[asterisk-users] AgentCallbackLogin via dialaplan and device state
With the 1.4 series I see that AgentCallbackLogin is deprecated and that to get the functionality it is suggested that it be implemented via dialplan. I've read through and followed the examples near as I can tell. Agent login/logout works fine with the agents being added to their respective queues and offered calls from the queues to which they've been added. The catch however is with monitoring device state on Local/ channels. Any agent logged in with a Local/ device is always shown as not in use and as a result can be, and often is, offered multiple calls from the queue. Local/ device definitions have been used to allow agents to log in using either SIP devices registered with the server or external numbers. A similar configuration using AgentCallbackLogin works fine with the agent state being correctly reflected in the queue when the agent is on a call. However, with it being a deprecated option, I'd like to avoid using it. In researching this, I found a reference on voip-info.org that indicated the Local/ channels do not support state information in 1.4 without a backport from 1.6. Near as I can tell the Ubuntu package that I'm using has this backport already. Can someone here provide pointers on how not only allow an agent to provide a dynamic call back number, but still limit the number of calls from the queue to agents? If samples of my configuration files would help illustrate, I'd be happy to provide them. -- Jamin W. Collins ___ -- 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] MeetMe w/ SMP was (My Phone Review- Large Scale Corp Deployment.)
Andrew D Kirch wrote: Jamin W. Collins wrote: Steve Edwards wrote: I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I haven't had a crash since. Meetme does not play well with SMP. Interesting, I've been running asterisk (v1.2.10) on an SMP system (dual Xeon 2.66Ghz) with several MeetMe conference rooms for quite a while now. The server has a current uptime of 24 days (moved the system to a new UPS). I have not experienced the crashes you reference. Are either of you running hyperthreading? Yes, the Dual Xeon server reports four CPUs. However, in reading the rest of the thread it appears the original reporter is using dynamic MeetMe rooms. We normally use static rooms. -- Jamin W. Collins ___ --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] MeetMe w/ SMP was (My Phone Review- Large Scale Corp Deployment.)
Steve Edwards wrote: On Sun, 24 Dec 2006, Jamin W. Collins wrote: Steve Edwards wrote: I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application is mostly meetme conferences being created and closed all day long. Peak load is around 200 SIP calls. I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I haven't had a crash since. Meetme does not play well with SMP. Interesting, I've been running asterisk (v1.2.10) on an SMP system (dual Xeon 2.66Ghz) with several MeetMe conference rooms for quite a while now. The server has a current uptime of 24 days (moved the system to a new UPS). I have not experienced the crashes you reference. I now suspect the DL380's. Others have said they were running HT and SMP systems without incident. I stil haven't had a crash since I booted the non-SMP kernel :) Another difference I noted between your reports and those (like myself) indicating success with SMP and MeetMe is that it appears you are frequently using dynamic MeetMe rooms, is this correct? We normally use static rooms. Perhaps the problem lies with the creation and tear down of the rooms? -- Jamin W. Collins ___ --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] MeetMe w/ SMP was (My Phone Review- Large Scale Corp Deployment.)
Steve Edwards wrote: I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application is mostly meetme conferences being created and closed all day long. Peak load is around 200 SIP calls. I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I haven't had a crash since. Meetme does not play well with SMP. Interesting, I've been running asterisk (v1.2.10) on an SMP system (dual Xeon 2.66Ghz) with several MeetMe conference rooms for quite a while now. The server has a current uptime of 24 days (moved the system to a new UPS). I have not experienced the crashes you reference. -- Jamin W. Collins ___ --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] VMware and Digium TDM400P card
Andy Green wrote: I have recently installed and upgraded trixbox in vmware on a win2000 server. So, you have VMWare running on Windows 2000 Server (host) and are trying to run trixbox within a VMWare session (guest), do I follow you correctly? Everything works as expected but I can't seem to get vmware to see the card (it was installed after the vmware/trixbox was set up) I don't believe VMWare can provide the guest physical access to non-standard hardware (the TDM400P in this case). In short the configuration you're attempting won't work, at least not in any way that I know of. -- Jamin W. Collins ___ --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] Tracking the source of a disconnect? - SOLVED
Doug Lytle wrote: Jamin W. Collins wrote: callprogress = yes The only thing I'm iffy about is the above entry. Maybe it's mistaking the progress as disconnect? That does appear to have been the issue. We haven't had a new occurrence of the random disconnects since disabling callprogress. Thank you. -- Jamin W. Collins ___ --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] Tracking the source of a disconnect? - SOLVED
Eric ManxPower Wieling wrote: The comments in /etc/asterisk/zapata.conf didn't tip you off? ; ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call ; progress attempts to determine answer, busy, and ringing on phone lines. ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, ; so don't count on it being very accurate. Based on the comments, I mistakenly thought the setting would be ignored on non-FXS devices. Specifically since the PRI already had the signaling out of band for all of this. I thought knowing for sure that this fixed the issue I reported might be useful to others. So, I reported back that it had in fact corrected it. I apologize if my error has offended your sensibilities in some way. -- Jamin W. Collins ___ --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] Tracking the source of a disconnect?
Jamin W. Collins wrote: periodically, I've been getting reports from users of being disconnected in mid-conversation. I've checked the system's logs for any indication of problems and they all appear clean. Eventually, I enabled both PRI and SIP debugging in an effort to track down the location of these disconnects. At this time it appears that the asterisk is initiating a disconnect of both the PRI and the SIP channel (see the log snippet below). However, there doesn't appear to be any indication of why the asterisk is deciding to terminate the calls. This problem has continued even after an upgrade to Asterisk 1.2.12, Zaptel 1.2.9, and libpri 1.2.3. All of the debugging so far (SIP and intense PRI) have only confirmed that the Asterisk box is the initiator of the disconnect request. In all cases I've reviewed so far, the Asterisk system initiates a disconnect on the PRI channel followed by sending a BYE request to the SIP party. The system has currently been taken out of service due to the frequency of calls being randomly terminated. Any suggestions or ideas on how to isolate the cause/source of the disconnects would be most appreciated. Motherboard: H8SSL-i rev: 1.02[1] Memory: 1 Gig DDR400 (2x 512M) T1 Card: TE110P rev: --B [1]-http://www.supermicro.com/Aplus/motherboard/Opteron/HT1000/H8SSL-i.cfm -- Jamin W. Collins ___ --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] Tracking the source of a disconnect?
Doug Lytle wrote: Jamin W. Collins wrote: Jamin W. Collins wrote: periodically, I've been getting reports from users of being disconnected in mid-conversation. I've checked the system's logs for Lets see your zapata.conf Here you go: [trunkgroups] [channels] context=default rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no resetinterval = never #pridialplan = unknown #pridialplan = callprogress = yes switchtype = national signalling = pri_cpe group = 1 channel = 1-23 -- Jamin W. Collins ___ --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] Tracking the source of a disconnect?
Doug Lytle wrote: Jamin W. Collins wrote: Doug Lytle wrote: callprogress = yes The only thing I'm iffy about is the above entry. Maybe it's mistaking the progress as disconnect? The calls in question are connected for varying time frames. In some cases 5 minutes, some 20 seconds, others 40 minutes before the disconnect occurs. This still a concern? -- Jamin W. Collins ___ --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] Problems Unpacking tarball For Asterisk Application
[EMAIL PROTECTED] wrote: I was successful in getting the tarball for a2billing [EMAIL PROTECTED] a2billing]# ls -all total 4872 drwxr-xr-x 2 root root4096 Sep 11 06:22 . drwxr-xr-x 20 root root4096 Sep 10 21:28 .. -rw-r--r-- 1 root root 165 Sep 11 06:16 download.php?get=Asterisk2Billing_release_Chameleon_beta.tar.gz -rw-r--r-- 1 root root 4960345 Sep 11 06:31 download.php?get=Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz ^ The above is your file name, note the additional download.php?get= on the file name. -- Jamin W. Collins ___ --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] Tracking the source of a disconnect?
, ourstate Null, peerstate Null Sep 8 08:50:55 VERBOSE[14054] logger.c: NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Sep 8 08:51:39 VERBOSE[14047] logger.c: 12 headers, 0 lines -- Jamin W. Collins ___ --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: Tracking the source of a disconnect?
Tony Mountifield wrote: It looks like the PRI connection is going down first, and when that channel exits, it causes the SIP channel to be hung up. So concentrate on the PRI. Yep, that's what I've seen so far. Been trying to concentrate on the PRI, but not seeing any indication of what is triggering the actual disconnect. Try enabling intense PRI debugging pri intense debug span N. You may want to direct the PRI debugging to a file with pri set debug file filename. I'll give the intense debugging a shot. It's not clear from the log you posted whether q931_hangup() was called because of a Q.931 message Asterisk received, or just because it decided to. Hopefully, the intense debug would make that clear. In the log posted there's a 6 second gap prior to asterisk initiating the disconnect. Sep 8 08:50:39 VERBOSE[14047] logger.c: Destroying call '[EMAIL PROTECTED]' Sep 8 08:50:55 VERBOSE[31079] logger.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request Doesn't this indicate that there was no inbound message on the PRI? Since the provided log section also indicates that pri debug (normal, not intense) was enabled? -- Jamin W. Collins ___ --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: Tracking the source of a disconnect?
Tony Mountifield wrote: Try enabling intense PRI debugging pri intense debug span N. You may want to direct the PRI debugging to a file with pri set debug file filename. It's not clear from the log you posted whether q931_hangup() was called because of a Q.931 message Asterisk received, or just because it decided to. Hopefully, the intense debug would make that clear. Alright, it happened again. Here's the log entries from Asterisk sending the disconnect: Sep 8 14:22:41 VERBOSE[14047] logger.c: --- (14 headers 11 lines)Sep 8 14:22:41 VERBOSE[14047] logger.c: --- (14 headers 11 lines)--- Sep 8 14:22:41 VERBOSE[14047] logger.c: Destroying call '[EMAIL PROTECTED]' Sep 8 14:22:43 VERBOSE[32398] logger.c: [ 00 01 ce be 08 02 02 56 45 08 02 81 90 ] Sep 8 14:22:43 VERBOSE[32398] logger.c: Informational frame: Sep 8 14:22:43 VERBOSE[32398] logger.c: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Sep 8 14:22:43 VERBOSE[32398] logger.c: N(S): 103 0: 0 N(R): 095 P: 0 9 bytes of data Sep 8 14:22:43 VERBOSE[32398] logger.c: -- Restarting T203 counter Sep 8 14:22:43 VERBOSE[32398] logger.c: Stopping T_203 timer Sep 8 14:22:43 VERBOSE[32398] logger.c: Starting T_200 timer Sep 8 14:22:43 VERBOSE[32398] logger.c: Protocol Discriminator: Q.931 (8) len=9 Sep 8 14:22:43 VERBOSE[32398] logger.c: Call Ref: len= 2 (reference 598/0x256) (Originator) Sep 8 14:22:43 VERBOSE[32398] logger.c: Message type: DISCONNECT (69) Sep 8 14:22:43 VERBOSE[32398] logger.c: [Sep 8 14:22:43 VERBOSE[32398] logger.c: [08Sep 8 14:22:43 VERBOSE[32398] logger.c: [08 02Sep 8 14:22:43 VERBOSE[32398] logger.c: [08 02 81Sep 8 14:22:43 VERBO SE[32398] logger.c: [08 02 81 90Sep 8 14:22:43 VERBOSE[32398] logger.c: [08 02 81 90] Sep 8 14:22:43 VERBOSE[32398] logger.c: Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Sep 8 14:22:43 VERBOSE[32398] logger.c: Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] Sep 8 14:22:43 VERBOSE[32398] logger.c: -- Hungup 'Zap/5-1' Sep 8 14:22:43 VERBOSE[32398] logger.c: == Spawn extension (outbound, 15551212121, 1) exited non-zero on 'SIP/webacd.net-081e3b40' Sep 8 14:22:43 VERBOSE[32398] logger.c: set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to Sep 8 14:22:43 VERBOSE[32398] logger.c: set_destination: set destination to 1.2.3.4, port 5060 Sep 8 14:22:43 VERBOSE[32398] logger.c: Reliably Transmitting (no NAT) to 1.2.3.4:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 4.3.2.1:5060;branch=z9hG4bK55f07d66;rport From: sip:[EMAIL PROTECTED]:5060;tag=as30c620b0 To: sip:[EMAIL PROTECTED];tag=3465 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 -- Jamin W. Collins ___ --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] PAP2 TUI Configuration Menu
Nabeel Jafferali wrote: I don't belive there is a way to turn it off, but you can prevent the IVR menu being used to factory reset the device using the provisioning tools. Would you happen to have any more specifics on this? Perhaps an example of the reset option being disabled? Can any of the other options, such as IP playback, be disabled? -- Jamin W. Collins ___ --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] PAP2 TUI Configuration Menu
Does anyone know of a way to disable access to the TUI interface (accessed via ) on the PAP2 devices? I'm looking at using these devices for lobby and door phones and would like to remove/disable the TUI interface if at all possible. -- Jamin W. Collins ___ --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] FXS adapters and Polycom phones
Mike wrote: I`m looking for a SIP-PSTN adapter, basically to switch a customer from a cheap PBX to mine, but resuing their own Norstar PSTN phones. They have 10 phones. From a price point of view, it seems that 10 individual GrandStream SIP adapters is the best way to go, but it seems so inelegant to me. What is recommended ? Not sure about the others, but I've had decent experiences with the Linksys PAP2 series, and they aren't that expensive. Second question: I have a GrandStream GXP-2000, that despite what everybody says I love. I am still looking for a replacement, if only because it doesn`t look as good and it does have a few quirks. I was looking at Polycoms, but some questions are unanswered by looking at their datasheet. - Does the Polycom 501 have an integrated router (like the GXP-2000, latest firmware, does) Not entirely sure what you're asking here. If you're wondering if it has a two NIC interfaces (a pass-through for the PC) then yes. - Can you have more than one SIP/account on the phone, each ringing in a way that lets the user know which account is ringing? (GXP2000 does it by making it possible to have each line linked to a separate SIP account) The 501 is capable of having 3 different line appearances, each of which can have a primary and secondary server configured for them. -- Jamin W. Collins ___ --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] an operational scenario
Bruce Ferrell wrote: the problem I'm seeing is one way audio between extensions. I've splpit up the numbering plan internal/external. All are in the same range. I'll try splitting them and see what happens. By one way audio between extensions are you talking about calls between extensions where one side is on the internal network and one side is on the external network? If so you might look at disabling reinvite and/or making sure the external party's RTP connection is able to make it through any firewall you might have in place. -- Jamin W. Collins ___ --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] Initiating a transfer from an analog handset?
Is there a way to initiate a transfer using an analog handset? For instance I'm looking for a way to do something like the following: External call comes in and is answered by user A. After talking to the caller they determine that the caller really needs to speak to user B. Is there any way for user A to initiate a transfer to user B, using only their analog handset? Now to make things possibly more complex, is the above still possible if the analog handset is connected to a Zhone Zplex channel bank? -- Jamin W. Collins ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI STREAM FILE command
On Mar 28, 2005, at 7:30 AM, Bill Kervaski wrote: Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a SAY NUMBER 123 # before and after the STREAM FILE and they both work fine, returning 200 OK, etc. I'm using Debian's 1.0.5 packages with no problems. I can verify that the AGI STREAM FILE command works fine. For example one invocation I'm using is: stream file confirm-reenter '12' -- Jamin W. Collins ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any Zaurus users??
On Mar 19, 2005, at 2:31 PM, Kristian Kielhofner wrote: Kris Edwards wrote: Success with PDA's as phones has never seemed possible. Power is one issue. Most people have reported getting pocket pc sip phones to work (with 802.11), but they have ridiculous short talk times, I want to say that I have never heard of anyone getting more than an hour out of any device/battery combination. With the Zaurus SL-5500 (the only PDA I have) I got stuck on the pinout of the headphone/mic jack and put it on the backburner... I've found that the pin out of the headphone/mic is compatible with that of a cell phone headset the only difference being that you need a jack converter due to the different sizes. However, I had no problem locating a converter at my local Radio Shack (electronic hobby store). -- Jamin W. Collins ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Experience using Gmail for Asterisk MailingList
On Tue, Sep 07, 2004 at 01:44:32PM -0300, Marconi Rivello wrote: Is top posting writing above the quoted text? Is it politically incorrect? Yes, on both counts. Posting below the quoted text and trimming your quoting to only include relevant information (that to which you are responding and perhaps limited context) normally helps to make it easier to follow your response. -- Jamin W. Collins Linux is not The Answer. Yes is the answer. Linux is The Question. - Neo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO line hum w/ Z-plex 10
I've recently begun integrating an Asterisk system into my house. I purchased the Dev Kit a year or two ago when Digium was selling it as a Z-plex 10 channel bank with the T100P. I've recently found that when I connect the serial monitoring port to my system it introduces a noticable hum on my incoming FXO lines. Anyone know (or have suggestions about) how to prevent this? -- Jamin W. Collins To be nobody but yourself when the whole world is trying it's best night and day to make you everybody else is to fight the hardest battle any human being will fight. -- E.E. Cummings ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P FXO, 2 slots?
On Sun, May 02, 2004 at 09:07:37PM +0100, Kevin Walsh wrote: The same Digium shop page suggests that two PCI slots would be required. I'll assume the card is too fat, with the daughter board(s) fitted, to fit into a single slot. This is something I would like to see confirmed, does this card really take 2 pci slots? I had hoped to make use of one of these and a T100P in an SS40G case for personal home use. -- Jamin W. Collins Never underestimate the power of very stupid people in large groups. -- John Kenneth Galbraith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Room Monitor
On Thu, Feb 19, 2004 at 04:06:50PM -0500, James Golovich wrote: To bring this back on topic. Have you considered leaving a phone with the handset off the base, or speakerphone turned on in the room? Set the zap channel to immediate and send it to a special context. Have the s extension send into a meetme that is talker only and then all you have to do is dial into the meetme and monitor the call. No, I hadn't. These features didn't exist when I first looked into Asterisk and started messing with it (about 2 years ago or so). That's one of the reasons I thought I'd ask here. I figured someone might have a viable suggestion. Right now we've just configured to speaker phones and call one from the other and mute the listener. It's working much better than the monitor (no static, YAY!). I'd be interested in any pointer you could provide on the above configuration. We did try the Radio Shack FM solution and it was marginally better than the existing baby monitor. -- Jamin W. Collins Facts do not cease to exist because they are ignored. --Aldous Huxley, Proper Studies, 1927 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Room Monitor
On Thu, Feb 19, 2004 at 10:28:09AM -0500, Walt Reed wrote: Hmm. Is it just me, or does this sound like a sledgehammer for a thumbtack kind of application? Radioshack has cheap intercoms that work fairly well. They have 900Mhz wireless and FM over powerline versions. Most cheap baby monitors are in the 27Mhz band which sucks and is very prone to static. They are also made for $3 in china with NO QA and horrible parts using 1970's technology. Actually the baby monitors tend to be in the 47Mhz band, but yes they still suck. There are newer models in the 900Mhz and 2.4Ghz range. However, reviews of the 900Mhz models are almost unanimous in declaring them to be worse than the 47Mhz models. While my experience with 2.4Ghz phones indicates that they will trash most 802.11b networks. So, I'm trying to move away from the wireless solution. Mainly, due to the interference between the two locations. The three station FM solution looks promising. That is if it can deal with the stations being on different breakers within the same residence. You may also want to look at a better model of intercom. http://www.fisher-price.com/us/babygear/product.asp?id=17605c=bgm Uses 900Mhz. Several reviews of this model indicate severe static problems. -- Jamin W. Collins Linux is not The Answer. Yes is the answer. Linux is The Question. - Neo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Room Monitor
On Tue, Feb 17, 2004 at 10:04:02PM -0800, David Liu wrote: Well use a Polycom IP 500 and put to auto answer and ringer off. Then you can use it as a room monitor device. Seems like that could do the trick. However, I was hoping for a sub $200 solution. Anyone know of a less expensive solution? -- Jamin W. Collins Remember, root always has a loaded gun. Don't run around with it unless you absolutely need it. -- Vineet Kumar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Room Monitor
On Wed, Feb 18, 2004 at 01:58:19PM -0600, Jonathan Moore wrote: OK, I think I was wrong on the Grandstream. I loaded up the web config and I am not seeing the option. I would suggest contacting them directly to see. If not that I am pretty sure the snom100 has an auto answer mode. It's listed as a feature of the 105e model but that seems to be back up in the ~$200 range. -- Jamin W. Collins To be nobody but yourself when the whole world is trying it's best night and day to make you everybody else is to fight the hardest battle any human being will fight. -- E.E. Cummings ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Room Monitor
On Tue, Feb 17, 2004 at 11:36:20PM -0600, Jonathan Moore wrote: What about a phone, analog or IP, put in an auto answer mode? Do you know of any off hand that support this? Perhaps one with the ability to turn off the ringer? -- Jamin W. Collins To be nobody but yourself when the whole world is trying it's best night and day to make you everybody else is to fight the hardest battle any human being will fight. -- E.E. Cummings ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Room Monitor
Do any of you know of a cost effect device that could be connected to an Asterisk station port to provide room monitoring? I'm looking to replace the wireless baby monitor we currently have, since there is too much interference between our daughter's room and our room for it to work effectively. I've found a few items[1-4] that seem to provide the feature I'm looking for, but they seem much more expensive than necessary. Essentially, I'm looking for something that I can assign an extension on asterisk to and then call from another station to activate monitoring. Any ideas are welcome. [1] - http://www.spyandsecuritystore.com/informer.html [2] - http://shop.store.yahoo.com/spytechagency/11435.html [3] - http://www.talkingelectronics.com/security/room_devices.html [4] - http://www.surveillance-spy-cameras.com/room-monitor.htm -- Jamin W. Collins To be nobody but yourself when the whole world is trying it's best night and day to make you everybody else is to fight the hardest battle any human being will fight. -- E.E. Cummings ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
