Re: [Asterisk-Users] An anniversary and a lament for FXOs
On Nov 5, 2004, at 11:25 AM, Greg Boehnlein wrote: As for the Interrupt issues and PCI issues, Digium isn't really responsible for broken PCI busses. You need to be complaining to the manufacturer of the $35 motherboard for that. I do agree, however, that a community developed Hardware Compatibility List would be a good thing. What about an expensive Supermicro dual Xeon PCI-X system with 1GB ECC RAM and a hardware RAID controller (it was SATA, though)? Echo was noticeable even on SIP-to-SIP calls internally with the system, with all sorts fo tweaks to tx/rx gain. Supermicro, too. Oh yeah, and we were on a T1 PRI, which is not *supposed* to have echo. Unfortunately when I left the company, they finally replaced the phone system to get rid of the echo and customer complaints. A motherboard list would be REALLY great, indeed. Best regards, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Echo SIP-T100P-PRI
On Aug 19, 2004, at 5:33 PM, Rich Adamson wrote: Mike Schwartz wrote: I'm experience echo on outgoing calls: Snom 200 Asterisk T100P PRI called party I am getting echo on the Snom 200 phone. The called party does not hear the echo. snip When that discussion was going on a few weeks ago, the echo issue seemed to have been narrowed down to two possibiliites; 1) interrupt service latency, or, 2) PCI bus latencies. Processor speed does not seem to be a driving factor as noted above. I've not heard anyone (as yet) come up with the tools or process for actually identifying the root-cause. Would be nice for those of us that aren't programmers. Some more echo food for thought. It's most noticeable on very short, hard sounds (like CH), so as someone mentioned, reverb might be the right description. I've spent the better part of several hours experimenting with various combinations of adjusting taps from 32 to 256, echowhenbridged on and off and txgain adjustments. I just flat can't get it to go away... I'm also one of those luck ones with a Supermicro box (dual Xeons and plenty of RAM). How in the heck would/should I go about figuring out what the interrupt service latency or the PCI bus latency is doing. Any other thoughts on the front? I'm using GS phones so maybe their echo can algorithms are to blame... hmmm... Here's to hoping, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B Dead?
On Jul 23, 2004, at 1:22 AM, Andres Junge wrote: What is a RMA? Return Merchandise/Materials(something like that) Authorization. It's a number from the mfr, that when the product arrives with it on the box, tells them to expect some dead hardware. rt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B Dead?
I just received in the mail my TDM04B card and put it in the computer, now the computer won't even show the video card bios or the post screen. From the digium website I could not find any specific requirements for the pci card, like 32 or 64 bit slot. The motherboard for the computer I put it in is an Asus A7V333 with PCI 2.2 compliant slots. I am thinking that maybe I just got a dud card. Is there anything I need to change or I can test to see why it is not letting the computer boot? Any help is greatly appreciated. This may be a stupid question, but did you plug in the power on the molex connector on the card? -- Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on a PRI
I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Welcome to the club. ;) You have the same exact problem I've got. The only difference is I'm using dual Xeon 2.4s and a Supermicro SuperWorkstation 7033A-T (X5DAL-TG2 motherboard http://supermicro.com/products/motherboard/Xeon/E7505/X5DAL-TG2.cfm ). Echo training=800 on a recent CVS helped, but did not totally resolve the issue. Best regards, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on a PRI
I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Oops. I need to correct my last post: I don't have the PBX in the mix. My config is dual Xeon 2.4s, 1GB RAM, HW SATA RAID, SuperMicro X5DAL-TG2 motherboard connected to: Telco - PRI (T100P) - Asterisk - SIP Phones (Budgetones 102s/Snom 200) The premise is still the same though: echo present despite our digital PRI that *should* make this impossible. It's usually only echo on our side when calling out as has been discussed here previously ad nauseum with no one being able to really figure out its source. I wish I knew where to really start poking around to try to help get to the bottom of this. Best regards, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange echo problem
We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS line, I hear everything I say in my earpiece about 1/4 second after I say it. It's very irritating.We have tried 2 different * boxes, using 2 different T1/PRI cards f/ digium. After calling digium about it, we set echotraining to 800 in zapata.conf. It got better but was still there, if I turn the volume down on the phone, it does almost go away, but it's still detectable. No where near as clear as calling a person that has a PRI or channelized T1 for phone service. The POTS persons we call that we do have the echo issue with all say the call sounds perfecto to them. Am I missing something obvious? We experience the exact same issue, and like Rich said in a subsequent post, I'm thinking there's a gremlin hiding somewhere in the * code. Everyone said you shouldn't have to even use echo canceling on a T1 PRI, but we do or we get serious complaints, instead of consistent minor complaints. FWIW, it still was around in the 6/29 CVS and we just updated again last night. For us the echo is a slight faint echo now that we implemented the echotraining=800, but it's still there. We haven't touched TX/RX gain. We can also give anyone access and a SIP account if that would be helpful. Best regards, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BudgeTone hold?
On Jun 11, 2004, at 8:04 PM, Seth Mattinen wrote: I can't seem to make the Hold button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press Hold button 3) Hang up phone You can sorta do this by pressing the speakerphone button prior to placing the receiver on the hookswitch. When you pick up the receiver, just press the hold button again to resume your call. I too found out the hard way. HTH, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail notify to external number
I'm not aware of any way of doing this currently, but this has made it to the planning board of Voicemail3... the timing for which is unfortunately undetermined at the moment. HTH, Ryan Thrash On May 25, 2004, at 11:14 AM, Bruce Komito wrote: When a user has voicemail, I would like * to call the user at a pre-determined number (internal or external) and play a message that the user has voicemail, and then give the user the option to login to voicemail and pick up the message. I know about the externnotify feature, but I don't see a way to use it to accomplish what I want. I've checked the archives, etc., but I don't see that anyone has ever done this. If you have, please respond. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream transfer, park and conference
Your English is just fine. :) What's your extensions.conf and sip.conf for your Grandstreams look like? What are your options in the GS config webpage for: 1) NAT transversal (and are you behind a NAT firewall) 2) Send Flash event 3) Send DTMF Best regards, Ryan Thrash On May 3, 2004, at 8:51 PM, Ing Isianto Istiadi wrote: I have 2 grandstream budgetone 100 series. I can call allright, but I cant do call transfer, park and call conference. (all features works with tdm devices) the The budgetone using 1.0.4.55. 1. If I called using sip to sip (from phone a to phone b), I cant transfer it at all or parking it or dial to conference. 2. if the call come from pstn, then the first phone who answer can park the call, and be picked up by the second phone, but after that the parking stuff wont work anymore. (it seems asterisk doesnt recognize #) 3. Ive already set dtmf to info 4. It seems on case 2 above, that even the # works for the first call from pstn to sip, but asterisk only recognize at most 2 digit after # being pressed (for example, I have ext 700 to park the call, when I look at * console, it only receive 70) Thanks and forgive my English ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Cancellation Feature
I would also offer feedback that we too have random calls with echo on our end, that can't be traced to a reproducible event. It's very odd and can be frustrating, as it's a big distraction for those that don't know better. Like a bad cell phone connection when you hear yourself talk. For us, this happens in a pure SIP environment on a network switch dedicated to Asterisk, a T1 PRI and 18 SIP handsets. HTH, Ryan On Apr 22, 2004, at 1:37 PM, Brent Franks wrote: I do feel the echo cancellation does need some work. Currently, other than listening to users, there is no way to benchmark or trouble shoot echo problems. We find that 2 to 3 out of every 20 calls will experience echo. While echo is a problem that naturally occurs from SIP - PSTN and vice versa, I am still baffled by the fact that the cancellation works randomly. snip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension buttons
I can verify that snom 200s will support up to 5 line appearances and you can happily change back-and-forth between them. Now actually successfully transferring those calls when more than one call is in those line appearances is another thing entirely, when using the soft keys or the transfer button. Quite frustrating, actually. Like building a sure to win race car and forgetting to put the lug nuts on the wheels... HTH, Ryan On Apr 22, 2004, at 11:22 PM, John Todd wrote: At 7:33 PM -0500 on 4/22/04, David Krider wrote: I've downloaded the entire archive of articles and searched through them for an answer on this, but I haven't come across one yet. I'm looking to replace a small phone system in my church with Asterisk, and I'm stuck looking for phones. I know that the staff are going to want a button for their commonly-called extensions, but I'm having trouble finding phones that have, say, 10 programmable buttons for this sort of thing. I'm left to conclude that most phones can do this sort of thing by clicking through some combination of buttons. However, it would seem that the average price for a nice SIP phone eliminates the possibility of just ordering some to find out. Can someone please tell me how this is handled in general? For instance, the Polycom 600 doesn't seem to have ANY buttons that can be programmed for particular extensions, but I have to think it can do so fairly easily. Perhaps these phones are being sold primarily for very large business (like my Fortune 500 company) where you use a directory for numbers, and have only a few buttons for programming. I guess the follow-on question is just to ask: what phone would be good for a 12-18 extension office, where people want to quickly ring up others because of long walks between the offices? Thanks, dk -- David Dunkirk Krider, http://www.davidkrider.com Acts 17:28, For in Him we live, and move, and have our being. Open Source: Will you use the power for good... or for AWESOME? (I'll assume you're asking for line appareances, and not single-button push-dial. If you're looking for single-button push-dial, something even as simple as the 7960 can work for you) What you're looking for are line appearances, and no, they're not common with SIP phones since the methods to support the presence activity on those buttons is a bit of a pain, and still fairly fresh off the presses (if off the presses at all.) The phones that are rumored to support these new methods (I haven't any of them in my hands at the moment, so I can't say if they do or don't): - SNOM 220 - Sayson 480i - Inter-Tel 8620 and 8662 - not sure about the Polycomm 500 and 600 - others? Now, even though some of these phones might actually be shipping, that doesn't mean that Asterisk supports line appearances. There is no SUBSCRIBE or NOTIFY method that currently exists for SIP phones with Asterisk to handle line appearances (though that might not be too difficult.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream and stun
FYI, with 1.0.4.55 and NAT set to off (but with the * config set as nat=yes), I'm able to bypass stun servers completely with a GS phone as well. HTH, Ryan On Apr 18, 2004, at 5:08 AM, Brancaleoni Matteo wrote: you don't need stun to make GS work under NAT with * Just set NAT=yes into the GS, and leave the stun server addr entry empty. And set nat=yes into the sip.conf entry. Il dom, 2004-04-18 alle 11:26, Richard ha scritto: Hi, I noticed some issues with how grandstream handles stun test. GS is running version 1.0.4.50. First I reset the NAT router. Then reboot GS, get results of restricted cone. Immediately reboot GS, get results full cone. I tried quite a few public and commercial stun servers. Also tried different model/version of linksys routers. I always got the same issue. Winstun on the PC doesn't have this issue. Some ngrep on the stund 0.91 on Fedora linux revealed winstun had about 20 UDP packets back and forward. However GS only had less than 10. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 Admin Password
I *think* the default password is (all zeros). HTH, Ryan On Apr 17, 2004, at 10:38 AM, WipeOut wrote: Pertti Pikkarainen wrote: There is a way. Right after reboot, and when you see the first text, hit any key and you will get a 'boot menu'. Give the phone an ip-address and define a tftp-server. The bootfile must be named snom200.bin ( e.g rename the latest snom sw ). After you have succesfully got it to download the code, the phone is also resetted to factory defaults. You will see erasing flash etc. If the download fails the phone will use the sw it has got and there will be no change in the config either. --Pertti Hmm.. That would mean I would have to setup a TFTP server which is a hassle.. :) I was hoping that there was some key combination or a reset busson hidden somewhere.. snip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
First, check zapata.conf to see what is in there. Next, I've not heard of any luck with the name portion on T1s, but the number can be changed for us. HTH, Ryan On Apr 16, 2004, at 10:30 AM, Mike Machado wrote: You can usually get CLI on an EM robbed bit T1 by configuring it right. Instead of just sending you the DNIS as a string of DTMF they usually send *cli*dnis*. The DNIS and CLI may be swapped, and there may be less than 3 *s in the string - wonderful consistency, eh? :-\ I am getting CallerID and DNIS on the inbound calls. What I really need is to be able to set callerID on outbound calls. I am trying to set the callerid using SetCIDNum just before using Dial on a zap channel, but it looks like the switch guys have it set to always stamp the same callerID on the my outbound calls no matter what I put in SetCIDNum or what channel on the T1 I use. Is this a misconfiguration of the switch or a limitation of the signaling protocol? If its the switch, can you give me any pointers as to what I could ask them to look for, or if its the protocol, do you know any other signaling protocol that lets me set outbound callerID (besides PRI)? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 188 and fax
On a Grandstream ATA and CVS HEAD from last night, and with echo off, I'm able to receive faxes. With echo on, no go. HTH, Ryan On Apr 15, 2004, at 1:58 PM, Ariel Batista wrote: Osvaldo Mundim wrote: Hi, Does anybody have ATA 188 working with any kind of fax machine? I've tried many different configuration following the Cisco Online Manual and I couldn't get this working with Asterisk. I don't know what the difference is between the 186 and 188 other then the extra nic port. But we gave up on the 186 for doing any fax or data calls. We switched to Sipura-2000 and using Ulaw faxing works. Data calls well we can get them working but only at 28,800 bps. Good luck I were trying do change the ATA Connect Mode and Audio Mode reading the (http://www.cisco.com/en/US/products/hw/gatecont/ps514/ products_configuration_example09186a00800d698e.shtml) and allowing all codecs on Asterisk and did not work either. best regards Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 update - timestamp issue within iax pkts
Would this fix also help random quality issues both on a LAN and also with a remote SIP based installation running CVS from 3/22. We're having too frequent complaints in conjunction with Grandstream phones and very stuttery/choppy sound, usually outgoing to land lines, to the point of being unintelligible, but also on internal voicemail messages. Our voice LAN is a dedicated 100Mb ethernet switch for voice, and no general network traffic. Thanks, Ryan On Apr 14, 2004, at 4:26 PM, Rich Adamson wrote: For those that might be using Cisco 7940/7960 sip phones and placing calls across an iax2 link, we think the voice quality problem has been identified and corrected. The dev cvs should be updated as of about 3:30pm CDT today (April 14). snip If you use iax2 and Cisco sip phones, please update from cvs and give it a try. I am not aware of any other sip phone vendor that is sensitive to these timestamps, but there could be others. Keep in mind the fix addresses iax2 timestamp problems at the distant end, therefore iax updates will be required at both ends of an iax link to address the end-to-end audio quality problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small question 3 way calling
The GS phones do not currently support conferencing on the phones using the conference button. You'll probably have better luck setting up a conference room, help with which I'm absolutely worthless... The on-phone conferencing should be addressed in a future GS firmware revision. HTH, Ryan On Apr 13, 2004, at 10:46 AM, Anthony Law wrote: According to voip-info.org, 3 way calling: Normally implemented by the phone I am using a Grand Stream 100 and not able to make this work. I can dial out to 1st number then with the flash button I am able to dial out again to a 2nd number. I am not able to bind them together into 1 conversation. Is there something I have to set on the phone config or in sip.conf?? Anyone knows? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TAPI driver
Could you post a link? Thanks! On Apr 13, 2004, at 2:59 PM, Nick Knight wrote: Just a quick note, I have been putting together a TAPI driver for Asterisk, this enables the user to perform things like click to dial from any TAPI enabled app (such as outlook or ACT etc). At the moment it is very basic and can only perform click to dial but further functionality will be coming. It uses the Asterisk manager to place calls. Please feel free to use it - not much documentation as yet but will be coming, can be found on sourceforge project name asttapi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] application Directory (Modified by Ryan Thrash)
Sent 12 hours ago and it never showed up (slightly reworded here). Sorry if this is a duplicate: - Scenario: a person selects an Auto Attendant option that fires off the Directory application (CVS circa 3/22). Three questions: 1) How do they escape if they didn't mean to go there in the first place (without having to hang up...)? Config of entry into the vertex directory below: exten = 1,1,Directory(vertex) exten = 1,2,Goto(s,200) 2) Why is there a five second pause before the directory instructions start? 3) Why no option for first name (without recording your own custom message and reversing names in voicemail.conf)? Thanks, Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] application Directory
Let's say an unsuspecting soul accidently selects the Directory option from an Auto Attendant (CVS circa 3/22). Three questions: 1) How do they escape if they didn't mean to go there in the first place (without having to hang up...)? exten = 1,1,Directory(vertex) exten = 1,2,Goto(s,200) 2) Why is there a five second pause before the directory instructions start? 3) Why no option for first name (without recording your own custom message and reversing names in voicemail.conf)? Thanks, Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] application Directory (Modified by Ryan Thrash)
On Apr 9, 2004, at 9:52 AM, Tilghman Lesher wrote: On Thursday 08 April 2004 22:41, Ryan Thrash wrote: Scenario: a person selects an Auto Attendant option that fires off the Directory application (CVS circa 3/22). Three questions: 1) How do they escape if they didn't mean to go there in the first place (without having to hang up...)? Config of entry into the vertex directory below: If you just wait, Directory will exit if there is no entry. Ah! So it does in fact. Thanks! Many people get impatient and start getting button-happy, often hanging up in frustration. Time to record a new message with instructions for the escape hatch! 2) Why is there a five second pause before the directory instructions start? Probably because you have another extension that begins with 1. Since Asterisk has no other way to know if the extension is complete, it waits DigitTimeout seconds (defaults to 5). And again, you are correct, sir. Internal extensions start at 100. Thanks. Time to re-record the message and assign new extensions for the prompts. 3) Why no option for first name (without recording your own custom message and reversing names in voicemail.conf)? Just wasn't written that way. You're welcome to submit a patch to add first name matching on the bugtracker (bugs.digium.com). Just signed up on Mantis today. Not being a coder, I'll see if I can poke around and get something to work with some help of some local friends that do have a clue. Again, thanks for your helpful response. : ) rt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restart Asterisk
You should be able to do a reload, not having to restart (and bringing the system down). On Apr 8, 2004, at 8:48 AM, Jain, Sonal wrote: Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Caller ID and Voicemail ?s
Wow... talk about a detailed response; thanks! In our situation, we've got a T-1 voice PRI from Allegiance Telcom. For the benefit of those of us who aren't as in the know as you are (and who have no affiliation with a CLEC), is there a way to be able to control what gets sent out as our name portion of the Caller ID (even if it means changing what's recorded at Allegiance)? We somehow manage to do so with the number part. In other words, type real slow and mention specific conf files if possible. This is pretty new stuff for me... Thanks again! -- Ryan On Apr 6, 2004, at 7:59 PM, Kyle Thomas wrote: SCP=Service control point (database that houses name to number) SCP DIP = Query to an SCP via the SS7 network ISUP = SS7 signaling for call setup and teardown (equivalent of invite,ringing,ok,bye) IAM = Initial address message (equal to the SIP invite ) LNP= Local number portability (uses the SS7 network as a backbone). This let's people keep thier phone number and switch service providers. There is nothing quick about quick caller id. The far end Telco will override the name infomration sent to the PSTN and perform thier dips regardless, overwriting the info you are trying sending out. We are a CLEC so, therefore we store, therefore it works.. On Tue, 6 Apr 2004, Andrew Kohlsmith wrote: The terminating telco is doing an SCP dip to thier local SCP's and the database probably does not have that name mapped to this number. First thing to do is make sure the generic name ISUP optional paramter is set in the outgoing IAM / ISDN setup from your GW. You could also store with an SS7 provider , if these are ported numbers you are sending out make sure that the CNAM field in the LNP line record is set to the point code alias of the provider you are storing with. The terminating switch will first do an LNP dip to see what CNAM alias to launch the CNAM dip to. If that is not found , will default to the local SCP thus not finding your record. Ok, and now for the rest of us... SCP? SCP dip? ISUP? IAM? LNP? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL CDR
Remove the semi-colon in front of [global] HTH, Ryan Thrash On Apr 7, 2004, at 8:15 PM, Jeremy Bogan wrote: Sounds like an error in your config file. Want to paste the contents in? Thanks... Sorry: ;[global] ;hostname=localhost dbname=asterisk password=password user=asterisk ;port=3306 sock=/tmp/mysql.sock userfield=1 -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] softphone (SIP) with multiple profiles
I think Xten is included in Lindows, but I could be totally wrong. Probably tweaked to work out of the box with that Distro... Xten is really pretty good. The one I know of is X-Pro/X-Lite from http://www.xten.com/ I doubt that there is a Linux version available... Markus I contacted X-Ten and they told me they are working on a Linux version of X-Lite... let's see... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quick Caller ID and Voicemail ?s
I'm trying to config a couple of things on Asterisk CVS-03/22/04-16:41:51. The number shows up, but I can't get the words to show on a local bell line. The text always comes up as unavailable. In sip.conf for each extension, I've tried: callerid=VERTEX 2142618000 callerid=VERTEX 2142618000 Neither one works. Suggestions? On the Voicemail front, I'm trying to set an extension to dial into a specific voicemail box that has (or in fact may have) nothing with the exact extension I'm dialing. For example, from extension 122, I want it to check 107 without having to enter the voicemail box. In extensions.conf, I've got the following for that extension: exten = 122,1,Macro(extVertex,SIP/122) exten = 122,2,VoiceMailMain(107) In watching asterisk -rc, I see the following when dialing the voicemail extension: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/indications.conf': -- Executing Answer(SIP/122-7161, ) in new stack -- Executing Wait(SIP/122-7161, 1) in new stack -- Executing VoiceMailMain(SIP/122-7161, 2142618000) in new stack -- Playing 'vm-login' (language 'en') Ideas? Thanks, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redhat 9 OVER, Fidora Support, comments please.
We had an issues with an Intel Zero Channel hardware RAID controller that wouldn't allow us to install either Fedora Core 1 or 2, so we couldn't test with *. Given that we didn't try to convert our 9 to Fedora, either. We got it running great under RH 9. HTH, Ryan Thrash On Apr 5, 2004, at 7:50 AM, James Gardiner wrote: Hi *ers, I recently got an Email from Redhat about the dropping of support for Redhat 9 on the 30 of April and that Fedora Project is the recommended future, otherwise, RedHat enterprise ($$$). Considering this, I would like some feed back on the Fedora Project from users who may be using it, and how its going with Asterisk? Are there any problems? Is the Asterisk development team got Fedora Project in mind and fully supported? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto connect to voicemail
I'm running into a similar situation. We have 3-digit extensions and a 4-digit DID numbers that get used for for outbound CID. Therefore, no $CALLERIDNUM direct access to voicemail. Suggestions? What do you do when $CALLERIDNUM of the caller isnt the 4-digit extension? I set all of my users Caller ID entries to their 10-digit phone # so that Caller ID appears correctly when I send their call out the PRI to the public network. The side effect of this is breaking convenient access to voicemail using this method, and I havent found a way to fix it yet. I think this is what you are looking for Exten = 1000,1,Answer,1 Exten = 1000,2,Wait,1 Exten = 1000,3,Voicemailmain([EMAIL PROTECTED]) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Options
How do I set configure my voicemail notification so that when I'm left a voicemail message it: 1) sends an e-mail to my inbox with the voicemail message attached 2) sends a message to my cellphone without the message attached I get notifications when I've got attachments turned off, but my cell doesn't like attachments in the messages and doesn't send them. An even better option would be for the * voicemail system function like a Telekol system: 1) * calls your cell after receiving a new message. On our system 888 puts you to voicemail of the extension you're dialing from, 887 prompts for your extension. It would in essence place an outbound call from 888 using your extension to your cell phone. 2) The voicemail system keeps trying every X of minutes until you answer your cell (a separate caller ID would be good for this), or it can time-out after Y attempts. 3) If you don't want to be notified any longer of the new message, answer your cell and press *# (star-pound), and it turns off notification until another new message comes in. 4) The remote notification number to call can be set through the voicemail options prompts. If you were going to be somewhere without cell service, you could change the number over the voicemail line. Thanks, Ryan Thrash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group
How did the launch meeting go? rt On Mar 29, 2004, at 1:36 PM, Steven M. Sokol wrote: The VON show has started off with a number of interesting announcements. First among these is a big announcement from Pingtel that they have created a not-for-profit corporation called SIPFoundry. This new company includes Pingtel (which has recently open sourced their SIPExchange PBX), Vovida and somebody else. snip More on-the-scene reports to come. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group
Actually, ignore that... forgot to take the check the calendar pill this AM. Doh! rt On Mar 30, 2004, at 11:46 AM, Ryan Thrash wrote: How did the launch meeting go? rt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
Does register_globals need to be on to work with this? And if so, any chance that will be turned off in the (hopefully near) future? Thanks, Ryan On Mar 24, 2004, at 9:09 AM, Areski wrote: I just finished an other version, all my apologies, cause I made it for mysql then I ve done the change to support postgresql and forget to re-test again... not really professional at all ;) snip http://www.areski.net/asterisk-stat-v1/about.php Download : http://www.areski.net/asterisk-stat-v1/asterisk-stat-v1_1.tar.gz If you have still some problems, share them with me ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming Fax Call to File
I can't seem to find an answer in the archives covering this (or maybe I just missed it)... Setting up * and hope to accomplish the following: 1) Use 5 of our DID numbers from our PRI for inbound fax reception 2) When * receives a call on one of these lines, it digitizes the incoming fax to a multi-page .tif file (ala eFax.com) rather than transferring it to an analog fax machine. 3) Based on the DID number, e-mail the resulting fax to a specific inbox The end result--when coupled with doing the same for voice mail messages--would be a unified inbox, which we really are hoping to have soon. To accomplish Part 2, do we need a fax board or some such piece of external hardware, or is the processing power of a dual Xeon server coupled with some as-yet-to-be-identified-DSP-esque software capable of translating fax-static into an image? Thanks for any ideas or pointers. -- Best regards, Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
