similar to: HST Saphir with Asterisk

Thought I did already, but I've been pretty absent minded recently :) http://voipnow.com.au Or more specifically: http://voipnow.com.au/xcart/catalog/Saphir-III-ML-PCI-p-16151.html It's the HST Saphir III ML PCI. Price is on the website. Enjoy! James > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- >

Hi there, our old pbx contract will expire soon and we have now 2 pbx companies trying hard to sell us a new VoIP one. I am usually a friend of open source software and read about the Asterisk poject, but I can not assess if Asterisk fulfills all our requirements. It would be great to receive some input from experienced users to be able to convince our management. My question is if the

Hello all, I am trying to connect asterisk to a Siemens Hipath 3750 PBX system. I have a physical connection issue. I know that I should use a crossover RJ48 cable to link the two systems. The problem however is that the physical interface of the Siemens system is very unfamiliar. From my digging around, I think that this is an S2M interface. http://www.mail-archive.com/asterisk-users at

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |

I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more

Hello, I just noticed an error in the current version of Klaus-Peter Junghanns bristuff package, especially in the HFC module. Everytime I try to unload the HFC module with "modprobe -r" I got a kernel panic and the complete server hangs up so I need to do a hard reset. Regards, Julian Pawlowski

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

I'm having issues with compiling libvirt on OSX 10.12  ~/Development/github/libvirt/ [master] make /Library/Developer/CommandLineTools/usr/bin/make all-recursive Making all in . Making all in gnulib/lib /Library/Developer/CommandLineTools/usr/bin/make all-am make[3]: Nothing to be done for `all-am'. Making all in include/libvirt make[2]: Nothing to be done for `all'. Making all in

> I'd like to create the following: a user picks up the phone > (gets a dial tone), dials '0' for an 'outside' line, gets a > second (different?) dialtone, and is able to enter an > external phone number. Klaus-Peter Junghanns has something like this on his page: http://83.137.99.170/jn/relaunch/asterisk/page19.html It didn't work for me correctly so I

Hi guys, I am new to Asterisk. It looks great, but I have a couple of open questions such as - What softphones are recommended and provide Call Center level functionality? - What hardware (especially E1 (Euro S2M) class) are supported? I would like to search the list archives, but can't find a searchable archive. Can someone please point me into the right direction? Thanks. Regards,

Hi - Anyone using zaphfc cards (bri-stuff-0.0.2) on a Portugal Telecom BRI service? I am getting loads of errors and the audio is very distorted all the time. System is a Dell PIII/933 MHZ with SCSI. Here in the UK we have perfect audio with a Compaq PIII/500 MHZ SCSI system. Files are identical in both systems as below: Is there anything magic about Portuguese ISDN? The PCI bus in the

Hi all, I've run into an issue we have with dovecot versions 2.0.12 through 2.0.14 on latest CentOS 6.2 when dealing with Maildirs located on NFS partitions. Although I saw Mr. Egbert's thread, his resolution does not work on our end. When dealing with users with over 500 messages in any of their Maildir folders, they occasionally receive an error from the server: Internal error

Something I noticed on a 2.1.7 director test cluster (two directors, three backends): 'doveadm proxy kick user' will kick all connections for that user on that director only. Any additional connections on other directors will remain active unless the command is run on all directors. Are the proxy and director sub-commands intended to be separate and distinct in their operation? If so,

Try using the dlls from windows 95 cd, I'm not talking about the dcom95 dlls, but the dlls win95 installs in c:windows/system. > > > puoti@inwind.it wrote: > > Did you use only win95 dlls? If you used dlls from any other version of > > win, htere could be problems with dcom 95. > I used dll's only from dcom95 which are listed at appdb.winehq.org > > Without

Hi, I have a strange problem, after some time my asterisk server starts using more and more CPU resource and after some time it becomes unusable. I need to kill the asterisk process before it renders the whole system unusable. I have a few SIP phones, IAX trunk to other system and voicemail running on that server. Now, how do I troubleshoot, which part of asterisk is causing the load on CPU ?

Hi there, I just finished work on my current PBX dialplan configuration. I didn't find anything compareable yet so I think it might be interesting for someone else too. Short overview: - seperation of the numbers in blocks (just definded) - error/cause handling - advanced access control (even vor ISDN Zap clients) - access levels - denylist with national number blocks (for Germany) -

Hi there! I just try to play with die wake-up function described in http://www.voip-info.org/wiki-Asterisk+tips+wake-up Everything looks fine but there seem to be missing some soundfiles like "wakeup-menu". Where can I get these files in order to make this feature usable? Regards Julian Pawlowski

Hello all, the scenario: Carrier ----S2M------ * -----S2M------Siemens | | SIP Clients and many other features With much help from the list, the PRI links are without alarms and inbound calls are working fine (from both: Carrier and Siemens). But I am not able to dial wether outbound nor to the Siemens PBX. I allways get the message: == Everyone is busy

Hi- In trying to track down a possible memory leak in asterisk, I've discovered that the "show memory allocations" command crashes asterisk (causes it to stop handling calls, although it doesn't seg fault). The related "show memory summary" works however. Before I post this to the bugs list, can someone else please confirm this problem? You need to enable malloc

Dear List, I am having a problem with dates and I would like to understand what is going on. Below is an example. I can produce a date/time using as.POSIXct, but I am trying to combine two as.POSIXct objects and keep getting strange results. I thought I was using the wrong origin, but according to structure(0,class="Date") I am not (see below). In my example a is a simple date/time