[asterisk-users] Introducing ToRELP.. A quick and dirty way to push notifications away from Asterisk to a Python Tornado process.
Heya everybody.
I work on a lot of AGI/AMI/AJAM/etc.. projects and recently discovered
RELP (available via rsyslog) which is defined here:
http://www.librelp.com/relp.html
I've been pimping out (yes.. pimping) the Log dialplan application to
quickly emit a message to my local syslog which is then delivered to
rsyslog. I have rsyslog configured to use RELP between itself and a
remote RELP server (ToRELP) in a reliable way.
Messages are queued and redelivered if for any reason the remote RELP
server dies.
After doing some testing I've found that setting the core debug and
verbosity to 0 does not negatively impact messages going to syslog..
which is wonderful.
I personally use this to send JSON commands off to a worker process
that helps with some screenpop processes.
https://github.com/whardier/torelp
My dialplan has a line like the following in it:
Log(NOTICE, JSON:{'i': 'havealovelybunchofcoconuts', 'dee': 'dledee'})
Have fun.. always looking for contributions to my projects. Hope
somebody out there can utilize this.
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Re: [asterisk-users] Remote hold on PRI
So, watching the asterisk console with full debug on shows something about Starting Music On Hold for Channel xx/yy-zz? Shane On Jan 7, 2008 11:00 AM, Gaëtan Minet [EMAIL PROTECTED] wrote: Hi Nobody has an Idea ? Should I try and fill a bug report (or feature request ?) at Digium ? The only solution I personally see is a patch in the source. Regards Gaetan On 04/01/2008, at 23:26, Gaëtan Minet wrote: Hi everybody We have a strange problem with several asterisk servers (Version 1.4.11) using PRI cards (tied to telco here in Belgium). Indeed we noticed that whenever a local user places an outgoing call through the PRI (and telco) to another IPBX (tied to telco using BRI or PRI), if the remote party places the call on hold, the caller hears the _local_ music on hold instead of the remote one. In fact we can briefly hear the remote music on hold start, then it is replaced by the local one. More precisely: Company 1 uses an asterisk server with a PRI card tied to the telco. Company 2 uses any PBX that ca place calls on hold and is tied to the telco using a digital interface (tested with BRIs and PRIs) A (company 1) calls B (company 2) B answers and park or places the call on hold A hears the MOH of company 1. The same happens when calling a mobile: when the mobile user puts the call on hold, instead of hearing the mobile operator's own moh, the calling user hears the moh of his own company asterisk. I think this has something to do with REMOTE_HOLD notifications on PRI lines that gets reported back to the calling asterisk server, which in turn somehow puts the bridged (SIP) channel on hold, but I can't find much more information about this. Is this the expected behavior ? A feature or a bug ? Do you know if this can be tuned/tweaked/disabled (i.e. filter or ignore this signaling on the zap channel(s) ?) Kind regards Thanks NB: Oddly enough, when the local user hears the music on hold, his own channel (a local SIP phone in this case) isn't reported as On Hold when issuing sip show channels in cli, and no AMI Hold/Unhold events are generated. I double checked, the MOH that gets played is the one specified in sip.conf, NOT zapata.conf. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can Local channels inhibit an Answer() until it is satisfied with the endpoint?
I'm trying to get dynamic agents/queues working for any type of telephone with a DID. I need an application or a method to inhibit a channel/technology from responding with an Answer() until the queue member accepts the call by hitting '#'. This way I can use any POTS line as a queue member, saving costs on ATA's for home workers who only spend a minority of their time at home. They can just dial in, an app will verify their caller ID and say You joined the queue. Then calls will ring in on their telephone line. They will pick up and get briefed on what queue they would be answering (or look at the Caller ID) and press a verification key to accept the call and send an Answer status back to the queue app for that channel. I've thought of several other ways of doing this, including remote pickup and using the dial-out system to initiate the call for that. I personally feel it would be frustrating if the call never terminated when somebody else answered the queue. Shane Spencer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Long call setup times on SIP to zaptel calls
do your sip phones dial after a timeout? If the timeout is set to around 5 seconds you may have a dialplan issue on your sip hardware. Shane On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote: I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Long call setup times on SIP to zaptel calls
I only say this because nobody in our office knew how to use the checkmark on snom phones to initiate a call, they always just waited for the phone to initiate the call for them :) On 2/15/07, Shane Spencer [EMAIL PROTECTED] wrote: do your sip phones dial after a timeout? If the timeout is set to around 5 seconds you may have a dialplan issue on your sip hardware. Shane On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote: I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Auto Answer (Paging)
I hate to say this, but voip-info.org has a few different methods of handling this already defined. If you are 'intercomming' to several styles of SIP based phones, you have but to only configure the phone to accept those types of calls and add a SIP header pre Dial(). Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Red alarms
I had this happen to me because I was not configured properly and for some reason the telco was automatically dropping me every so often. I called the telco and they corrected me. Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Red alarms
PRI's are sold with SLA's so get them to diagnose your problems using their own testing equipment. Ideally this would work, however I typically have to bring a case of beer to my local telco and find my support team - if that fails I just set it somewhere that makes it easy for people to trip over. Its fun having a prankster relationship with the telco. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Howto use PRI lines (E1 or T1) for data calls?
point to point E1 lines? Or are you interfacing to a PSTN network for local calling/receiving? PTP E1 http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration On 2/5/07, Roger Schreiter [EMAIL PROTECTED] wrote: Hi, I'm looking for a mean to send digital data over an E1 line, just like isdn4linux or Capi via AVM's FritzCard is able to do it with BRI lines (e.g. for PPP or ISDN raw connections). I'm not looking for modulated audio data representing digital data, like fax or the analogue modems of former times. I want an interface to the ISDN raw data, with an outgoing call marked as data, not voice. Best would by the behaviour of /dev/ttyIx. Thus I thought about iaxmodem. Could please anyone tell me, whether it is possible to make an ISDN data call using iaxmodem asterisk ISDN PRI line ? Thanks for any hint! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] volume control in VoIP
I hate to think of the possible echo if you change the volume and a sip device didn't know about it. It wouldn't effectively use the echo cancellation on board, I am not an expert with echo cancellation however. It should be possible to just multiply each byte of a waveform by a percentage if the codec is slinear/ulaw/alaw or similar almost as quickly as it passes from one network stream to the next. I would snag the source for like xmms if it can do software volume control and peek into it.. could be valuable. Let us know how well the echo cancellation works in speakerphone eh! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Talk without gmail accout?
You need to at least register AFAIK. Download gaim and use its facilities to rejister. Jabber is not for the faint of heart when it comes to IM platforms, read up on it if you haven't already. On 2/3/07, Ian Hailey [EMAIL PROTECTED] wrote: Hello all, I am having trouble getting gtalk to work with my account which is not using a gmail.com email address. When I do this there an error from the Jabber module: [Feb 3 20:51:17] ERROR[6286]: res_jabber.c:573 aji_act_hook: JABBER: Node Error [Feb 3 20:51:17] WARNING[6286]: res_jabber.c:1495 aji_recv_loop: JABBER: Got hook event. JABBER: gtalk_account INCOMING: stream:errorhost-unknown xmlns=urn:ietf:params:xml:ns:xmpp-streams/str:text xmlns:str=urn:ietf:params:xml:ns:xmpp-streamsSet the 'to' attribute of stream element to the domain part of the user's JID. Example: to='gmail.com'./str:text/stream:error/stream:stream So does this only work if you have email accounts from gmail.com? Thanks Ian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamically Adding A Context
Seriously? You want serious! You can't handle the serious! I would assume that editing a file and refreshing a system by means of a program or self intervention which causes no interruption in service could be concidered dynamic. How does asterisk realtime handle this thats so radically different that it can be the only true dynamic method of doing this. BTW.. Did you figure it out yet? On 1/31/07, j [EMAIL PROTECTED] wrote: Seriously man. I don't want to be testy here, but what part of *dynamic* didn't you understand? Adding a context to a flat file and reloading the server is NOT dynamic. And, as I explained in a previous post, realtime is not a solution I can use for this issue because I'm updating proxy software that uses the AMI so realtime is not an option. For everyone else; Thanks for trying to take a stab at this. It seems there simply is no way to do it. Perhaps I'll submit a patch to digium so at least we have this simple functionality in the future... j On Tue, 2007-01-30 at 13:17 -0900, Shane Spencer wrote: Reload.. Reload.. Reload.. On 1/30/07, Shane Spencer [EMAIL PROTECTED] wrote: Realtime.. Realtime.. Realtime.. On 1/30/07, j [EMAIL PROTECTED] wrote: On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote: On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote: In order to do this, I have to add a couple quick extensions to the dial plan dynamically, so I have to be able to add my own context. from API use Command to run the CLI command add extension But you can only add to an existing context with that. Yes exactly. I tried the 'add extension' command. With *and* without the 'replace' argument, if the context does not already exist the command gives an error ;( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamically Adding A Context
Cool. My first attempt would have been to find out how to use asterisk variables in the dialplan since I can set those like crazy mad via an AGI. Then i would have cried and become horribly demotivated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [asterisk-dev] Dynamically Adding A Context
Reload.. Reload.. Reload..! /me ducks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [asterisk-dev] Dynamically Adding A Context
Hahaha,, I think thats a freaking SWEET suggestion :) On 1/31/07, Andrew Furey [EMAIL PROTECTED] wrote: On 01/02/07, Yuan LIU [EMAIL PROTECTED] wrote: What Lee suggested is to have the AGI script to actually parse, insert a new context in extensions.conf, or deleting from it, then reload extensions.conf. This would at least achieve what you wanted to do. Or alternatively, to avoid complete disaster, why not have extensions.conf include another file (#include somefile.conf) and edit that one with your script? I've done that before (although I was actually recreating the entire file each time by populating from an external database). Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heartbeat on Digium T1 PCI cards?
http://www.junghanns.net/en/ISDNguard_produkt.html I am no longer with the company that got frelled by ATT and/or asterisk. However this unit would have definitely helped out. Just disconnect when heartbeat not found. On 1/29/07, Shane Spencer [EMAIL PROTECTED] wrote: It was either down or asterisk was frozen. Either way a heartbeat could fix that. On 1/29/07, C F [EMAIL PROTECTED] wrote: Shane, are you trying to say that the PRI was actualy down (the D channel was NOT up) for the time that ATT is billing you? On 1/29/07, Shane Spencer [EMAIL PROTECTED] wrote: Tell that to ATT who socked us with multiple $20k bills. We cant figure out where the error was. Or why a call was established for over 50 hours between two states with completely different PBX hardware. On 1/29/07, C F [EMAIL PROTECTED] wrote: If Asterisk Is Down Then The D Channel Is Down Hence No Calls Can Remain Active On 1/29/07, Edoardo Serra [EMAIL PROTECTED] wrote: Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamically Adding A Context
Realtime.. Realtime.. Realtime.. On 1/30/07, j [EMAIL PROTECTED] wrote: On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote: On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote: In order to do this, I have to add a couple quick extensions to the dial plan dynamically, so I have to be able to add my own context. from API use Command to run the CLI command add extension But you can only add to an existing context with that. Yes exactly. I tried the 'add extension' command. With *and* without the 'replace' argument, if the context does not already exist the command gives an error ;( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamically Adding A Context
Reload.. Reload.. Reload.. On 1/30/07, Shane Spencer [EMAIL PROTECTED] wrote: Realtime.. Realtime.. Realtime.. On 1/30/07, j [EMAIL PROTECTED] wrote: On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote: On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote: In order to do this, I have to add a couple quick extensions to the dial plan dynamically, so I have to be able to add my own context. from API use Command to run the CLI command add extension But you can only add to an existing context with that. Yes exactly. I tried the 'add extension' command. With *and* without the 'replace' argument, if the context does not already exist the command gives an error ;( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heartbeat on Digium T1 PCI cards?
Tell that to ATT who socked us with multiple $20k bills. We cant figure out where the error was. Or why a call was established for over 50 hours between two states with completely different PBX hardware. On 1/29/07, C F [EMAIL PROTECTED] wrote: If Asterisk Is Down Then The D Channel Is Down Hence No Calls Can Remain Active On 1/29/07, Edoardo Serra [EMAIL PROTECTED] wrote: Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 Wire Level Tapping
Wow, thanks for the awesome reply :) On 1/28/07, Leo Ann Boon [EMAIL PROTECTED] wrote: Shane Spencer wrote: I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 bridging using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium hardware on me until the project begins. There are a number of systems using ISDN digital taps. The proper way requires a high impedance bridge - you don't want to load the line that you're tapping. Anybody found a method of spying on a D-Channel and all voice channels using standard T1 equipment? I am making a rough assumption that if I can trick the zaptel drivers into operating without anything responding to a TX signal then I can do the following: You can directly bridge the 2 ports and extract what you need as you bridge - see pridump.c in libpri. You don't even need asterisk, just the zaptel and libpri. The only problem with this approach, is that the bridge becomes a point of failure. Your box down, your PRI goes down as well. S-T1 = T1 to Spy On T1-1 = Digium T1 card #1 T1-2 = Digium T1 card #2 Map S-T1(RX) to T1-1(RX) and S-T1(TX) to T1-2(RX) and decode the D-Channel where appropriate, should I be able to spy on the RX/TX channels enough to make a recording including CID information? This would help in situations where the monitoring system needs to be replaced or taken down without bothering in-progress calls. This is technically correct, but I don't know how well it works. Eicon recommends a similar technique to do monitoring with their Eicon Server cards. For the BRI, it's done this way. But for the PRI card, they actually suggest using a custom cable. Eicon cards have a special Hi-Z monitoring mode to support this application. http://www.eicon.com/worldwide/solutions/How_To_Call_Tapping_and_Monitoring_with_Diva_Server FYI, Voicetronix has a Hi-Z version of their OpenPRI card that work with an open-sourced voice logging application available from their site. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 Wire Level Tapping
I am very interested in the DACs capabilities of Digium cards, there is no information anywhere on this. I could always do pri bridging via libpri like you suggest however. But having hardware handle the bridging onboard a single PCI card would help reduce my server requirements for a final product, as long as I can spy on active channels somehow. I don't think its going to work that way, I wil test out libpri for a bit. Shane+ On 1/28/07, Leo Ann Boon [EMAIL PROTECTED] wrote: Shane Spencer wrote: I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 bridging using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium hardware on me until the project begins. There are a number of systems using ISDN digital taps. The proper way requires a high impedance bridge - you don't want to load the line that you're tapping. Anybody found a method of spying on a D-Channel and all voice channels using standard T1 equipment? I am making a rough assumption that if I can trick the zaptel drivers into operating without anything responding to a TX signal then I can do the following: You can directly bridge the 2 ports and extract what you need as you bridge - see pridump.c in libpri. You don't even need asterisk, just the zaptel and libpri. The only problem with this approach, is that the bridge becomes a point of failure. Your box down, your PRI goes down as well. S-T1 = T1 to Spy On T1-1 = Digium T1 card #1 T1-2 = Digium T1 card #2 Map S-T1(RX) to T1-1(RX) and S-T1(TX) to T1-2(RX) and decode the D-Channel where appropriate, should I be able to spy on the RX/TX channels enough to make a recording including CID information? This would help in situations where the monitoring system needs to be replaced or taken down without bothering in-progress calls. This is technically correct, but I don't know how well it works. Eicon recommends a similar technique to do monitoring with their Eicon Server cards. For the BRI, it's done this way. But for the PRI card, they actually suggest using a custom cable. Eicon cards have a special Hi-Z monitoring mode to support this application. http://www.eicon.com/worldwide/solutions/How_To_Call_Tapping_and_Monitoring_with_Diva_Server FYI, Voicetronix has a Hi-Z version of their OpenPRI card that work with an open-sourced voice logging application available from their site. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels staying offhook - restart required
Try setting AbsoluteTimeout() as the first parameter in your dialplan entry. Check it out on voip-info.org On 1/28/07, kjcsb [EMAIL PROTECTED] wrote: Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? I tried the following: unload chan_zap.so load chan_zap.so That seemed to reset the offhook status without a reboot. How do I access this in a dialplan or via the Manager? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 Wire Level Tapping
I wanted to know if there was a peekaboo factor to it all. You can flow data under a glass window :) On 1/29/07, Leo Ann Boon [EMAIL PROTECTED] wrote: Shane Spencer wrote: I am very interested in the DACs capabilities of Digium cards, there is no information anywhere on this. I could always do pri bridging via libpri like you suggest however. But having hardware handle the bridging onboard a single PCI card would help reduce my server requirements for a final product, as long as I can spy on active channels somehow. I don't think its going to work that way, I wil test out libpri for a bit. Pardon if I'm wrong, I don't think the DACS mode is really applicable if you're trying to monitor the channels. As I understand it, if you use DACs - the data will just flow between the 2 ports and not to the PCI bus. So logically, you won't be able to spy on the channels. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heartbeat on Digium T1 PCI cards?
It was either down or asterisk was frozen. Either way a heartbeat could fix that. On 1/29/07, C F [EMAIL PROTECTED] wrote: Shane, are you trying to say that the PRI was actualy down (the D channel was NOT up) for the time that ATT is billing you? On 1/29/07, Shane Spencer [EMAIL PROTECTED] wrote: Tell that to ATT who socked us with multiple $20k bills. We cant figure out where the error was. Or why a call was established for over 50 hours between two states with completely different PBX hardware. On 1/29/07, C F [EMAIL PROTECTED] wrote: If Asterisk Is Down Then The D Channel Is Down Hence No Calls Can Remain Active On 1/29/07, Edoardo Serra [EMAIL PROTECTED] wrote: Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 Wire Level Tapping
I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 bridging using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium hardware on me until the project begins. Anybody found a method of spying on a D-Channel and all voice channels using standard T1 equipment? I am making a rough assumption that if I can trick the zaptel drivers into operating without anything responding to a TX signal then I can do the following: S-T1 = T1 to Spy On T1-1 = Digium T1 card #1 T1-2 = Digium T1 card #2 Map S-T1(RX) to T1-1(RX) and S-T1(TX) to T1-2(RX) and decode the D-Channel where appropriate, should I be able to spy on the RX/TX channels enough to make a recording including CID information? This would help in situations where the monitoring system needs to be replaced or taken down without bothering in-progress calls. Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Heartbeat on Digium T1 PCI cards?
I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels staying offhook - restart required
Just for giggles can you set an absolute timeout in the dialplan for all calls in and out of that span? On 1/25/07, kjcsb [EMAIL PROTECTED] wrote: I have a situation where the two Zap channels on a TDM400 are staying offhook after a random period of time; it is not (I believe) related to the FXO side not hanging up. Actually I suspect the server is overheating but I need to do more analysis. Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? Also suggestions on debugging this would be appreciated. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dacs support on Digium T1 equipment.
Heya everybody. I have been peering into the code for zaptel for a while now, I am keenly interested in the dacs support, being able to apparently redirect certain spans to other spans. Not sure if this has to be on the same T1 interface or can be used between T1 interfaces on the same board or possible two different cards. Any information on the functionality of this would be greatly appreciated! I don' t have the equipment any more to test it out. I have been wanting to bridge T1 devices together outside of the dial plan for a long time. However this time I need to be able to monitor the audio data and call information as well. I am fine programming something that can talk to the zaptel drivers, but I need to know if channels placed into a dacs configuration can be monitored at all. If I do what needs to be done with just using a simple dialplan I have echo concerns, I am wondering how much of a concern echo will be between two spans placed into a dacs configuration. I knew if there is echo I can do nothing about it, asterisk needs to be in the middle to help adjust that. If there is a low chance of echo it would save me quite a bit of money by not requiring echo cancellation capable T1 boards. Shane R. Spencer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
