Re: [Asterisk-Users] TDM04B vs Dell
If this is a single CPU system try loading the smp kernel "yum install kernel-smp" then force this to boot in grub.conf This will force your interrupts to be handled with IO-APIC handler instead of the old style XT-PIC interrupts handler. Switching to APIC from XT will normally fix a lot of interrupt issues. If you have them. On Wed, 2005-01-05 at 17:35 -0800, Michael Swan wrote: > At 06:53 PM 1/5/2005 -0600, you wrote: > > > I've struggled for several days trying to get a Digium TDM04B 4-port > > > wxfco card working on a Dell 1U PowerEdge 750 machine running > > > Fedora Core 1. I finally got a call back from Digium who indicated that > > > there is a fundamental conflict between the card and the PowerEdge > > > having to do with PCI interrupts. Asterisk version is stable v1-0 > > > 12/29/04. > > > >That sounds a little hard to believe. > > I agree. Perhaps I have too much faith in Digium support. Does > anyone else disagree with Digium's assessement? > > > > > The symptoms of the problem were as follows: > > > > > > 1. issue "modprobe zaptel" which immediately returns with no feedback > > > > > > 2. issue "modprobe wcfxo" which returns > > > init_module: No such device > > > Hint: isnmod errors can be caused by incorrect module parameters, > > > including invalid IO or IRQ parameters > > > > > > 3. issue "modprobe wcfxs" which immediately returns with no feedback, > > however > > > the four lights on the card go on and then the machine locks up > > completely, > > > requiring > > > a power cycle to get it running again. After the power cycle, if I look in > > > /var/log/messages > > > >If you have a tdm04b, that says you have four fxo ports. Why are you > >trying to load wcfxs? > > Actually, I tried "modprobe wctdm" which was supposed to load the > correct TDM driver and this resulted in the same behavior described > above (lights on, system locks.) In an attempt to figure out why the > system locked up I subsequently issued a "modprobe wcfxs" to > confirm that was causing the problem. > > > > > I see a long cycle of the following messages before reboot: > > > kernel: Dazed and confused, but trying to continue > > > kernel: Do you have a strnage power saving mode enabled? > > > kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 > > > > > > 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. > > > > > > In any case, I did follow the setup instructions on the Digium site (make > > > install in /usr/src/zaptel, edit /etc/zaptel.conf, edit > > > /etc/asterisk/zapata.conf, etc.) > > > and we currently have a X100P wcfxo card in another machine running well > > > so we've already had experience getting a card working. > > > > > > If anyone has insight into what might be wrong, please do let me know. > > > Ultimately, if I trust the Digium support information, then this card will > > > never work, so I'd be grateful to hear about any other PCI card that > > provides > > > four or so wcfxo interfaces that might work with the PowerEdge. > > > >I don't use Fedora, but it seems those that do have had problems > >loading the drivers. Try the modprobe wcfxo then zaptel, then check > >your /proc/interrupts. If that doesn't work, try modprobe zaptel only. > >I think someone mentioned a readme in the src/zaptel directory for > >Fedora as well. Might look. > > > > Thanks for the advice. However, "modprobe zaptel" didn't > do anything (that I could tell) and "modprobe wcfxo" returned > the error. And, greping for Fedora in src/zaptel didn't turn up any > matches. > > Michael Swan > Neon Software, Inc. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * + Codecs + Hardphones??
Quick question what happens if you go over your channel licenses? Mark Spencer wrote: So it looks like the best codec is the GSM codec as far and badwidth vs voice quality, but I can't seem to find which hard phones support the GSM codec or if * supports the G.729 codecs or others.. Which phones do the * user commumity find work the best?? and which codecs do you use?? You can purchase G.729 from Digium at $10/channel. Contact Greg Vance (256-428-6262). The G.729 is currently considered "beta". Mark ___ 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] Cisco 7960 (SIP) & XML
More then likely your web server is sending the phone some environment variables It does not like. The Cisco phones XML parser is very very picky. There is a perl module specifically for Cisco phone Cisco::IPPhone; You should be able to find it in cpan, if not google will. Mike Reiling wrote: I know this isn't really related to *, but I thought some of you might be able to shed a little light. I have a 7960 running SIP. No matter what I send the phone, I get CMXML parse errors. I have the services_url set to a perl app that I keep changing with no luck. I have set the correct content-type. Does anyone have example XML that I can look at. I don't have the cisco call manager. Thanks, Mike ___ 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
[Asterisk-Users] Does anyone know.
Out of band DTMF for SIP seems to be broke. I tried switching to dtmfmode=inband this works fine for local phones, but any phones over non LAN link, can not enter digits without duplicates showing up, this is most sever for the user name prompt in voice mail main. Is anyone working on getting out of band DTMF working again? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Ouch ... error while writing audio data: :Broken pipe
I just moved all my configs to another box. made no changes, did the exact same compile, And it worked. I have no ideal at this point what "file is to large" and causing the segment fault. I did have to move back past the "(no Nat)" code. for other reasons, but I'm up and working. I'll check the latest CVS again in a few days. when I have time to trouble shoot any errors. Mark Spencer wrote: I'm unable to duplicate your problem. Can you contact me off-list with some output so I can see where it's dying for you? Mark ___ 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] ParkedCall and SIP.
I have the "Tt" option on the "Dial", is there something else I needed to do to get "#transfer" support working? It does not. exten => 3874,1,Dial(SIP/3874|20|tT) I first tried to emulate park with the SIP transfer. I would transfer the call to a meetme, and pick it up later, You had to be careful not to park more the one call in one room. I however had to abandon this method of parking because the proxy software I use sends a retransmit request on hangup. So hangups always fail with Asterisk. Not a big deal with voicemail or calls. Because Asterisks timeout kicks in and does hangup the call. But in the meetme rooms there is no timeout! This caused a large amount of fake calls to build up and consume massive amounts of bandwidth. Because Asterisk never dropped the calls. She keeps sending RTP to far end. Just my little test setup with four phones could saturate a T1 in no time flat. On that note, could we have "Hangup" drop SIP calls Without waiting for the last ack? I other words drop the in progress call when "hangup" is run after a small timeout regardless of weather it ever gets a ack back from the phones? This would at least get me something close to park. Mark Spencer wrote: SIP does not yet support parking unless you do "#transfer" support. The reason is that once you have done a transfer in SIP, the original call is gone, so there is no way to announce where the call has been parked. Mark On Fri, 7 Mar 2003, James O. Sizemore III wrote: I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and SIP. To use it I should be able to hit "#" then get a prompt? ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ParkedCall and SIP.
I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and SIP. To use it I should be able to hit "#" then get a prompt? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
