been a low priority. I am just a low level c hack. Before I go out and write any thing to do this snmp admin stuff, are there any linux tools I could use to do this? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] [ot] Grandstream hardware
Scott Stingel wrote: What *I* want to know is why someone has not made a CHEAP PCI card with 4, 8, or 16 of these DSPs on it. This kind of card would provide Expanding a bit on Nicolas' message, DSP software is complex, and there is not a huge number of people who do it well. So along with the board layout and production cost (not trival for a 6- or 8-layer board), you have the programming cost for both the PGA (programmable gate array) device(s) and the DSP. You also have the cost of the DSP simulators, driver development etc etc. All of these must be amortized over the number of boards you expect to sell - that's why the board price can get so high. Dialogic's D600-2E1 JCT boards etc cost well over US$1. The whole point of the asterisk/digium exercise is to move the complex software to the PC and take advantage of the economies of scale that it brings. Don't forget power and HEAT! When I was making Portmasters at Livingston/Lucent we made modem boards with a bunch of DSP's sitting on TDM's. Some of those DSP's are great BTU generators. Some times you have to clock the DSP at slower speeds just to keep the heat down. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] E100P - Error 500
Daniel Bichara wrote: Hi, I am running * with E100P board. At least every our I got an Error 500 message and ISDN-PRI restarts: Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 12:54:27 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 13:33:01 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 13:33:48 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Any clue? Unknown error 500 is an ELAST return code from zaptel driver. It is telling libpri that there is an event in the queue. If the read/write routines see that there is an event in the queue, it just returns ELAST. Libpri needs to do an ZT_GETEVENT to clear the event and should do some error handling if needed. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Mailing list growth
Mark Spencer wrote: I still think we need something more fine grained. I think we can add the asterisk-biz list, and eventually something akin to a newbie list, but need a more appropriate name, IMHO. like an asterisk-virgin * for the very first time Now lets see how long it will take you to get that tune out of your head. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] Cisco to Cisco - poor quality
I have never used Cisco phones, but I have had problems in the past relating to * RTP talking to a widget with VAD turned on. * RTP stack can not run on its own. It relies on receiving RTP packets for doing its timing. A simple test is to sniff the line to make sure the phones always send packets. If you see pauses, you may need to disable some type of VAD setting on the phone. Or just never quit talking when using the Cisco phone. Terence Parker wrote: I have set canreinvite=no in the sip.conf for each user (well, there are only two) using a cisco phone. What does this imply? As for whether the problem is due to the phones or asterisk however, indications would suggest both, because: - Voicemail works fine (and is clear) - I can initiate a call between MSN and Cisco, and that would sound fine. This might suggest a problem with my phones. However : - When using Vocal previously, Cisco to Cisco conversation was fine. This has led me to be completely stumped! I notice some mention elsewhere about asterisk lacking certain codecs because of license restrictions? Is this anything to do with me? Or should the phones still - in theory - be able to talk to each other without any problems? I have tried the cisco phone on both g729a and g711ulaw. I'm currently *trying* to get ahold of an updated firmware for my phone. I will see if this fixes the problems. Thanks again, Terence -- How are the phones talking to each other? Directly, or through asterisk? (canreinvite=what? in the sip.conf for each of them?). What I'm trying to get at here is, it is a problem between the phones, or are you having a problem possibly with the asterisk box? Some other things to know: are you running voicemail yet? If so and you can dial into it from either of the phones, how does it sound? If not, how about anything from the * boxlike the demo annoucment stuff? Daryl - Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream Quality Survey.... :P
Is that FCC sticker on the back of the phone for real? A customer could not use his computer while talking on his GS BT102 phone. The customer was using a major name wireless keyboard/mouse with his pc. The keyboard/mouse stops working if the GS phone is too close. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: time to build an open phone?
Bill Schultz wrote:
ACES - Asterisk Communications Endpoint System
{the following could be used by any IP-PBX but the name pays homage to Mark Spencer and friends who
cannot be lauded enough for their fine work}
As you read this it will be obvious I am not a professional engineer but I do have enough knowledge
to be fairly certain what I'm proposing is feasible from not only an engineering, but production
cost and perhaps most importantly, marketing standpoint.
An open phone is a great idea but as soon as you get physical you add a quantity issue that
doesn't exist in software. Multiply this for keypads, handsets, bells, etc. etc. etc. and you have
a lot of work but more importantly NO ONE has built a phone that can simultaneously be brain-dead
simple to operate for one person yet offer the advanced user whatever functionality they might
want. You will never solve that issue as long as you have a keypad of any kind.
An open phone is open. It does specify any type of I/O device, only how
to interface to them.
We just start with something like a light weight netbsd/* code base.
Folks can add whatever from there.
So you end up with what started this open-phone thread in the first place... a plethora of IP,
analog or digital phones with a dizzying array (or lack thereof) of bells and whistles all trying
to achieve a balance between quality, ease of use and functionality which will sell enough units to
make their manufacturing and distribution profitable. In this environment you will always have at
the low end manufacturers competing on price and inevitably that results in quality issues. Right
now it's Grandstream but next year it'll be someone else at a $30 price point and the same issues
will apply all over again.
I have no interest in trying to make money by manufacturing widgets.
I only want control of my own destiny.
I don't care what the phones cost. I just want control of the code.
I've never seen stats, but it's probably a safe assumption that the majority of IP phones are
sitting next to a PC and the additional expense has been incurred because people want a phone that
looks and works like a phone. That's certainly been my experience far outweighing any technical
issues with quality or reliability of a PC-softphone. In every market I can think of with the
possible exception of hospitality I think ACES could be successfully sold a substantial number of
times even though it does not look like a phone because it affords a much better way to resolve
the conflict between ease of use and functionality. For the unconvinced, a more elaborate version
could include the obligatory keypad and cosmetic plastic but I would submit that the ability to
pick up a handset and place a call by saying call Pat alone would sell most potential customers
on learning how to operate a two position switch on a device that doesn't have a conventional
keypad. At it's simplest, to use the phone you need to know that position A is used to hangup and
dial by saying dial 1-800-555-1212 (or whatever number you want called) and position b is used to
talk.
The markets I work in not only do not want to use pc's as phones.
They do not want voice on their data networks.
Some of my customers, including my office have no pc's at all.
Just unix work stations.
Of course, I could always be wrong :-)
I would not say you are wrong.
You are just looking for something different than I am.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
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Re: [Asterisk-Users] time to build an open phone?
Bruce Ferrell wrote: only problem is the protocol stack isn't open. Good chip We would only use that code for examples of how to bolt in the bottom end drivers. We would roll out our own os/scheduler, a little * code and drivers. I have not found a data sheet for the 1001 yet, but I did look at the 1050. Great looking chip. Just a few questions. Any idea how much it cost? It does have a jtag debug interface. Do you know of any gui debuggers running on linux for this chip? We really need a nice friendly debug environment to make it as easy to write/load/debug code as doing it for linux. CW_ASN wrote: How about to build an ip phone with this IC? http://focus.ti.com/docs/apps/catalog/general/applications.jhtml?templateId= 969path=templatedata/cm/general/data/bband_ipphone_tnetv1001 - Original Message - From: Bob Knight [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 4:30 PM Subject: [Asterisk-Users] time to build an open phone? Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] time to build an open phone?
Open software seems to work. Why don't we try it with hardware. 1. pick an embedded processor. It should have a nice linux gui support (like x jtag debugger). 2. pick a linux based cad system we all have easy access to and place schematics under cvs. 3. pick some type of gpio or serial interface for keyboard/display. 4. pick some basic functionality. 5. code it up. A stripped down *. Let everyone do their own thing with the expensive part. Tooling/packaging. We could let Digium be the distributor, so they are not left out of the loop. A board set would be offered with NO support. If Digium wants no part of it, we just build them on our own for our own use or sell them on ebay. What we would provide is schematics and source code. Everyone can take this to their favorite fab house and crank em out. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
Dawid Mielnik wrote: Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for the telephone. Something that would combine the functionality of a (adsl modem+) router and a SIP telephone adaptor in one box. I would appreciate any info that you might have on this. regards, Dave Take a peek at Intertex IX66+PF or look for anyone coming out with a TI AR7 based solution. It's like a $25 single chip solution. I would expect to see boxes in the $100 - $200 price range soon. If you find anything else, please let me know. I am starting to play in mid to large cat 3 environments and doing the BLEC thing bolted up to *. Very swt set up. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
John, did you ever get any feedback from the GS wish list? I love the BT-102's with 1 exception. The speaker phone. I have not come up with a combination that makes it acceptable. If I had a way to cover up that button I would go ahead and deploy the phone. But the db level and echo to the far end user makes it unusable. If anyone on the list has successfully configured and used the GS speaker phone, could use please share thanks, bk. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
Walker Haddock wrote: If anyone on the list has successfully configured and used the GS speaker phone, could use please share Great fix, replaced with Cisco 7960 Almost the same fix as mine, Snom 200's. Now I just need to fix the bottom line. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Party in Paris
Is the party at the Paris Hilton? sorry, couldn't help it... -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk freezing HELP
TeleSIP wrote: UPDATE: We were able to consistently reproduce this problem using a Grandstream phone with buggy firmware. Mark Spencer logged into our Asterisk and identified the issue. He said it was a typo in an ast_mutex_lock. After fixing it, the problem seems to have been solved. We have now repeated about 100 calls and no lockup (with the buggy phone we were able to lock it up in under 7 calls). CVS should now reflect his fix. And by the way, do not use firmware 1.0.4.18 on GS phones. It contains a nasty SIP Port bug. Regards, Andres. One bug I found on my GS ATA adapter. I had it pointing to my * server for ntp. I did not have ntpd running. You could start a call, but during the call it would do a ntp request. The server was sending an ICMP message back and then the adapter would terminate the call. I fired up ntpd and all is well. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
John Todd wrote: At 11:04 AM -0800 12/5/03, Bob Knight wrote: Greg Boehnlein wrote: On Thu, 4 Dec 2003, Bob Knight wrote: Steve Dolloff wrote: I would be seriously wary of putting a DS3's worth of voice traffic on a TNT. I don't believe they are rated to handle that much voice. The APX1000 would be a much better platform, but I don't know if you can find one used. Skip the TNT's. They are really a joke. I will admit, I am a bitter X-Livingston employee. First Lucent bought us for our cool gear, then they bought Ascend for sales and marketing.. I still can't believe they kept the TNT alive and killed PM4. The PM3 LIVES ON DUDE! :) I'm all about Livingson, and have refused to put the Asscend stuff in my data center. Seriously, Jake over at portmasters.com is doing some good stuff with the PM3. Now that we've got control of ComOS, it is just a matter of time before new ComOS releases start coming out for the unit. Several people have already rolled their own and added a few niggling fixes to the 3.9.1c1 code branch. It would be great if we could find a way to use the PM3 as an inbound channel bank for Asterisk though. I have like 7 of them sitting in the back doing nothing.. I like that idea. I wrote all the drivers for the PM3 and it would fairly easy to do. Looking at the prices on portmasters.com, you could have a 2 t1 inbound channel bank for about $350. Add another $150 for an extra t1. I think we used the same Dallas framers that Digium uses. I am a very big * fan and I am feeling a little guilty that I am using an ethernet only solution. No Digium cards. I would really like to support Digium, but I do not want to start pluggin any PCI cards into the box other than an extra ethernet or 2. I would love to see Digium come out with a t1/e1 to ethernet channel bank. Compared to when we made the PM3 there are some way cool processors with built in TDM and ethernet. Yo Digium, I am hanging out here in CA with nothing better to do than play with *. Why don't you contract out and let me and a few of my unemployed friends build a little channel bank for you. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 Bob - You make two good points: 1) The PM3 might be an interesting and inexpensive TDMoE Device, or maybe even a stupid IAX2 channelizer. I suspect that Digium will not help you with this unless you allow them to be the exclusive reseller, since this takes away from their core business of selling cards. However, even with a bit of a markup, this would still be a pretty decently priced multi-T1 solution, as long as the used market can reliably offer these devices at good pricing. 2) On the larger discussion, a separate device that provides T1 termination in a more dense footprint than a PC is obviously showing some interest, as judged by the number of followup posts on this list to my original question. There are two devices that I see as useful: - an FXO and FXS selectable solution, via RJ11 or Centronics-style bus connector, in a 1u package that delivers IAX2 out (or, sub-optimally, TDMoE) Options for this would be built-in codecs. Pricepoint: $1100 (the cost of a T100P and a well-equipped channel bank.) To be successful, this device _must_ support FXO and FXS. Fail-over dialplans for 911 or other failsafe dialing methods would be good (typical in such devices.) There exist already devices that fit this description, though they are only SIP or H.323, and they tend to be way too expensive. - a high-density T1 termination system that can handle 8 T1's in a very small amount of rackspace. DS3 de-muxing onboard would be optimal, since anyone with 8 T1's is probably getting a DS3 delivery method, and removing the M13 mux from the rack would be great. Optimally, a 1u rackmount with T3/E3 coax _and_ 28 RJ-45 connections (only 17 of which would be used for E3/E1 muxing) Out of this unit would come IAX2 or (sub-optimally) TDMoE packets to Asterisk peer(s). This solution quickly gets into the discussion of why you might need SS7 for large installations, but I will not address that here, and we'll assume this is all PRI delivery. JT I would really like to see both of these devices. I would buy both of these devices. I do not want to build and sell these devices. I want Digium to build and sell these devices. I want Digium to contract out to me to help them bring these to market in a timely fashion. OK, I am just looking for a way to make a little money, ie unemployed nerd. This would be so much easier to build with todays processors compared to what we had to work with when we built Portmasters. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
Greg Boehnlein wrote: On Thu, 4 Dec 2003, Bob Knight wrote: Steve Dolloff wrote: I would be seriously wary of putting a DS3's worth of voice traffic on a TNT. I don't believe they are rated to handle that much voice. The APX1000 would be a much better platform, but I don't know if you can find one used. Skip the TNT's. They are really a joke. I will admit, I am a bitter X-Livingston employee. First Lucent bought us for our cool gear, then they bought Ascend for sales and marketing.. I still can't believe they kept the TNT alive and killed PM4. The PM3 LIVES ON DUDE! :) I'm all about Livingson, and have refused to put the Asscend stuff in my data center. Seriously, Jake over at portmasters.com is doing some good stuff with the PM3. Now that we've got control of ComOS, it is just a matter of time before new ComOS releases start coming out for the unit. Several people have already rolled their own and added a few niggling fixes to the 3.9.1c1 code branch. It would be great if we could find a way to use the PM3 as an inbound channel bank for Asterisk though. I have like 7 of them sitting in the back doing nothing.. I like that idea. I wrote all the drivers for the PM3 and it would fairly easy to do. Looking at the prices on portmasters.com, you could have a 2 t1 inbound channel bank for about $350. Add another $150 for an extra t1. I think we used the same Dallas framers that Digium uses. I am a very big * fan and I am feeling a little guilty that I am using an ethernet only solution. No Digium cards. I would really like to support Digium, but I do not want to start pluggin any PCI cards into the box other than an extra ethernet or 2. I would love to see Digium come out with a t1/e1 to ethernet channel bank. Compared to when we made the PM3 there are some way cool processors with built in TDM and ethernet. Yo Digium, I am hanging out here in CA with nothing better to do than play with *. Why don't you contract out and let me and a few of my unemployed friends build a little channel bank for you. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
Steve Dolloff wrote: I would be seriously wary of putting a DS3's worth of voice traffic on a TNT. I don't believe they are rated to handle that much voice. The APX1000 would be a much better platform, but I don't know if you can find one used. Stephen Skip the TNT's. They are really a joke. I will admit, I am a bitter X-Livingston employee. First Lucent bought us for our cool gear, then they bought Ascend for sales and marketing.. I still can't believe they kept the TNT alive and killed PM4. -Original Message- From: Ernest W. Lessenger [mailto:[EMAIL PROTECTED] Sent: Thursday, December 04, 2003 4:51 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Port density: DS3 cards? At 02:34 PM 12/4/2003, you wrote: However, considering the traffic volumes that you are talking about, is it really true to say that the traditional telco cards are astronomically priced, given the amount of revenue that can be generated per month on a DS3? Eight quad-span T-1 cards from Digium: $8,970 Three reasonable-quality asterisk servers: $1,000 One T-1/DS-3 MUX: $5000 Total system cost: $14,970 That actually sounds quite reasonable to me. However, if I were doing this myself I would look hard at getting a MAX TNT with VoIP capability off eBay. The price would be equivalent or less, the interface would be more complicated, but all the DSP would be done by the MAX. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer via # on Grandstream not always working
Anton Yurchenko wrote: Hello, After a while the transfer on grandstream stops working, only the reboot fixes the problem. It also seems that it may be the phone I`m trying to transfer _to_ also sometimes requires a reboot. After that it starts working. I`m using RFC2833 signlaing between phones and *. Does anybody see this happening also? Thanks When I first started using GS phones with *, I tried RTP signaling and had a problem with bouncy keys. I switched to SIP signaling and all is well. From what I can remember looking at the sniff traces, it appeared to be an * bug, not a GS bug. But SIP works well.. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip speaker phone for hands free intercom
Has anyone used the speakers on sip phones as part of an intercom? Are there sip messages you can send a phone to simulate key strokes, like someone hitting the speaker phone button on a GS? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax name resolver
I have a few * boxes spread around at different locations with different ISP's. I have 1 location with a static IP, the rest are all dynamic and all are NAT. I can tell when ever the remotes have a change of IP from looking at the IAX registrations and now know the new IP. I was thinking of letting the static box keep track off all the dynamics and host an IAX name server. Before I go off and so something really silly and a waste of time, is there an easy way to do this? Has anyone done this with DDNS? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to demo * on a notebook
Chris Albertson wrote: One other idea might be to use the USB FXO interface but I don't know how well it works. Some peole here have complained about sound quality We use sip fxo gateways via enet. Works great. We go in and give demos on their phone lines. --- costas [EMAIL PROTECTED] wrote: I want to be able to demo * on a notebook at a client's site. This means no FXO gateways; just 2 sip phones (like SNOM) and maybe a softphone (GnoPhone?). I already have RH9 running on my notebook. I would like to have one SIP phone dial and go through IVR before making a choice and ringing the other phone extensions. Of course the notebook would have to be running Asterisk. How can i setup one of the SIP phones to be the outside caller and go to IVR? What would the outside phone's dial out plan do. I assume the configuration files affected would be extensions.conf and sip.conf. If someone has an example of a couple of lines of .conf would be appreciated. Thanks -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 Double Digit problems
I can't afford Cisco phones, but I'll tell ya what I see with GS phones.
They seem to be bouncy as hell.
I'll hit a key and see 4 - 6 rtp dtmf event messages.
I am going to try and just debounce this in * and see what happens.
John Todd wrote:
Hello -
I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having
problems with routines that input long strings of numbers, in that I
am getting more than a small number of double digit entries. As an
example, I have a section that asks for the user to enter a call
forwarding number, and then puts that number into a database. Almost
always, there are double digits when the user only intended to type a
single digit, no matter how carefully they entered their string.
Can anyone comment on how they may have solved this issue with Cisco
devices? The units in question are running 2.16.
JT
snippet of code where I'm inputting the number - line has already
been Answered
[class4.6]
exten = change,1,ResponseTimeout(5)
exten = change,2,Playback(special/edting-spd-dial-number)
exten = change,3,SayDigits(${SPEEDDIAL})
exten = change,4,Background(silence/1)
exten = change,5,Background(special/entr-nmbr-fr-spddial-entry)
exten = change,6,Background(special/and-prs-pound-whn-finished)
exten = change,7,Background(silence/3)
exten = change,8,Goto(5)
; strip off any extra pound or * symbols, and then set the variable
exten = _X.,1,GotoIf($[$[${EXTEN:-1:1} = #] | $[${EXTEN:-1:1} = *]]?2:4)
exten = _X.,2,StripLSD(1)
exten = _X.,3,Goto(1)
exten = _X.,4,DBput(${MYNUMBER}/FEAT/SPEED/${SPEEDDIAL}=${EXTEN})
exten = _X.,5,Goto(class4.5,verify,1)
exten = t,1,Goto(change,5)
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Re: [Asterisk-Users] ATA-186 Double Digit problems
Sorry all you GS phone fans. I did not read the trace correctly. Looks like it was * that cause my double digit. A hunting I will go.. Bob Knight wrote: I can't afford Cisco phones, but I'll tell ya what I see with GS phones. They seem to be bouncy as hell. I'll hit a key and see 4 - 6 rtp dtmf event messages. I am going to try and just debounce this in * and see what happens. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem with * RTP timing? Sip phone to sip phone works fine. I connect 2 GS and place one on hold. The GS that is receiving MOH from * is working great because the GS keeps sending back RTP packets. IAX connections work fine. I call an extension on another * box and place it on hold. MOH over IAX/IAX2 is great. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
Juan, thank you very much. Turning off VAD did it. All is well. Juan J. Sierralta P. wrote: On Wed, 2003-11-19 at 16:10, Bob Knight wrote: I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem with * RTP timing? Its known problem, Asterisk SIP channels get the timing from the source, so if the source stops transmitting (i.e. VAD) the MoH gets choppy. Try disabling VAD on your Media Gateway. When VAD is active it is usually signaled by an specific RTP payload type, maybe the SIP channel should check that an starts using a local clock. Sip phone to sip phone works fine. I connect 2 GS and place one on hold. The GS that is receiving MOH from * is working great because the GS keeps sending back RTP packets. IAX connections work fine. I call an extension on another * box and place it on hold. MOH over IAX/IAX2 is great. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange Music on Hold between SNOM, Grandstream and Asterisk
I have the exact problem with a little different configuration. I am using GS phones and a sip/fxo gateway. I make a call from PSTN == sip/fxo gateway == * == sip phone (GS). Life is good until GS places call on hold. What I see on the wire: 1 - RTP packets every 20msec both directions for a second or 2. 2 - Then 2-3 second pause. 3 - Then a few RTCP packets. 4 - then back to 1 Are you using ztdummy? John Brown (CV) wrote: Hi List, Here is the config ext 2601 is a GS BT-101 phone ext 2062 is a SNOM 200 latest public firmware on both asterisk is Asterisk CVS-11/14/03-22:55:45 Make a call from 2601 - 2602 life good, call works have 2602 place call on hold. The music on 2601 IS NOT my music on hold. It seems its a MOH server SNOM has. take call off of hold on 2602 and 2601 still trys to play parts of the music from SNOM's server. Make a call from 2601 - 2602 life good, call works have 2601 place call on hold, SNOM plays my music but its real choppy and doesn't play well. have two GS's call each other and MOH works, not choppy, etc. So questions are: 1. how do you get the SNOM to use Asterisk as the MOH source ? 2. how does one get the music to not be choppy when a GS places a SNOM on hold john brown ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
What a timely subject. I am setting here trying to bring up a 1204. I receive a sip invite from the 1204 but * is returning 404 extension not found. I am a newbie to * and am still fumbling around with config files. Could you please save a few of us a little time and share your * config files relating to the 1204. thanks in advance, bk. Sean P. Robertson wrote: Ryan Tucker wrote: I have used the Mediatrix 1204 to terminate a POTS line. It does work OK. I've had some problems with caller ID not showing up all the time, but otherwise it's been pretty solid. The configuration, however, was perhaps the most horrible VoIP-related task I've ever done. -rt We are Mediatrix's US distributor and have used them with Asterisk in our lab and have had several resellers purchase them to use with Asterisk. They seem to work well with Asterisk, but I have to agree that the configuration leaves a lot to be desired. Their SIP units use SNMP exclusively and the way that their MIB is arranged, it is a little like configuring a Windows PC via the registry editor. Thankfully their are only 6 or so settings that need to be changed from the default to get it working so once you know where everything is, it is not that bad. One truly embarrassing issue that the current FXO (1204) units have is that they are using SNMP v1 and can not be password protected in any way. A new version of the firmware will be out in a couple of weeks and will support SNMP v3 and will have password protection. Hopefully they will come up with a web browser configuration in the future. Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204
Thanks for the reply Sean and look forward for more. I do believe I have the 1204 configured ok and I am able to place outbound calls (from * to PSTN). I think my only hang up is some type of * extension config on incoming calls. * 101 type of stuff. I am still just learning. As a side note. I found (with help from the Mediatrix folks) that the getwalk feature was a great tool for configing the 1204. I just looked at the output for all the nat.0.x addresses to see where to plug in my nat.* address. That was my biggest hang up with the 1204. Now it is * config time. I really like the syslog feature on 1204. I have the logging cranked up to a level 5. Now I just have to figure out what all these messages mean. Sean P. Robertson wrote: - Original Message - From: Bob Knight [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 8:54 PM Subject: Re: [Asterisk-Users] Mediatrix 1204 What a timely subject. I am setting here trying to bring up a 1204. I receive a sip invite from the 1204 but * is returning 404 extension not found. I am a newbie to * and am still fumbling around with config files. Could you please save a few of us a little time and share your * config files relating to the 1204. thanks in advance, bk. Sure. I just saw another reply to this come in and he has a good start on the Mediatix config steps. I will get together a list of some of the other Mediatrix configuration parameters and the Asterisk relevant config files that will work for you and email them to you tomorrow. Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400p loading errors
I finally received a phone call from Silicon Labs. They left a voice mail saying they were going to email me a data sheet for Si3210. I have not received it yet. As soon as I do and I get a little free time I will kick the chip around a little and try to narrow down the problem. A few questions: 1. Has anyone received a new (since sept 1) tdm400p card that works? 2. Why isn't digium looking into this? OK. Now it is time for me to go back to my full time job of trying to find a job. Azher Amin wrote: Hi, I have received a new card TDM400P revision E, from digium. When I tried to modprobe wcfxs it gave me the following errors: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 255 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 255 should be 2 Module 1: Not installed ProSLIC on module 2 insane (1) 255 should be 2 Module 2: Not installed ProSLIC on module 3 insane (1) 0 should be 2 Module 3: Not installed /lib/modules/2.4.20-8/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wcfxs.o: insmod /lib/modules/2.4.20-8/misc/wcfxs.o failed /lib/modules/2.4.20-8/misc/wcfxs.o: insmod wcfxs failed I have another TDM400P revision C (few months older) which works perfectly on the same slot of the system. The machine is AMD750 and I have tested several other cards and they worked fine. Plz suggest me about this problem and how to correct it. Regards Azher Do you Yahoo!? Yahoo! SiteBuilder http://us.rd.yahoo.com/evt=10469/*http://sitebuilder.yahoo.com - Free, easy-to-use web site design software -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk cvs commit list
Is there an asterisk cvs commit email list? Any project I have ever worked on in the past, always had a cvs commit email list. Anytime someone does a commit you receive the file name and comments. You can then make the decision if want to update or not. It can also help you narrow your focus when someones commit has broke your tree. thanks, bk -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Problem
I may have the same problem.
When I try to load wcfxs driver, it fails after reading register 8 on
the ProSLIC.
See following log:
kernel: ProSLIC on module 0, product 0, version 5
kernel: ProSLIC on module 0 insane (1) 0 should be 2
kernel: Module 0: Not installed
Can anyone point me to a data sheet on this ProSLIC device.
I would like to dump the regs and kick it around a little to see if it
can do anything that makes sense. It is aways hard to tell when you
are bit banging on a device.
btw: digium support suggested taking the mounting bracket off.
Did not work for me.
Steve Totaro wrote:
I had the same thing and just
figured it out yesterday!
the problem is that the tdm400p is
failing calibration. type "dmesg" and it will tell you.
uncomment in zaptel/Makefile KFLAGS+=-DNO_CALIBRATION in the
source code and "make clean install"
It worked for me but I wonder if
there is a bad batch of cards? I was in the "backordered till Sept
2nd batch" I am assuming its not good to fail calibration?
Steve Totaro
-
Original Message -
From: How Peng
Kaiam
To: [EMAIL PROTECTED]
Sent:
Thursday, September 11, 2003 9:52 AM
Subject:
[Asterisk-Users] TDM400P Problem
Hi,
Just received the TDM400P and
X100P.
PC can detect the X100P, but not
the TDM400P.
Tried to load the wcfxs module,
reported:
modprobe wcfxs
/lib/modules/2.4.20-20.9/misc/wcfxs.o: init_module: No such
device
Hint: insmod errors can be caused by incorrect module
parameters,
including
invalid IO or IRQ parameters.
You may find more information in syslog or the output
from dmesg
/lib/modules/2.4.20-20.9/misc/wcfxs.o: insmod
/lib/modules/2.4.20-20.9/misc/wcfxs.o failed
/lib/modules/2.4.20-20.9/misc/wcfxs.o: insmod wcfxs failed
Any advise.
Thanks.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
