search for: umrk

...esponse: Remote host can't match request NOTIFY to call '1742fb0c598dc6ec444a5ea64a2103ae at 192.168.10.10'. Giving up. The IP address belongs to my server. At the same time, I used tcpdump to see what else might be going on. I found the following: 19:18:22.540113 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 545 E..=.... at ...... .........)..NOTIFY sip:1000 at 192.168.10.5:5060 SIP/2.0 Via: SIP/2.0 19:18:22.571452 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 308 E..P.......f....... .....<a_SIP/2.0 481 Call Leg/Transactio...

...ied) D 0 UNKNOWN After some tests, I found out that the SPA3102 is indeed trying to register, but that Asterisk seems to be ignoring it. Using tcpdump, I can see that registration packets are regularly being sent to the Asterisk server (bitis): 15:30:49.567288 IP spa3102.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 482 Eh........%........ .......xREGISTER sip:192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 12 15:30:49.568390 IP spa3102.umrk.to.sip-tls > bitis.umrk.to.sip: SIP, length: 492 Eh........%x....... .......%REGISTER sip:192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 12...

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, 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 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 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, 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 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 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, 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 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 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 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 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 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, 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

Hi list, My Asterisk v1.4 system now has two ISDN channels and two SIP channels. The idea is to make a dialplan that mostly uses the SIP channels for outgoing calls, but I'd like those to fall back automatically to ISDN if the SIP channels aren't available, possibly in combination with a warning issued to the caller before the call is actually placed. Is this possible with

Hi list, After recently setting up voicemail for Asterisk 1.4.14 on my Debian etch server, I noticed that I can't delete any old voicemail messages. The voicemail menu option "Press 7 to delete this message" is available, but when I press 7 the response is always "message undeleted" and the message is still there. What could I be missing here? Thanks, Jaap

Hi list, The default file name format for touch monitor (automon) recordings is: auto-${EPOCH}-caller-calee It's possible to use the ${TOUCH_MONITOR} variable to change the 'caller-calee' part, but what about the 'auto-${EPOCH}-' part? I've been trying to use ${MONITOR_EXEC_ARGS} to add some more commands after the somix sequence for mp3 conversion. This should