Re: [asterisk-users] OT: USB T1/E1 Interface?
[EMAIL PROTECTED] wrote: Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Imagestream's low cost (about US$500) Envoy T1/E1 router actually uses a USB T1/E1 WAN 'Card'. I wonder how difficult is it to repurpose that card for voice :). 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
Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording
Gavin Henry wrote: Dear All, Is it possible to install * in front of a Avaya IP 406 system via a T connector E1 tap so it's external to the Avaya system? Voicetronix has an open sourced solution using their OpenPRI in Hi-Z mode. http://www.voicetronix.com/open-source.htm#logger 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
Re: [asterisk-users] Transfer via CTI
Phil Menico wrote: I used autodial to allow a user to make a call by clicking on a web directory and placing a call file into the Asterisk outgoing directory. That works perfectly for me. What if I want to click on the web directory and transfer my existing call? Is there a comparable interface? Use the manager interface. 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
Re: [asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M -followup with log
Just curios, does the CS1000 now support RFC2833? Previously, I know the NRS can only support SIP-INFO. Leo Jerry Geis wrote: Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=X disallow=all allow=ulaw allow=alaw context=nortel host=XXX canreinvite=yes qualify=yes usereqphone=yes - Use 'exit' when done Asterisk 1.4.2, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found [0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [0mConnected to Asterisk 1.4.2 currently running on hfemsrv (pid = 18420) hfemsrv*CLI Verbosity is at least 5 [Khfemsrv*CLI sip debug hfemsrv*CLI SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. [Khfemsrv*CLI Reliably Transmitting (no NAT) to 192.168.45.129:5060: OPTIONS sip:192.168.45.129 SIP/2.0 Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK2508d83c;rport From: asterisk sip:[EMAIL PROTECTED];tag=as2cc96e52 To: sip:192.168.45.129 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 19 Apr 2007 19:25:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- ? [Khfemsrv*CLI --- SIP read from 192.168.45.129:5060 --- SIP/2.0 200 OK From: asterisksip:[EMAIL PROTECTED];tag=as2cc96e52 To: sip:192.168.45.129;tag=812da8c0-13c4-46277c06-279cd106-42ff Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK2508d83c Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 Content-Length: 0 - ?--- (10 headers 0 lines) --- ? [Khfemsrv*CLI Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS ? [Khfemsrv*CLI-- Attempting call on SIP/QuadNortel/7113 for [EMAIL PROTECTED]:1 (Retry 1) ? [Khfemsrv*CLI Audio is at 161.49.142.250 port 1 ? [Khfemsrv*CLI Adding codec 0x4 (ulaw) to SDP ?Adding codec 0x8 (alaw) to SDP ? [Khfemsrv*CLI Reliably Transmitting (no NAT) to 192.168.45.129:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK11268a7d;rport From: Admin System 34 sip:[EMAIL PROTECTED];tag=as4e5a553d To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 19 Apr 2007 19:25:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 212 v=0 o=root 18420 18420 IN IP4 161.49.142.250 s=session c=IN IP4 161.49.142.250 t=0 0 m=audio 1 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- ? [Khfemsrv*CLI --- SIP read from 192.168.45.129:5060 --- SIP/2.0 100 Trying From: Admin System 34sip:[EMAIL PROTECTED];tag=as4e5a553d To: sip:[EMAIL PROTECTED];tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 Contact: sip:[EMAIL PROTECTED] Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 - ?--- (11 headers 0 lines) --- ? [Khfemsrv*CLI --- SIP read from 192.168.45.129:5060 --- SIP/2.0 180 Ringing From: Admin System 34sip:[EMAIL PROTECTED];tag=as4e5a553d To: sip:[EMAIL PROTECTED];tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 Contact: sip:7113;[EMAIL PROTECTED]:5060;maddr=192.168.45.129;transport=udp;user=phone Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 - ?--- (11 headers 0
Re: [asterisk-users] Is Allison going to be banned from foreign travel over polar bears?
Steve Prior wrote: I read this story and thought of Allison's prompt to try not to think about blue eyed polar bears. Will she be banned from foreign travel now? I supposed it's ok since blue-eyed polar bears are fictitious and thus protected by the first amendment :) 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
Re: [asterisk-users] moving WiFi phone
I, too, have heard about that best practice of using different channels for different AP's on the same SSID. As far as I can tell, This is standard textbook stuff. Read Cisco press's 'Deploying License Free Wireless Wide-Area Networks' by Jack Unger. it's BS. I don't know who started it, but it has never worked in any of the situations I've encountered. In fact, I know of at least one AP manufacturer (Apple) that has a utility to auto-configure WDS networks, and it auto-configures to use the same channel. That's Using the same channel is bad, because the APs will interfere with each other and your throughput will be reduced. Imagine if you have a total of 2 APs with 10 clients each, the bandwidth will have to be shared amongst the 22 devices. So, if you're able to get 54Mbps on that channel, the net result is everybody gets 54/22 = 2.45Mbps each. Not a very pretty sight. Roaming with multiple APs on the same channel is OK for small set ups. 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
Re: [asterisk-users] Mini-ITX board + FXO PCI card?
Karsten Wemheuer wrote: Hello, Am Donnerstag, den 15.02.2007, 10:55 +0800 schrieb Leo Ann Boon: 1. The smallest mini-ITX case I found that accepts a PCI card is the Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know if it fits? I didn't find its width, and apparently, the C138 will not accept a PCI card bigger than 17,52cm. The C137 can fit 2 TDM400P with the right riser. If You are using the riser card, there will be shared interrupts. The two slots of the riser card are using the same IRQ AFAIK. You can get risers that don't share interrupts, need to ask the vendor. Yes, I did remember the first batch we got had shared interrupts. 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
Re: [asterisk-users] moving WiFi phone
Bruce Reeves wrote: In my experience having ap's with the same SSID and 3 channels of separation overlapping worked if the phone could roam. Recommended is 5 channels of separation. Ronald, Just be aware that even if the phone supports AP roaming, there's no guarantee that the call will continue smoothly from AP to AP. In some cases, it might take a few seconds to handover. 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
Re: [asterisk-users] Mini-ITX board + FXO PCI card?
1. The smallest mini-ITX case I found that accepts a PCI card is the Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know if it fits? I didn't find its width, and apparently, the C138 will not accept a PCI card bigger than 17,52cm. The C137 can fit 2 TDM400P with the right riser. 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
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Matt wrote: Eric, I understand what you are saying about APIC... and from my understanding the O/S takes over control of the IRQs.. but aren't there still only 15 physical IRQs that you can set in the BIOS for devices? I've never seen a machine in which I could go above 15 for a device in the BIOS. Matt, Have you read this? http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html 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
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Matt wrote: Leo, Yes I did read this. And I have ACPI turned on. Unfortunately lspci -vb still is showing devices sharing IRQs. You mean IO-APIC? ACPI is a different beast altogether. lspci -vb and lspci -v should show different results on a proper IO-APIC system. lspci -vb shows what the card thinks it's using. On a XT-PIC system, what it thinks and what it's assigned should be the same. On an IO-APIC system, the interrupts are routed through secondary APICs that can assign additional hardware (local) interrupts (15) to each card. But, for real-mode compatibility sake, the motherboard is required to route the new interrupts to IRQs 1-15. Hence the different value in lspci -vb. Is the 2950 using a riser? You might want to check if there are jumpers on the riser. Some brain dead risers actually share the IRQ lines unless you change the jumpers. Frankly, I really hope that Digium will change the PCI controller on their cards. I think it's buggy that's why it doesn't work properly with some IO-APIC system. If Sangoma can do it, I don't see why not. 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
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Matt wrote: Leo, I am sorry. Yes I mean IO-APIC. So basically the output of lspci -v are the same as cat /proc/interrupts. It is a riser, I will check on that. So here's my questions then. If APIC routes the IRQs to 1-15 for real world usecan you safely have two devices on, say, 14? APIC will assign one to maybe 23 and one to 20. But are they really both on 15 with a potential for conflict? The conflict only happens if your OS is not APIC aware or buggy hardware. In fact 15, is usually used for the secondary IDE port. The reason APIC exists is to support SMP and the plethora of new devices that are present on any modern motherboard. On my nforce motherboard with IO-APIC, lscpi -vb will show lots of devices using IRQ 15. But, I've never seen IRQ misses on any one of them. The same goes for our production systems running Pentium D or Xeon 51x0. 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
Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
Matt wrote: I guess the question is... is it even possible to have a real-time VoIP card running on PCIe? Or with 1,000 Interrupts a second.. does it simply need to have its own IRQ? Have you tried the Sangoma PCIe cards? APIC is supposed to fixed the PCI IRQ problem. AFAIK, APIC is not a virtual interrupt. It requires an additional interrupt controller to deal with the additional interrupt lines. The BIOS cannot see it because it's still stuck with the 8086 15-interrupt mindset. When you run a modern OS like Windows XP and Linux, the OS can will make the CPU aware of the additional interrupts from the secondary interrupt controllers. At the BIOS level, you'll see 'shared' interrupts for APIC system because the mobo designer need to cascade the new interrupt controller to the standard controller. Otherwise, the interrupts from the secondary controller will not be available to real-mode applications. I believe the Digium cards (and some other cards as well) are picky about interrupts because of a faulty PCI controller. That said, the problem is usually more apparent in systems with PCI risers and entry level chipsets. In other words, you get what you pay for. The other alternative is to use industrial PCs with a PCI backplane bus. So far, I've never encountered any interrupt issues with IPCs. 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
Re: [asterisk-users] Skutch AS-66 and an X100P
I don't know anything about a line simulator but your description certainly points to a problem with the simulator. As I'm also doing tests on X100P, I'm interested to know what does a simulator give you that your PBX doesn't. (I wish I had a PBX to play with.) How about just using a working Asterisk PBX :)? Or use a good fxs gateway that allows you to configure custom tones. 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
Re: [asterisk-users] TDM400 with 1 FXO
Klaverstyn, David C wrote: Hi All, I cannot get my TDM to work correctly. In my /etc/zaptel.conf file I have loadzone = us defaultzone=us fxoks=1 Shouldn't this be fxsks if you're using an FXO module as analog trunk? 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
Re: [asterisk-users] Asterisk outbound calling does not wait for answer before playback
Alyed Tzompa wrote: Had the same issue time ago, but Eric shed good light on it, have a look at: http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html Summary: sorry, no nice work around. At least, not in the analog TDM world. Personally, I'll advise everyone to use ISDN if you need to detect call progress for TDM circuits. Or if you're in North America, try the callprogress=yes option in zapata.conf. Analog lines are perfectly fine is you're happy with call progress detection in Wetware(TM) :). For automated call progress detection, you would be happier with a digital line or VOIP. 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
Re: [asterisk-users] TDM400 with 1 FXO
Klaverstyn, David C wrote: Hi, Yes it should, I have changed it back and is still causing the same problems. Did you also missed out the following line in zapata.conf? signalling=fxs_ks 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
Re: [asterisk-users] TDM400 with 1 FXO
Klaverstyn, David C wrote: Yes, I have also since put that in and I get the error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring signalling And if I put in rxwink I get this error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring rxwink It's all very strange. please post your complete zapata.conf - I think there's a preceding line that's confusing the parser. 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
Re: [asterisk-users] TDM400 with 1 FXO
Klaverstyn, David C wrote: My original post does have the contents of the file exactly. In my /etc/asterisk/zapata.conf file I have [trunkgroups] [channels] context=from-pstn 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 group=1 callgroup=1 pickupgroup=1 immediate=no You need to insert signalling before the channel statement: signalling=fxs_ks channel = 1 Yes, I have also since put that in and I get the error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring signalling This warning happens when you change the signalling in zapata.conf without restarting asterisk. I suspect you did a zap reload on the console. I'd suggest you restart both zaptel and asterisk to keep everything in sync. 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
Re: [asterisk-users] Skutch AS-66 and an X100P
Yuan LIU wrote: Kind of do. There are times when it feels like trying to fit two spinning wheels, though:-) 'Zee trick to fit two spinning wheels is to stop the wheels :)'. That why, your first working system is the most important. It's easier to built on once you have a solid foundation. Everyone has to go through this rite of passage. 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
Re: [asterisk-users] Interact with IVR
Yuan LIU wrote: I remember a thread similar to this a while ago but couldn't find. How do I make Asterisk to interact with an IVR? (Nothing fancy, just plain predictable voice menus like a conference bridge.) I get stuck at Dial(), which seems to wait for hangup after the other end picks up. You can send dtmf to the IVR with the D option in the dial command. show application dial on the console will show you the syntax. 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
Re: [asterisk-users] Local hangup after Dial()?
Yuan LIU wrote: Another dumb question: Can a dial plan continue after local hangup when using Dial()? For example, [incoming] exten = s,1,Dial(Zap/1) exten = s,2,Congestion() exten = s,3,Hangup() --- Asterisk seems to insist that a dial plan is complete when Zap/1 hangs up and do not go into priorities 2 and 3. Use the h extension. ___ --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] kewlstart disconnect threshold
Stephen Bosch wrote: The reason we have these complaints is not because Asterisk doesn't detect the drop -- it's because a great many telephone companies don't do remote party disconnect signalling, or they don't do it properly. When people call for technical assistance they usually end up talking with someone who has no idea what Calling Party Control or remote party disconnect actually is. Case in point: I am with Telus in Alberta, Canada. By default, the loop drop (it's actually a battery drop, as near as I can tell, but kewlstart will detect both) occurs after more than a minute. On some lines it doesn't A minute is rather long. CPC when working should be almost immediate (see Mark's kewlstart test). What happens if you change it to loopstart? Does asterisk detect the drop? My analog line with Singapore Starhub does a battery drop after about 90s (i can hear a short crackling sound), but it's not detectable (no console message with verbose 6) unless I set to kewlstart. Not even with hanguponpolarityswitch=yes. So, I guess zaptel doesn't do anything if you set signaling to loopstart. The long and short of it, busydetect maybe the best solution to force asterisk to hangup after the remote party hung up. 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
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Eric ManxPower Wieling wrote: Leo Ann Boon wrote: Eric ManxPower Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. I've never seen one. Tell that to my cheap analog caller id phone :) BTW, the sample zapata.conf in Asterisk also have the caller id names quoted. Maybe Mark can enlighten us :) 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
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Yuan LIU wrote: From: Leo Ann Boon [EMAIL PROTECTED] Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. Leo Ain't working. 27935GE3-B simply says unknown or displays a blank if the string contains quote. I know that I can configure a softphone (e.g., Xten) to display correctly, because it has a user id and a display name. Anything similar in Asterisk? Can post your zapata.conf? You need to ensure Asterisk is sending the FSK signal at the right time. This is from my zapata.conf: signalling=fxo_ks sendcalleridafter=2 usecallerid=yes cidsignalling=bell cidstart=ring 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
Re: [asterisk-users] 3 PCI slot with exclusive IRQ ? please advice!
Stephen Bosch wrote: snip ...and have zillions of dollars :) Industrial PCs are pretty expensive. Over here, they're actually quite reasonably priced. A 2U rackmount P4 D930 3.0GHz, 1GB RAM system with 4 PCI (32bit) slots starts around US$1K. 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
Re: [asterisk-users] kewlstart disconnect threshold
Good question. Anyone knows if the TDM-400 actually detect loop drops? Well, that's really what kewlstart (and loopstart) means. If it couldn't, then Asterisk wouldn't know that the call had been hung up, and hog the channel. For loopstart lines, I don't think Asterisk detects loop drops. If it does, we won't have lots of people complaining about asterisk not hanging up when the remote party hangs up. a quick grep of the asterisk source turns up only chan_vpb has any mention of loop drop, not in chan_zap nor in the zaptel driver. 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
Re: [asterisk-users] CallerID Name not available.
Shivram u wrote: Hi, An incoming call is redirected to another number by our asterisk server. In the incoming call the caller name is present but when redirect the call, the end receiver is not able to see the callerid name. The caller id number is visible. If you're calling PSTN, caller id name is not guaranteed to be supported. 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
Re: [asterisk-users] 3 PCI slot with exclusive IRQ ? please advice!
Alessio Focardi wrote: Hi, I'm looking for an hardware platform for an * installation that should have at least 3 PCI slot with no irq sharing whatsoever. Use an industrial PC with a backplane bus. You can easily get 3-4 usable slots in a 2U and 10-14 slots if you use a 4U. 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
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. 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
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Eric ManxPower Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. 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
Re: [asterisk-users] kewlstart disconnect threshold
Stephen Bosch wrote: Hi, folks: Can the loop drop detection threshold (normally defined in milliseconds) be set on the Digium TDM-400 cards? Most PBXs let you set this value. Good question. Anyone knows if the TDM-400 actually detect loop drops? 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
Re: [asterisk-users] NTL Hangup
Kyle Gordon wrote: snip Hi Leo, That appears to have done the trick. fxs_ls does seem to detect it hanging up more reliably. I don't know what the difference is, but it works :-) If there's any change, I'll be sure to let you know :-p No problemo. Glad to know it worked for you. Like Tzafrir said, this is one of the less documented aspect of asterisk. 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
Re: [asterisk-users] T1 Wire Level Tapping
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
Re: [asterisk-users] T1 Wire Level Tapping
Shane Spencer wrote: I wanted to know if there was a peekaboo factor to it all. You can flow data under a glass window :) Well - you can always use a logic probe :). Bridging does add a little latency to the whole thing. Why don't you consider a passive tap solution like the hi-z OpenPRI card from voicetronix? It doesn't cost much more than a solution based on digium hardware. 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
Re: [asterisk-users] Response on dialin - no extension
chester c young wrote: On a SIP phone is it possible to enter the dialplan when the user picks up the phone without having to wait for the user to press an extension? You need a phone with a hotline function. Consult your phone's user manual. 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
Re: [asterisk-users] T1 Wire Level Tapping
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
Re: [asterisk-users] Does X100P decode caller ID?
It is, and is identified by wcfxo as a Wildcard FXO: Wildcard X100P. So much for The DigitNetworks X100P is detected as an actual X101P card. IIRC, there were 2 Digium single FXO cards - the X100P using the Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps have 2 RJ-11 jacks. Functionally, they're all Winmodems - effectively just DAAs connected to the PCI bus. The Zaptel driver is responsible for the caller ID and DTMF detection. Maybe you have a borked card or it could be due to impedance mismatch. I know that the X101P only works with FCC 600 Ohm impedance. For other parts of the world, YMMV. 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
Re: [asterisk-users] Voicemail from sip phones
[EMAIL PROTECTED] wrote: Hmm. Nope. Still same thing. I added pedantic=yes both in the general context in sip.conf and in the user's context in sip.conf with no change. Just for fun, I also changed it to pedantic=no in each place with no luck either. (I stopped and started asterisk between each change). Other thoughts? check that your phone is not using *8 in its own dial plan. Also, do a sip debug and see that the phone is actually sending *8 to asterisk. 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
Re: [asterisk-users] Re: Delay in Call Distribution using the Queue Application
[EMAIL PROTECTED] wrote: Thanks for the info, is there a patch available for version 1.2 that adds the autofill option? Gavin Hamill has back ported some of the 1.4 queue features into 1.2. See his post to this list http://www.mail-archive.com/asterisk-users@lists.digium.com/msg171158.html 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
Re: [asterisk-users] Voicemail from sip phones
[EMAIL PROTECTED] wrote: Here's the debug output from the console, it's somewhat long. Could the key line be (towards the bottom) this? [Jan 28 20:39:10] NOTICE[31924]: chan_sip.c:13519 handle_request_invite: Nothing to pick up for OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. Ah ha - your features.conf has *8 (the default) for group pick up. 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
Re: [asterisk-users] Does X100P decode caller ID?
Yuan LIU wrote: From: Leo Ann Boon [EMAIL PROTECTED] It is, and is identified by wcfxo as a Wildcard FXO: Wildcard X100P. So much for The DigitNetworks X100P is detected as an actual X101P card. IIRC, there were 2 Digium single FXO cards - the X100P using the Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps have 2 RJ-11 jacks. Functionally, they're all Winmodems - effectively just DAAs connected to the PCI bus. The Zaptel driver is responsible for the caller ID and DTMF detection. Maybe you have a borked card or it could be due to impedance mismatch. I know that the X101P only works with FCC 600 Ohm impedance. For other parts of the world, YMMV. Leo Is DTMF pass-through and caller ID fundamentally different? This card does not seem to cause significant difficulty in DTMF detection. The high DC voltage As I mentioned all the DSP work is done in software. So there shouldn't be any fundamental difference. during ringing could be one factor. Any easy way to test serious mismatch? The 600 Ohm is actually on-hook resistance, right? Or is this audio impedance? The 600 Ohm is off-hook AC impedance for US and countries that follow the same specs - consult your local regulatory docs. card is indeed sold and used in North America. But, where in the world are you using it? 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
Re: [asterisk-users] Does X100P decode caller ID?
Yuan LIU wrote: A little googling made me realize that Asterisk demo may not be the best application to look for caller ID because it tries to pick up at first ring. So I zapped demo context with a plain one. This time, no more failed success. But Asterisk only receives New User, no matter which caller calls. (Callers can be correctly identified from other devices.) You need to know when does your carrier send caller ID, some carriers send between 1st and 2nd ring, others after 2nd ring. Try a Wait(1) before Answer to give asterisk a little more time to pick up the callerid. 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
Re: [asterisk-users] NTL Hangup
Kyle Gordon wrote: fxsks=1 #X100P Is your line truly a kwelstart line? try fxsls SNIP busydetect=yes You may need to add these 2 values to help the busydetect busycount=3 busypattern=375,375 busypattern tells asterisk how your busy tone sounds like, in UK it should be 400Hz 0.375s ON and 0.375s OFF. The busycount tells asterisk how many consecutive cycles it must detect before dropping the line. You'll have to determine the best value for your setup, by trial and error. Too low - you might get premature hangup, too high - you'll have to wait for a long time for the line to hangup. A value of 3 will cause Asterisk to hang up in about 2.1s. SNIP switchtype=national This is not needed for analog lines. signalling=fxs_ks Change to fxs_ls to match zaptel.conf SNIP I don't know the tone plan for NTL. They seem to use a different tone for hanging up from BT, but I'm not sure how to go about implementing any changes in the configs to reflect it. If it's different you'll need to modify zonedata.c in the zaptel directory. 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
Re: [asterisk-users] NTL Hangup
Tzafrir Cohen wrote: On Sat, Jan 27, 2007 at 07:40:31AM +0800, Leo Ann Boon wrote: Kyle Gordon wrote: fxsks=1 #X100P Is your line truly a kwelstart line? try fxsls And if the line is ls, indeed, what harm is there in setting it up as ks? I understand ks is ls with a wink start. In some cases, use ks on a ls line will cause bizarre problems. Consider, e.g. http://svn.digium.com/svn/asterisk-gui/trunk/tools/zapscan.c SNIP busydetect=yes You may need to add these 2 values to help the busydetect busycount=3 busypattern=375,375 this should have been progzone=uk , only it turns out that the UK progzone actually sets it to 400. The progzone uk is actually correct, 400Hz, 375ms ON and 375ms OFF. But ,I believe it's not actually used in the busy detector. See this explanation from Steve Davis on why busypattern was added to zapata.conf http://bugs2.digium.com/print_bug_page.php?bug_id=4830 I'd like to ask again: where are you using specific progzones and buzypatterns successfully? Those magic values should be better documented. I agree with you, this is voodoo magic :). I'd only figured out for myself by trial and error. 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
Re: [asterisk-users] X100P - zttools says red status
Charlie Grosvenor wrote: Yes the line is connected, a standard phone works fine when connected to the line. There're 2 ports on the card. Which port are you using? One of the ports is for connecting another phone in parallel to the card. 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
Re: [asterisk-users] NTL Hangup
Kyle Gordon wrote: Hi all, I'm currently on an NTL PSTN line, using Asterisk 1.2 and a X101P cheapo card. The problem lies with detecting when the far end has hung up. It fails to detect it, and will only cleardown when the silence timeout has been reached. Now, I've seen the thread at http://www.mail-archive.com/asterisk-users@lists.digium.com/msg32337.html, to which nothing has come of it. That was almost 2 years ago, so I was wondering if there's been any progress? 2 things: a. You need to show us your zaptel.conf and zapata.conf. b. Do you know the tone plan used by ntl? I guess it should be the UK standard. 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
Re: [asterisk-users] No D-channels available! Using Primary channel 16 as D-channel anyway!
snip zaptel.conf --- loadzone=uk defaultzone=uk span=1,1,1,ccs,hdb3,crc4,yellow span=2,0,1,ccs,hdb3,crc4,yellow I don't think yellow alarm is necessary unless you've been advised by your carrier. bchan=1-15,32-46 dchan=16,47 bchan=17-31,48-62 --- where span 1 is to the provider and span 2 is to the PBX zapata.conf - context=from-pstn switchtype=dms100 signalling=pri_cpe callerid=asreceived group=1 callgroup=1 pickupgroup=1 rxgain=0.0 txgain=0.0 channel=1-15,17-31 context=from-pstn switchtype=dms100 signalling=pri_cpe If you are connecting the second span to the 11c, shouldn't this be pri_net? And, since you're using E1 I believe both your switchtype should be euroisdn instead of dms100. 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
Re: [asterisk-users] No D-channels available! Using Primary channel 16 as D-channel anyway!
Kong Zhen Shin wrote: i tried without yellow as well.. and according to zaptel drivers, the yellow don't do anything, just put a yellow signal where there is nothing from the provider. and yes, i did put a pri_net on the span 2, the config is a typo.. thanks for reminding me.. but still i got those errors :( Did you change your switchtype as well? Have you tried swapping the spans? 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
Re: [asterisk-users] OT: Optimum voice problems.
C F wrote: 1. When they tell you that they are putting all your lines in a hunt, it realy is not a hunt but just CallForwarding No Answer/Busy, what Some PBX implement line hunting that way. So, you need to Answer before you do anything else. Otherwise the PSTN switch will cheerfully go on its merry way while you're scrambling to route the call. that means is that if I have asterisk setup to first ring a phone for 5 times and then go to an IVR and answer the phohe, it will go to the next line and stop ringing the first line, and therefore never end up in Voicemail or my IVR. 2. No CPC, hung channles, blank voicemails, and all the other goodies that come with no hangup supervison, is a daily thing. Make sure you configure your zaptel signaling correctly. If you have loopstart (fxs_ls) then the best solution is use busydetect. On my loopstart line, it will always hangup in 4s if the other party hangs up. Of course, in my loopstart setup I'll still get the occasional blank voicemail if the other party hung up just as asterisk goes to voicemail. But, that's rather normal for most voicemail system. 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
Re: [asterisk-users] OT: Optimum voice problems.
C F wrote: On 1/22/07, Leo Ann Boon [EMAIL PROTECTED] wrote: C F wrote: 1. When they tell you that they are putting all your lines in a hunt, it realy is not a hunt but just CallForwarding No Answer/Busy, what Some PBX implement line hunting that way. So, you need to Answer before you do anything else. Otherwise the PSTN switch will cheerfully go on its merry way while you're scrambling to route the call. I disagree about this, this is NOT line hunting, but CallForward No Answer. It's an ignorance from Optimums side to offer it as line hunting. IMHO, Regardless of how they market or implement line hunting, you'll still need to get your Asterisk box to answer the call before ringing your user's phone. Otherwise, their switch will just assume no answer and move on to ring the next line. Flame Retardant From a user's POV, there are no perceptible differences between call forward on busy/no answer and linear line hunting. In linear hunting, the switch will try a line and move on if it's busy or there's no answer after a set time. Call forward on busy/no answer will also work pretty much in the same way, albeit more slowly. The main difference is in the provisioning: configuring a single hunt group vs individual call forwarding setting for each number. /Flame Retardant 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
Re: Fwd: [asterisk-users] Some queries on g729 license.
Andrew Joakimsen wrote: Most of the Cisco phones sold cheap are UNLICENSED (global spare) thus you would not be able to purchase (or at least aren't supposed to) the smartnet contracts, you need to buy the license ($100+) and the contract ($10 or so) I'm always surprised by by the number of people who don't read the fine print :). Even if you have a new licensed unit, it's only licensed to run Skinny out of the box. SIP requires additional licensing. Back to the G.729A licensing, I just received a new 'low-volume' quote from Sipro. For 1,000 channels - it's US$6 per channel (US$4 for 5,000) just for the right to use G.729A. You'll still have to fork out money to separately licensed a working codec - unless you're happy with the suboptimal ITU implementation or Intel's IPP sample. One vendor we spoke to asked for US$2,000/year to license their G.729A implementation on top of the Sipro licensing. That works out to a total of US$8/channel if we use 1,000 channel in a year. Throw in the cost of license administration (because Sipro will require audit), the US$10 charged by Digium looks very reasonable. FYI. 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
Re: [asterisk-users] NAT solutions
Voip Asterisk wrote: I know that NAT is something no one really likes to talk about, but does anyone know how work with it elegantly? There are many providers which deal with it on a daily basis in fact they cater to it, is this possible to do with asterisk or does it require other exotic setups? I even know of a provider which uses asterisk with many different types of devices, and they handle all NAT config on their end even to the point of deciding to stay in the media stream or not (ie when two endpoints are behind NAT you almost have to stay in the media stream unless you got it figured out like skype does). What is the best way to work with NAT, and build a production system? Use a far end nat traversal appliance. Acmepacket , kagoor and Jasomi are some examples. 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
Re: [asterisk-users] Possibility to catch DTMF when 2 users are in a conversation
Antoine Fressancourt wrote: I will sum up the results of my investigations : - When canreinvite is set to yes, I manage to make a video call between the 2 parties, when I emit a DTMF signal, it triggers the playback of a sound clip correctly, but I can't playback a video clip. What's the format of the video clip? I don't think Asterisk supports all formats. And, shouldn't it be canreinvite='no'? - When canreinvite is set to no, The DTMF I emit is not detected by Asterisk, although I see the SIP INFO message in the SIP debug messages of Asterisk. Should be canreinvite='yes'. This might be a bug. On the other hand, in your case, even if Asterisk did detected the messages. Without being in the media path, it still won't be able to playback video to the endpoint. 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
Re: [asterisk-users] Audiocodes GPL
Andrew Joakimsen wrote: I have some Audiocodes units which appear to be running Linux, according to the unit's own System Log kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 Googling turns up: http://www.jungo.com/openrg/openrg.html OpenRG is a Linux based device platform. So, Audiocodes probably licensed it from Jungo. Just because the unit runs Linux, doesn't necessarily imply that there's a GPL violation. 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
Re: [asterisk-users] Possibility to catch DTMF when 2 users are in a conversation
Antoine Fressancourt wrote: Hello, Thank you Leo for your answer, I manage to do what I want perfectly when both the caller and the callee are set in SIP with canreinvite=no using SIP INFO method for DTMF. Now, I can't figure out why this can't work when I set canreinvite = yes with the same DTMF method. Running Wireshark on my machine, I see that the SIP INFO messages are sent to the Asterisk box running as a proxy, but the INFO message doesn't trigger any action. Relooking at your requirements, I'd say you must use canreinvite=no. Otherwise, there's no way for Asterisk to inject audio into the stream. 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
Re: [asterisk-users] Echo...
3. It seems to be only incoming calls that have an echo and only on the inside, the outside never hears one, what does this mean? Why don't you record the call at asterisk? Leave the zaptel settings as default, i.e. standard echo cancel and rxgain=txgain=0. Don't use MixMonitor, just leave everything as in and out. It will help you isolate whether it's PSTN echo. If you don't hear any echo in the recording, then the problem is most likely your phone. 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
Re: [asterisk-users] Echo...
Eric ManxPower Wieling wrote: *sigh* Any time a call hits an analog 2-wire circuit there will be echo. In normal PSTN only situations the echo is so FAST that you do not hear it. It is only where there is a high latency path in the circuit like a VOIP phone where you will hear the echo. Recording the call on Asterisk will not let you hear the echo because there is no high latency portion of the circuit. I'm fully aware of these facts. I only suggested this to help Ken isolate whether is the echo on his PSTN side. He has been tuning his FXO interface without results, which leads me to suspect he's got echo elsewhere. Ken also left out what kind of phone he's using. Ken should also read Cisco's excellent introduction on troubleshooting echo problems: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080149a1f.shtml 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
Re: [asterisk-users] Directory too difficult?
Colin, Thanks mate for the first laugh of the day. Colin Anderson wrote: I got a requirement list just now, with my comments inline: (showing it just for a giggle) User requirement: 1) Directory set up by name - If person calling does not know employee's name, how will they access? -Why, using app_telepathy.so of course! Is app_telepathy GPL'd (General Psychic License) or do we need a bunch of mind reading license lawyers. ^_^ User requirement: 3) Not all mobile phones have the albphabet on their dialpads, how do they access our directory? -Shout really loud. Telus should have a class action against it for selling Razrs with no DTMF. Wow, didn't know there are pulse dialed mobile phone. ^_^ What about speech recognition option for directory? Not necessary a full speech to text engine, just enough to recognize the names would do. 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
Re: [asterisk-users] Possibility to catch DTMF when 2 users are in a conversation
exten = 1234,1,Dial(SIP/1234) exten = 5678,1,Dial(SIP/5678) The SIP phones (X-lite) are configured to send DTMF's using RFC 2833 mechanism. I want to know if it is possible in Asterisk to catch a DTMF event sent by one of the phone to trigger an action, for example to play a sound/video clip to one of the phones. google for features.conf, But you'll need to keep asterisk in the callpath, i.e. canreinvite=no, otherwise the RFC2833 DTMF codes will only be sent between the end points. If you need to reinvite, then you might have to try using SIP-INFO for DTMF instead of RFC2833. 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
Re: Fwd: [asterisk-users] Some queries on g729 license.
David Thomas wrote: This is by far the most volotile list I have ever been on. I'm not sure that's exactly the reputation Digium/Asterisk is shooting for, but even so it does provide some much needed comedy relief. Alas, it was't even related to the OP's problem. He was just trying to figure out why his licenses are invalid after a server failover. It just happens that the backup server's license was overwritten by the primary server license (because of disk mirroring). Regarding the IPP-based unlicensed codec. IIRC when it first surfaced - the general consensus then was that we should not be telling people where to download the binaries. Anyone can download IPP from Intel and compile Readytech's codec on their own. But, please don't use this list to propagate the unlicensed (and not to mention mostly illegal) binaries. The same goes for requests for Cisco phone firmware, etc. If you don't have a TAC accout, get a smartnet contract or find someone who has TAC access. After seeing the G.729 pricing direct from SIPRO, I now take the shut-up and be thankful position. I think Digium has done us a great service by working out favorable pricing with SIPRO. Agreed. 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
Re: [asterisk-users] Dimensioning a 50 sip phone installation
Erick Perez wrote: The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx. Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw phones +voicemail and *no* call recording? Make sure there're no interrupt sharing issues. My old EPIA 1GHz with 2 LAN and 6 USB, had no IO-APIC. It took lots of trial and error to make sure the digium card was not sharing an interrupt. 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
Re: [asterisk-users] Some queries on g729 license.
Xue Liangliang wrote: Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? It shouldn't be a problem if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --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] Some queries on g729 license.
Xue Liangliang wrote: Hi, actutally it is kind of shareing storage, because we use drbd and vserver technology, the fail over is at vserver level, and vserver is synced through drbd storage. drdb - that's what I suspected. Off the top of my head, the fastest way is to reactivate using the new master's MAC. The proper solution is to only use drdb for data that should be shared like the conf and database. The license key portion should not be on a device that's being mirrored by drdb. 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
Re: [asterisk-users] Problems with park
I have followed the (very brief) instructions on voip-info.org titled Asterisk Call parking. Basically, I confirmed that features.conf was already set up properly, and made sure parkedcalls was included in my local context. If I dial in via the FXO and answer the call on x102, then hit transfer 700, I hear the announcement 701. When I hang up the phone, the LCD displays Transfer Failed and I get a bunch of orphaned channels. Are you doing a blind transfer or attended transfer? I'm assuming you're using the phone's transfer button. You may need to press transfer a second time to complete the transfer. 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
Re: [asterisk-users] Dimensioning a 50 sip phone installation
Erick Perez wrote: what if I go with full g711-no transcoding? remember that I will have an E1 coming in, so my usage can be up to 30 channels at once. if that is an overkill machine config, and for obvious reasons I cant use old hardware, what are your suggestions? I would suggest you go for a box that has redundant PSU. Most 1U boxes can't support redundant PSUs. IMHO, a 2U industrial PC with a single dual-core Pentium Dxxx 2.8GHz+ (or Xeon 3xxx) with hotswap RAID-1 HDD and PSU would be more than enough. I generally prefer 2U over 1U, because it's easier to cool and there's space to accommodate PCI cards of various sizes. 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
Re: [asterisk-users] over 200 queues, anyone?
Lenz wrote: You are correct, this is more or less the scenario involved - the problem is that people want to call a personalized line AND speak to the same subset of agents preferably. I have never seen such a setup myself - I have seen CCs with 30 or 40 queues, never 200 - so I was wondering if anybody ever trued something on these lines; or if there are better solutions to the same problem. Best regards You just need a single queue and use DNIS to differentiate the various tenants. This is a very typical setup for a virtual secretary service. 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
Re: [asterisk-users] over 200 queues, anyone?
lenz wrote: HI Gavin, wish we could do that! :) the problem is that they want to have personalized agents too - so that each client has its own line AND his own agents, so that they get back to speaking to the same people all of the time. SO we need many different queues to accomodate all those differences. Your sript looks very useful thoiugh! :) l. Why don't you 'invert' the problem? Group the agents into fixed groups and put each group in a queue by itself. Each tenant will be assigned a group queue. If you have 30 agent in groups of 5, you only need 6 queues to handle 200 tenants. Even if you put each agent in a group by herself, you're still looking at 30 queues as opposed to 200 queues. 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
Re: [asterisk-users] Dual Ringing Tones
Troy - Purple Oranges wrote: Hi all and Happy New Year. I have a couple of interconnected asterisk boxes connected to several providers. With one provider in particular (ATP in Australia) there are two ringing tones heard on outbound calls. It is not the end of the earth - I am not reselling our services yet - but it is strange being that none of the other providers we are connected to exhibit that behavior. I think your provider is providing early media. Check your sip messages, look for 183 with SDP in the response from the provider. 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
Re: [asterisk-users] Binary AGI Scripts
Lee Jenkins wrote:
Moises Silva wrote:
use agi debug command from the Asterisk CLI to see what is going on.
Also, the last time I checked, \n is needed at the end of any
command sent to Asterisk.
Regards.
Hi, sorry I have already done that, but did not mention it. The
output that is displayed when I turn agi debug on is simply the list
of env. variables being pushed out to the application and of course,
the last empty line.
After that is when my call to EXEC PLAYBACK is made and I get no
response.
As for \n, I think pascal WriteLn automatically appends a newline
character, but I have tried appending it myself too like so:
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile\n'); // no work
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile' + #13); // no work
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile' + #13#10); // for SG's.
Have you tried using the agi unit at
http://home.cogeco.ca/~camstuff/agiunitpas.txt?
Leo
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Re: [asterisk-users] Determining invalid extensions.
Eric ManxPower Wieling wrote: Leo Ann Boon wrote: Phil Finkler wrote: Hi all, I’m trying to incorporate using the i extension in my callplan to determine if someone enters an invalid extension. My internal extensions are all 3 digits (100-104). The problem is, the callplan doesn’t see that say, extension 600 is invalid, it just goes back to the beginning of the callplan and repeats. If I enter a single digit, it works perfectly. Anyone have any ideas? Here’s the incoming callplan. It's because 600 will match _XX. Why don't you just use the 's' extension, instead of '_XX.'? Because the s extension is only matched when there is NO dialed number. My bad for not being clear. I meant that he should send his incoming calls to the 's' extensions and do a WaitExten in 's'. Extension i is designed for use within an IVR. We use 4 digit extensions and use exten = _,1,Whatever to match invalid extensions. ___ --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] How accurate is show translation?
Eric ManxPower Wieling wrote: Leo Ann Boon wrote: Hi all, I'm using 'show translation' to help dimension my system, but I confused by the results I get. My 2 test systems (results below): an AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar results (D930 is slightly faster). Googling shows that others have similar results running on other CPU speeds 2.0GHz. Does show translation recalc 30 show any different results? Eric, Before I posted, I ran tests with various recalc values between 10 and 200. The results are pretty much the same, give and take 1ms on either side. Forgot to add, I'm running 1.2.11 on both systems. 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
Re: [asterisk-users] How accurate is show translation?
Vicky wrote: I tried it on a intel 3 ghz p4 box and a athlon 3000 768 mb ram running vista and host for centos 4 ( vmware ) considering the load on athlon running asterisk ( that too under vista plus vmware ) while intel 3 ghz p4 1 GB ram box was sitting idle with centos , there was hardly a 1 ms difference in show translation on both machines . Besides i just compared my p4's results to ur D930 results and there is no difference ( infact my g729 results are better than ) .. But this doesnt mean both are same dual core cpu's will definitely give much higher number of channel transcoding then lower p4's . Put both the box under some cpu load by other programs and then use show translation recalc 30 and you will see performance difference between them ;) Vicky, The point of the exercise is that you should run 'show translation' with no load to get the baseline value. Your results confirmed my suspicion that the value is not tied to the number of CPUs - which indicates that the test was run on only 1 CPU. My concern is why the performance plateau. It makes no sense that a 3GHz CPU should take the same amount of time as a 1.3GHz CPU - that is unless there's something else is holding back the transcoder. It's like those graphics benchmarks - at some point, all the CPUs show the same FPS because the refresh rate is the one holding up the CPU. At this point, I don't feel that 'show translation' is a useful indicator of actual transcoding performance. It's OK for relative comparisons but utterly useless if you need the figures for sizing purposes. 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
Re: [asterisk-users] Determining invalid extensions.
Phil Finkler wrote: Hi all, I’m trying to incorporate using the i extension in my callplan to determine if someone enters an invalid extension. My internal extensions are all 3 digits (100-104). The problem is, the callplan doesn’t see that say, extension 600 is invalid, it just goes back to the beginning of the callplan and repeats. If I enter a single digit, it works perfectly. Anyone have any ideas? Here’s the incoming callplan. It's because 600 will match _XX. Why don't you just use the 's' extension, instead of '_XX.'? 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
Re: [asterisk-users] How accurate is show translation?
Tzafrir Cohen wrote: If you had just one call, then adding extra CPUs wouldn't have helped. 'show translations' mainly helps you compare different codecs. It is also handy as a benchmark because it's there. However I agree with you that with 1 call, more CPU won't help. I'm just surprised that a 3GHz CPU is not much faster than a 1.3GHz CPU. I'm actually trying to find an analytical model to dimension an asterisk box. I need to transcode 120 channels of IAX (speex) into g711 to fed into 4xE1. My current guesstimation is a single Intel D930 should be up to the job. Without hard numbers, it's not very convincing. This is one aspect of asterisk that's annoying - you can't size a system reliably without resorting to lots of empirical testing. IMHO, this usually leads to over-engineering which drives up the cost. Regards and happy holidays. Leo 'In God we trust, others must have numbers.' ___ --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] more than 32 callgroups pickupgroups
Conrad Wood wrote: On Thu, 2006-12-21 at 12:07 -0700, Douglas Garstang wrote: I'm no C programmer, but is this 32 limit just an array definition somewhere? Wouldn't it be a no brainer to track it down and increase it so some very large number? I think pickupgroup is defined as 'unsigned int' somewhere in channels.h. 32 is the number of bits in a 4-byte integer, so it's probably using a bitmask to define which pickupgroups a channel belongs to. I suppose if you are on a 64bit machine/os you /could/ try to make it a 64 bit pointer, but you should really check the source a bit more to see how exactly it's accessed (I didn't!) I don't know any .32bit integers on 32bit machines. gcc supports a 64 bit integer on 32-bit process via the long long or int64_t types. 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
[asterisk-users] How accurate is show translation?
Hi all, I'm using 'show translation' to help dimension my system, but I confused by the results I get. My 2 test systems (results below): an AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar results (D930 is slightly faster). Googling shows that others have similar results running on other CPU speeds 2.0GHz. At first glance, it would look like the AthlonXP gives better bang for the buck :). But, I'm sure that are other reasons. I know show translation times how long it takes a convert 1s of full duplex audio. I suspect the test is using a single CPU (since it's in a single thread) and there are some constant overheads that makes a 3.0GHz produce the same numbers as a 1.3GHz. I would love to hear how others are using the results from show translation in system dimensioning. So far, I feel that dimensioning an Asterisk box is still mostly guesstimation :). Currently, I'm using the 30MHz per call rule to dimension. Leo Results from show translation: On the athlon: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - 3 2 2 2 2 1 3101113 gsm - - 2 2 2 2 1 3101113 ulaw - 3 - 1 2 2 1 3101113 alaw - 3 1 - 2 2 1 3101113 g726 - 3 2 2 - 2 1 3101113 adpcm - 3 2 2 2 - 1 3101113 slin - 2 1 1 1 1 - 2 91012 lpc10 - 4 3 3 3 3 2 -111214 g729 - 4 3 3 3 3 2 4 -1214 speex - 4 3 3 3 3 2 411 -14 ilbc - 4 3 3 3 3 2 41112 - On the D930: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 5 91014 ulaw - 2 - 1 2 2 1 5 91014 alaw - 2 1 - 2 2 1 5 91014 g726 - 2 2 2 - 2 1 5 91014 adpcm - 2 2 2 2 - 1 5 91014 slin - 1 1 1 1 1 - 4 8 913 lpc10 - 3 3 3 3 3 2 -101115 g729 - 3 3 3 3 3 2 6 -1115 speex - 3 3 3 3 3 2 610 -15 ilbc - 3 3 3 3 3 2 61011 - ___ --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] Match a Numer - then continue with dialplan
Douglas Garstang wrote: I just know someone is going to ask 'why would you ever want to do that?'. Here's my answer. We have two companies, each with a dialplan similar to what's below. In the event that the number being dialled does not match any number within our OWN company, we want to set the caller id to be a generic one for the company, NOT one for the user. This is a pretty normal requirement that most companies want. So, in the event that the logic flows beyond coo1_OnNet, we want to reset the caller id of say, 3254001 Doug, to 3254000 Widgets Inc. If there was a way to match against a number in the dialplan, and then continue execution after that point, we could put this statement at the end of the coo1_OnNet context and it would all be sweet. Without that, I don't have a clue how to do this... unless we stick with out current 3,000 line python script. If you're not using realtime to store your SIP registry, you should be able to look up the number in the family SIP/Registry (case sensitive) using the DB functions. If you're using realtime, then you'll have to do an SQL query. 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
Re: [asterisk-users] Asterisk + Orion E1 GSM Gateway
yusuf wrote: Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Sanity check: Have you read the fine manual :)? I understand Orion makes both ISDN PRI/Q.SIG and MFC/R2 type E1 channel banks. If it's the PRI type, standard zaptel with the appropriate NET/CPE setting on the CB should be ok. If it's a MFC/R2, then you'll have to try unicall. 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
Re: [asterisk-users] Motherboard 3.3V PCI for TE412P
Jesus Mogollon wrote: Hi all Does anyone know of any motherboards with PCI slots that can take the TE412P card? Is there such a MB for Athlon 64 or P4 procs? I have a TE410P working with an ASUS P5MT mobo with Intel Pentium D processor. ___ --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] IBM Server / USB Ports
Matt wrote: I see that the digium card doesn't share the IRQ however Digium has recommended diabled USB still... additionally the Digium card is on 169 which isn't a valid IRQ.. how can I find out what it is sharing with? the tdm card is not sharing an interrupt with your USB. It's your LAN card. 169 is valid if you're running on uniprocessor IO-APIC or SMP kernel. Guess you have to look elsewhere for the source of your crackling. Try unloading the USB modules from the kernel, i.e. rmmod uhci_hcd 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
Re: [asterisk-users] IBM Server / USB Ports
Matt wrote: So you are saying that the card is on it's own IRQ and is not sharing anything with anything? I realize the eth0 and usb are sharing, but am not too concerned about that. What's your zttest result and did zttool reported any irq misses? If zttest is mostly 99.98%, then the zap device is fine. See http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting 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
Re: [asterisk-users] Running Asterisk on a Home rotuer
Dovid B wrote: - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, December 08, 2006 12:07 PM Subject: Re: [asterisk-users] Running Asterisk on a Home rotuer Dovid B wrote: tacking pn = adding on - sorry for not being more specific. I have seen that people in the past have used a linksys router to run asterisk. It would be to expensive to bring in a PC for every location. So we want to import cheap home routers put asterisk on them as use them as the go in between the IP phones and the asterisk server. Check with Brian Capouch. He deployed Asterisk on Linksys WRT54G in some rural areas. Caveat here: Cheap = not enough horses :). Don't expect to pass many calls through one of those things. You might want to look at deploying a lightweight SIP proxy on the router instead of asterisk. Leo Ping Brian Capouch. Anyone have his contact info ? See his post to the dev list. Not sure if the address is still valid. http://lists.digium.com/pipermail/asterisk-dev/2004-December/008181.html ___ --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] RDNIS question
Julian Lyndon-Smith wrote:
snip
this works well, with one exception: when I take the call on the
mobile, the callerid info is the number of my switchboard. I presume
that this is because I am dialling out from the switch board.
Enter RDNIS. I added an extra line to the dialplan
snip
2 issues here:
a. For PSTN, you should use Set(${CALLERID(num)}) to set your outgoing
caller id.
b. Does your PSTN line allow you to set the outgoing caller id? If
you're using analog, it's not possible. For ISDN (both BRI/PRI), it's
usually possible if you subscribed to the feature. But, you're normally
only allowed to set the caller ID to one of the numbers allocated to
your ISDN line. You can't just set it to any arbitrary number (Note:
might work if your local exchange is mis-configured).
Leo
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Re: [asterisk-users] Running Asterisk on a Home rotuer
Dovid B wrote: tacking pn = adding on - sorry for not being more specific. I have seen that people in the past have used a linksys router to run asterisk. It would be to expensive to bring in a PC for every location. So we want to import cheap home routers put asterisk on them as use them as the go in between the IP phones and the asterisk server. Check with Brian Capouch. He deployed Asterisk on Linksys WRT54G in some rural areas. Caveat here: Cheap = not enough horses :). Don't expect to pass many calls through one of those things. You might want to look at deploying a lightweight SIP proxy on the router instead of asterisk. 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
Re: [asterisk-users] G.729E
Michael Iedema wrote: Greetings list, Does anyone have any information (providers' support) about G.729E? Voip-info.org came up empty, the implementers guide from the ITU wants my credit card and the rest of the pages I found simply made a few comparisons between it and iLBC. From what I understand, the codec is supposed to play nicely on lower power hardware but I can't find much more info than that. It's a 11.8kbps codec that's supposed to improve the quality for non-voice signal etc. And, according to this http://www.voiceage.com/prodg729.php, it's more CPU intensive than G.729A. It's listed at 25MIPS vs 10MIPS for G.729A. Also needs more RAM compared to G.729A. I don't think it qualifies as 'play nicely on lower power hardware'. 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
Re: [asterisk-users] Help with dial plan - two attempts at calling agent before logging agent off?
snip I have tried setting another variable as a counter with some logic tests to see the number of attempts to call the agent, but this is failing as the variable appears to be lost when the call goes back to the queue. Local variables are destroyed once the call terminates. You'll have to use a global variable (yuck) or use the DB functions. 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
Re: [asterisk-users] Asterisk: SIP Gateway or Proxy
yusuf wrote: Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? Short answer: Gateway. This has been discussed to death many times on this list. Please search the archive for more details. 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
Re: [asterisk-users] SIP Port 5060
Brad Templeton wrote: snip My understanding was that the port= field on a particular SIP channel defines the port used at the remote end, ie. The user's phone will be talking on port X of their IP address, it does not alter what SIP port Asterisk is listening on on the Asterisk box. The host and port pair is used by Asterisk to identify a static peer. If host=dynamic, then Asterisk will use the host/port from the Register message. That is what bindport does, and that's a global setting, I was not aware you could have multiple bindports but that is very useful if it works. 1.2 certainly doesn't support the use multiple ports. If you put bindport=5060;6060, only 5060 is use. snip a) You might get around carrier SIP blocking If the carrier is really determined to block you, they will use content inspection rather than just blocking by port number. The only viable solution is by VPN or SIP encryption. I know there are various proxies that support hash encryption of SIP/RTP packets to get around the blocking. The only problem is finding equipment that support the hash encryption which is vendor specific. IIRC, opensipstack has a working implementation of such an encryption scheme. 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
Re: [asterisk-users] Asterisk and TDM400P ?
Noc Phibee wrote: Thanks Giogio, but no i don't have this module bye Check your zapata.conf. Your signalling and channel settings are wrong for FXO module. signalling=fxs_ls channel= 4 FXO module use fxs signalling, FXS module use fxo signalling. 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
Re: [asterisk-users] Asterisk and TDM400P ?
Noc Phibee wrote: thanks for this information, but no change: Nov 24 10:32:42 WARNING[6346] chan_zap.c: Unable to specify channel 4: No such device or address Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to open channel 4: No such device or address here = 0, tmp-channel = 4, channel = 4 Nov 24 10:32:42 ERROR[6346] chan_zap.c: Unable to register channel '4' Nov 24 10:32:42 WARNING[6346] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 10:32:42 WARNING[6346] loader.c: Loading module chan_zap.so failed! Can you check if your /dev/zap directory is created correctly? On my machine with a TDM400P with 2xFXS and 2xFXO. [EMAIL PROTECTED] ~]$ ls /dev/zap/ 1 2 3 4 channel ctl pseudo time If you don't see anything then you'll have to check if your security setting is prevent access to /dev/zap. 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
Re: [asterisk-users] Card don't hangup but Asterisk hangup
Jesus Jimenez wrote: Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. I'll apreciate all your suggestion. Greetings, txus. The asterisk output: -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' zapata.conf [channels] language=es context=from-pstn signalling=fxs_ks Is your PSTN line really kwelstart? If it is loopstart, please use fxs_ls and busydetect. 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
Re: [asterisk-users] Cisco media gateways in general
Pavel Jezek wrote: is possible to control ci$co gateway from asterisk via mgcp? i.e. asterisk as mgcp call agent? PJ I've tested the old Cisco ATA-186 MGCP (firmware 2.16) with Asterisk 1.2. Works pretty well. 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
Re: [asterisk-users] Recordings.
Marcus Franke wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael Welter wrote: Has anyone tried recording to a ramdisk? To an NFS mount? Was there a benefit? RAM disk? Interesting idea, but what to do in case of a server crash loosing these recorded files? Or use something like Gigabyte i-ram, PCI SATA RAM disk with battery backup. http://www.gigabyte.com.tw/Products/Storage/Products_Overview.aspx?ProductID=2180ProductName=GC-RAMDISK You will get very angry customers if you have to explain them, that your server, where you did record their complaints, crashed and lost their problems :) Id recommend this as a cache drive where you would move the files away from, when the call is finished. But thats extra cpu cycles and it would be kind of an effort to trigger the move the files after call is finished.. Well, you should always archive the calls automatically to CDR, DVD-R or tape. Most commercial call recording systems will do. Some will 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
Re: [asterisk-users] Re: Asterisk to listen for sip traffic on 80 and 5060
kjcsb wrote: I have Asterisk listening for sip traffic on port 5060. I want to allow users to use either port 80 or 5060 if they want. Hopefully this will avoid some firewall issues. If you're think that by sending SIP on port 80 will fool the firewall into thinking it's HTTP traffic, then I'd suggest you look elsewhere. For a start, most firewalls only allow HTTP on TCP/80 not UDP/80. 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
Re: [asterisk-users] Re: Monitor, MixMonitor and volume levels
Steve Davies wrote: *bump* No suggestions at-all? Does anyone use this facility in a similar way and NOT have problems? Check the gain on your ISDN interface. The monitor command doesn't modify the volume by default. Have you tested calls via IAX to your cell? 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
Re: [asterisk-users] some simple newbie help with dialplan needed...
Evert wrote:
Hi! :)
Thanks for the tip. I'm almost there now, the only problem that I have
left is that I do NOT want Asterisk to check whether the extension
entered is valid. In the current setup Asterisk will refuse to forward
the call since it thinks the extension is invalid... :-/
Is ${SERADDRESS} the name of a valid SIP peer or just plain ole IP
address? It should be a SIP peer.
Leo
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Re: [asterisk-users] Re: Port Range
Zeeshan Zakaria wrote: By default asterisk install rtp.conf with following settings: [general] rtpstart=1 rtpend=2 I usually change rtpstart to 10001 so 1 can be used for webmin. On some servers I keep rtpend on 14000 (no You should stick to even numbered ports. For each even number RTP port, the next higher odd number port is usually the RTCP. 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
Re: [asterisk-users] Re: Port Range
Zeeshan Zakaria wrote: I'll keep that in mind for future. I read about using 10001 as start port on Nerd Vittles website. Is there some good material online to read more about RTP, SIP, RTCP and UTP? Search the RFCs. 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
Re: [asterisk-users] Zap channel shows answered as soon as outbound ringing starts
shadowym wrote: Just to follow up on this, After some testing tonight I found the following. Watching the Asterisk CLI, when making a call from an extension to a ZAP channel the channel shows as answered as soon as the zap line starts ringing. That would explain why Followme was not working. It thought the PSTN line was answered It's the correct behavior because you're using analog FXO. Call progress is only available with digital lines of if you turn on the analog call progress detection. Analog CP is very experimental, and for the 1.2 branch only usable with US tones. Use ISDN BRI or PRI if you want proper call progress. 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
