similar to: Logging the CID from the Privacy Manager

Hi all, My latest Asterisk system is based on Debian squeeze with Asterisk 1.6.2.6-1 and SIP only. One of my favorite features that I had working with Asterisk 1.4 is the PrivacyManager. However, this was not straightforward, because anonymous SIP calls arrive with ${CALLERID(num)} = "anonymous", instead of being blank. So, to get it to work I added the first three rules to

Hi list, After wrestling with the voicemail system for a while (Asterisk 1.4.14, Debian etch), I got it to work, but I still have lots of questions, like: * Why can't I delete any voicemail messages? (Response: "Message undeleted.") * Why can't I listen to the messages in the Old folder? * Why can't I use the advanced options? (Response:

Hi, why would Asterisk core dump with the following test dialplan extension ? exten => 8100,1,Answer() exten => 8100,n,Set(CALLERID(all)="") exten => 8100,n,PrivacyManager() exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid) exten => 8100,n,NoOp(Number is ${CALLERID(num)}) exten => 8100,n,Hangup() exten => 8100,n(nocid),Playback(vm-goodbye) exten

Hi list, Has anyone been able to get ISDN-BRI support to work reliably on Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, kernel, modules, versions, config files). I've tried to get it to work on a Debian etch system with an HFC-PCI card and the zaptel package (v1.4.7, also from xorcom.com), but with no luck: all three channels that are created when the

Hi all, Previously, I had the PrivacyManager working for me exactly as would be expected, but after upgrading the OS to Debian lenny and Asterisk to v1.4.21.2 that's no longer the case. Anonymous callers are still confronted with the PrivacyManager, but now no matter what I set the minlength value to, e.g.: exten => jaap,n,PrivacyManager(1,1) ... (I'm not using a

Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA

Hi folks, Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1. As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can support IPv6. However, it seems that I can't get it to support both IPv4 and IPv6 at the same time. For example, if in sip.conf I set the bindaddr variable to '::' it will only listen on IPv6 and none of my IPv4-only

Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext.

Hi all, The kind of configuration that I use in my sip.conf to connect to various commercial SIP providers looks like this: [general] context=incoming-calls canreinvite=no qualify=yes register => jwinius:passwrd at sip.provider.com/0201234567 [provider] type=peer host=sip.provider.com fromuser=jwinius secret=passwrd This works. However, how would I

Hi folks, My client and I are having a problem getting a portable Esaote ultrasound machine to connect to a Samba server. The unit has an integrated laptop with a Windows XP version that can hardly be modified. Upon delivery the vendor only changed the user name and workgroup for us. When I asked for the user password to make a matching Samba account, the vendor refused because they use

Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I

Hi folks, Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider, so it must be that they no longer have the same basic requirements. The relevant part of my

Hi all, Perhaps someone here could help me with this. I'm new to Asterisk, but have already met with some success at getting my first system to work with two different VoIP (SIP) providers: XS4ALL and InternetCalls.com. The config for the former works fine, but my InternetCalls.com config works only intermittently for incoming calls. It currently looks like this: [general] port=5060

Hi folks, Recently I learned how to configure libvirt with USB pass-though functionality. In my case I configured my guest domain with this block of code: <hostdev mode='subsystem' type='usb' managed='yes'> <source> <vendor id='0x0c93'/> <product id='0x1772'/> <address bus='1'

Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS

Hi all, My Asterisk problem today involves getting a SIP client on a private net to register with a server somewhere else on the Internet. This worked for me about a year ago no problem, but now I see an error message on the remote server every time the client attempts to connect (the server is running Debian lenny with Asterisk 1:1.4.21.2~dfsg-3). Here's an example: [Nov 11

Hi list, Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run into some serious problems. The first thing I noticed was this message that would show up every five seconds on the CLI: Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 1 down

Hi folks, Has anyone encountered the following PCI passthrough error? error: internal error Process exited while reading console \ log output: char device redirected to /dev/pts/1 assigned_dev_pci_read: pread failed, ret = 0 errno = 2 It's produced after I've detached the PCI device from the base OS and have tried to start up the guest domain. To get to this point, I

Hi all, Is it possible to display or print variables in Asterisk (e.g. in the CLI) for debugging purposes? For example, I'm using two different types of SIP phones: the Snom M3 and the Siemens S675IP. However, when anonymous callers submit a number to the PrivacyManager, only the Siemens displays the new CID correctly; the Snom shows "unknown" (even though the new CID looks

Hi all, My problem boils down to these errors: ... Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time This is triggered by lines in extentions.conf such as: exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W) The system is CentOS v5.2 with Asterisk 1.4.23 (druid-asterisk-1.4.23.1-2), a Sangoma A104