Re: [asterisk-users] trixbox, sangoma a200, dell poweredge 2550 issue
Is there any reason why I should be experiencing such bad line quality on inbound calls from PSTN? Call quality is perfect when plugging in a regular analogue phone. So outgoing PTSN calls are fine but incoming PTSN calls have poor quality. Do both parties hear the crackling, etc? Can you reproduce this, ie if you use your cell to call in, does the problem occur every time? What happens if you call your PTSN number from your handsets? Does the call go out onto the PTSN and come back in with poor quality? Do you have other phone lines you can try the A200 with? Have you asked Sangoma support? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tftp issue
Jerry Geis wrote: I have xinet tftp running on centos 5.1 It seems to be running on the local network eht0 fine. My box has 2 nics. however when I connect to eth1 for tftp I get: in.tftpd[5084]: tftpd: read(ack): Connection refused How can I get tftp working on BOTH eth0 and eth1 for my phone config files. man page for in.tftpd says it automatically runs for all local networks on port 69. Is eth1 not a local network? How do I get tftp to response on eth1? Which networks are connected to each card? What is the IP/subnet of the client? You could be connecting through eth1 to tftpd, but the server is sending the packet back out eth0. The client would then refuse the connection as its coming from the IP address of eth0. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extenspy and chanspy
Brian J. Murrell wrote:
Does anyone have an implementation of this they'd like to share?
I cut out the authentication stuff we do, but this is part of the macro
we use to spy and record calls arbitrary calls. All of our users have
sip handsets. Asterisk 1.2.
exten = s,n(getext),Read(SPY,extension,4)
exten = s,n,GotoIf($[ ${LEN(${SPY})} != 4 ]?nospy)
exten = s,n(spy),UserEvent(ChanSpy,User ${CALLBACKNUM} spied on ${SPY})
exten = s,n,Chanspy(SIP/${SPY},r(monitor-ext-${SPY}))
exten = s,n,Hangup()
exten = s,n(nospy),Playback(sorry-cant-let-you-do-that3)
exten = s,n,UserEvent(ChanSpy,User ${CALLBACKNUM} failed to spy on ${SPY})
exten = s,n,Hangup()
___
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Re: [asterisk-users] Asterisk and the Mitel SX 200 integration
John covici wrote: OK, this is exactly what I would like to do, can you either write me on or off list for further details. This would be the first baby step toward the 20th Century!! I'd love some pointers on integrating * with a sx-200. I have a system where a fork lift upgrade is impossible. Ideally as we add new extensions, I'd like them to be in *, and have the mitel system know to route the calls correctly. I have a manual for the sx and a few half baked thoughts (put the pstn on *, and have the mitel system send all unknown numbers to *. Then * can route them properly to the outside world or to the new extensions), but it will be slow going. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk temporary hangs when no internet connection
Marius Muja wrote: My guess is that the asterisk server tries resolving the names of the SIP providers when it tries to re-register to them and because there is no internet connectivity it hangs there for a while. However in that time all the local calls to the asterisk server stop working. Try using a local DNS server. Sounds like its waiting on DNS lookups... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Bill Andersen wrote: a) IF... I expect a phone system to just work. Once it is configured, a phone system should just work with very little attention. My previous system was a Comdial with external voice mail on a DOS based PC. I LITERALLY WENT OVER 4 YEARS WITHOUT HAVING TO REMOVE POWER TO THE COMDIAL CONTROL OR RE-BOOT THE VOICE MAIL PC. I don't think you'll ever find the reliability of those old solid state systems. They have very few features and little to go wrong. I don't judge my asterisk installations against those systems. I judge asterisk against comparable systems, like a cisco voip systems or a Nortel BCM. While I can't say asterisk is more stable than they are, I can say the support is much better. With closed systems I never quite know whats going on and get told to reboot regularly to fix problems (the days of a BCM taking 20 minutes to reboot during business hours of a call center...). Or sometimes told the latest version will fix my problems, assuming I pay for it. With asterisk I can tackle the issues myself and get a continually improving product with no extra charges. The power of open source. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Bill Andersen wrote: Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? Anyone? Just a user? I'm just a user at work, I manage multiple asterisk installations, including our call center. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
Ex Vito wrote: On Tue, Feb 26, 2008 at 10:51 PM, Joshua Kinard [EMAIL PROTECTED] wrote: Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very likely, 380's as well). I just learned this the hard way. --J ...can you expand on that please ? I'm on my way to getting one of the newer Digium TE220B PCIe dual T1/E1 to put on such a system. So far I'm having nothing but problems with my DL360's and TE220B's. While many of the problems are slowly starting to seem to like problems with the telco, some of them definitely aren't. I can't pass the loop back test (patlooptest) unless the hpasmd (the system management software) has been stopped. Though I will try asterisk and zaptel 1.4 soon to see if that helps. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC errors
I have a suggestion. Have you contacted Digium technical support for assistance with resolving this issue? Excellent suggestion. Make sure you can give them SSH access and screen so you can see what they are doing. Before that, check (remake) your T1 cables and if it is punched down on a block, re-punch it. I'm used to vendors that aren't responsive, so I never even thought of it. They've told me to try running patlooptest (which I will tonight), to see if the problem is in the card. I received very good support from Digium. After running patlooptest they recommended that I return the card. During the loop tests I would sometimes pass a 60 second test. I very rarely passed a 300 second test on either span. I moved the card to the other slot and did some IRQ swapping as well to the same results. Thanks for all your suggestions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC errors
I have a suggestion. Have you contacted Digium technical support for assistance with resolving this issue? Excellent suggestion. Make sure you can give them SSH access and screen so you can see what they are doing. Before that, check (remake) your T1 cables and if it is punched down on a block, re-punch it. I'm used to vendors that aren't responsive, so I never even thought of it. They've told me to try running patlooptest (which I will tonight), to see if the problem is in the card. Thanks for your suggestions. Hopefully I'll learn something tonight. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC errors
Andrew Joakimsen wrote: I'll assume you chose trixbox to make your life easier when it comes to dealing with others regarding the PBX. Pretty much, yes. What is between the smartjack and your T1 card? What sort and length of cable? Any splices? Punchdown or patch panels? About 100 feet of RJ48 (yes, STP) which tests fine. Though I was thinking of moving the server to be in the same room as the telco box to see ensure its not the cables. Also I'm not sure if Trixbox has this but ssh in and see if there is an application called zttool. What are the statistics it is providing? I don't have any IRQ misses: IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 24/ 24/ 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC errors
You mention went into production, Did this imply moving of the system from a testing room into a server-location? Other (longer) cables? Unplugged the current system and hooked up a new, longer, cable to the asterisk system. The cable is RJ48 STP, about 100 feet. However we ran several cables and swapping them around doesn't make a difference; they all test good too. We could be having bad luck with them :-) I was thinking of moving the server to be beside the telco box, but that is a large undertaking. Perhaps you can check with your telco wether they receive bad frames coming from you I gave them a call and they'll run a report and get back to me. Hopefully the patlooptest tonight will point to the problem. Thank you everyone for all your suggestions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
