similar to: Touch monitor file name format

Hi list, The default automon (touch monitor) output file name format is: auto-epoch-caller-callee.wav A variable is available to modify the second half: auto-epoch-${TOUCH_MONITOR}.wav But, I can't modify the first half, 'auto-epoch-', with any variables that I know of, including ${MONITOR_FILENAME}. I want to immediately convert this output file to mp3, e.g. with

Hi list, Can someone please explain how to get one touch recording (automon) to work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My current configuration includes the following settings: In /etc/asterisk/sip.conf: [2000] ; Siemens Gigaset S675 IP wireless SIP phone. type=friend secret=1234 context=phones-j dtmfmode=rfc2833 qualify=yes

Hi all ! I'm trying to make a automonitor generated filename to "make its way" into CRD(usrefiled), so I can keep track of recorded conversations in CDR logs. Looking how to do that, I have found cool (but almost undocumented) option of res_monitor: if you set monitor format in form of "format:<string>" (i.e. "wav:monitor"), res_monitor will prefix the

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, 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 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 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 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 there, We used to record all the calls with the Monitor function. Now, I haveimplemented on-demand recording with automon instead... Everything is working fine apart from the generated filename, which as per all docs, should be auto-epoch-caller-callee....but in my case, it is auto-epoch-who_started_record-the_other_end. Thank you all in advance. Regards, Marta -------------- next part

Hello All, Our old Lucent Argent system had a feature whereby when you initiate recording during a call, it would afterwards send the recording as a voicemail message to the user who initiated the recording. We use the automon *1 recording function in asterisk, which allows users to record a call if necessary on the fly. Unfortunately there doesn't appear to be an easy way for the user to

Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints?

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, 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

On Fri, Mar 2, 2012 at 11:32 AM, Evandro Menezes <emenezes at codeaurora.org> wrote: ... > Figure 3.3 on page 16 of www.x86-64.org/documentation/abi.pdf is not > normative. See foot note 7 in the same page. Figure 3.4 on page 21 > confirms that the use of a frame-pointer is optional. > > So, if one doesn't use ENTER in the prologue and uses RSP to access local >

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 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, 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