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Hi, I'm trying to get the latest version of darwine working with slickedit on my mac os x box. Everything is using the latest version. When I try to run my application I get this error: > wine: created the configuration directory '/Users/GMan/.wine' > err:module:DelayLoadFailureHook failed to delay load setupapi.dll.InstallHinfSectionW > wine: Call from 0x7b830f70 to unimplemented function setupapi.dll.InstallHinfSectionW, aborting > wine: Unimplemented function setupapi.dll.InstallHinfSectionW called at address 0x7b830f70...

...ur subnets network address and subnet mask. Are you recommending that I make it more restrictive? Thanks, Geoff > -----Original Message----- > From: Gene Willingham [mailto:gwillingham@comcast.net] > Sent: Tuesday, June 14, 2005 9:13 PM > To: asterisk-users@lists.digium.com > Cc: gmanning@zoom.com > Subject: RE: Call being answered, but no audio on either end > > > > I think I found the source of this. Been tracing it for a > week. Look in > sip.conf. It appears the definition of localnet has a > bearing on how some > sip devices handle invite...

I am trying to plot a lognormal density curve on top of an existing histogram. Can anybody suggest a simple way to do this? Even if someone could just explain how to plot a regular normal density curve on top of an existing histogram, it would be a big help. Also, is there some way to search through the R-help archives other than simple browsing? Thank you so much. Your help and time is greatly

Correction: The hardware is a Wildcard T100P (not a TE110P) Thanks! > -----Original Message----- > From: Geoff Manning [mailto:gmanning@zoom.com] > Sent: Wednesday, May 18, 2005 9:07 AM > To: Asterisk Users (E-mail) > Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX > <--Newb > > > I have been successful in setting up asterisk and making > workstation to > workstation SIP calls...

...better results at handling sip devices behind NAT devices. Gene > 19. Call being answered, but no audio on either end > (Intermittent) (Geoff Manning) > ------------------------------ > > Message: 19 > Date: Tue, 14 Jun 2005 17:30:31 -0400 > From: Geoff Manning <gmanning@zoom.com> > Subject: [Asterisk-Users] Call being answered, but no audio on either > end (Intermittent) > To: "Asterisk Users (E-mail)" <asterisk-users@lists.digium.com> > Message-ID: > <D1696C471C6CD511A0BE00D0B7A932DE0957C97C@southe01.zoomtel.com> >...

brett@websmyths.com wrote: > Also - an outside chance - make sure Tip and Ring > are correct. You could be getting ground loops - depends on the noise. > I am having noise and slip errors between my TE110P and a legacy PBX T1 card. Could this be the same symptom? The connection is made using a 15 pin serial on the T1 Card side to RJ48 on the TE110P side. I can't determine what the

Greetings all; I currently have a task to put together a SAMBA (3.2) server that can authenticate users to our local AD server. I was told recently that in order for that to happen, the authentication needs to be in "mixed" mode vice "native" (whatever that means), or it won't work. Can someone a bit more knowledgable than I confirm or deny this statement, or point me at

Say I had a crossover cable that connected a Mitel SX200 to a TE110P and the pinout was done as such: 1 - 4 2 - 5 5 - 1 4 - 2 (the 5 and 4 are transposed on the left side) Instead of the proper way of: 1 - 4 2 - 5 4 - 1 5 - 2 What would the results be? We have had the former as our cabling for a few months and the connection has been fine. Slip errors here and there. But we have had major

We use SetVar(TIMEOUT(digit)=8) In our dialplan to make sure that the user is done dialing before Asterisk executes the call. I just recently came across the piece I've copied below. It says for new incoming ZAP connections, the default digit timeout is 3 seconds and can only be configured in the source code. Is that true???? ============ How long will Asterisk wait?

Eric "ManxPower" Wieling wrote: >> >>> span=1,1,0,d4,ami >>> e&m=1-24 >>> > > Looks like you have told Asterisk to get it's timing from the Mitel. > I'll bet the Mitel is trying to get it's timing from Asterisk. > > Try span=1,0,0,d4,ami and run ztcfg -vvv > I just set this back. It was originally set to your

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

Our users are experiencing some unacceptable delay when trying to have a conversation. The delay is so noticeable that they keep stepping on each others words and resort to calling the customers via cell phone. Here is the setup SDSL Connection (PPPoA) Speedtouch 610 SDSL Modem 3Com 2224PWR Plus Switch (phones on separate VLAN) 8 Cisco 796 Phones All connecting to a remote Asterisk Server. We

I just installed Xorcom Rapid and I'm trying to connect with Xlite. In my SIP Proxy I have set the Domain/Realm and SIP Proxy as the IP Address of the new install. I can ping that box. When I try to connect I get hung on the "Awaiting Proxy login information" and the log reads: ======================================================================== ? 2004 Xten Networks, Inc. All

I have been successful in setting up asterisk and making workstation to workstation SIP calls. But I am lost when it comes to anything past that. We are trying to integrate this asterisk server into with our Executone (432?) PBX to allow us to make outbound SIP calls between our disparate locations. We have a T1 card in our PBX, and the Digium TE110P card in the Asterisk. We have the T1 card

I haven't been able to find much support for the RDNIS or DNID variables online. I am trying to prove a concept of call routing before we move towards development of a production system. I need to have calls routed coming into a call center based on DNIS. What type of syntax is needed in the extensions.conf file and how can I test it with a softphone (ie: can I emulate the DNIS from xlite)?

When I start Asterisk, I receive these errors: Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on peer 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'zoom_trunk' without zaptel

The best type of error possible, intermittent. We have PSTN numbers being switched to SIP then forwarded to our Asterisk server which sits inside our LAN Every once and a while (maybe 1 out of every 20 calls) goes like this: -- Executing Answer("SIP/213.199.36.50-0818e3e8", "") in new stack -- Executing Ringing("SIP/213.199.36.50-0818e3e8", "") in

Hello, I am not too familiar with the settings in our PIX (learning though). Here is the only access-list setting that we have in place for Asterisk: access-list acl-prod permit udp any host EXTERNAL_*_IP_HERE eq 5060 In rtp.conf we are allowing ports 10000 - 20000. We are not using SIP Fixup in our PIX due to firmware version. How do I go about adding the ability for udp ports 10000 - 20000

jglucky@blueware.net wrote: > Has anyone attempted or have had any issues getting Asterisk and > Digium cards working on a Dell SC420 server? I have it up and running on 2 SC420's with TE110P cards. Haven't had any problems as of yet (knock on wood). Dell SC420 TE110P 2x40 GB SATA drive in a RAID 1 setup (Linux 2.6 kernel needed for SATA drives; turn off RAID detection in BIOS).

Mike Clark wrote: > I am interested. We have just started using Nagios, so this could be a > nice add-on. > > Mike Clark > > Jeremy Melanson wrote: > >> If anyone is interested, I can send the script (it's in Perl) and an >> example of how to check the PRI status in Nagios. >> >> I'd love to hear from other people about how to use it to