Re: [asterisk-users] IAX complaints? What are they?
How recent? I tried switching from 1.2 to 1.4 about 4 months ago, and asterisk would stop accepting IAX connections in less than a day and would need to be restarted. This is with about 50 to 100 calls at a time on each box for about 10 or 12 hours a day. Less for the other half. And all IAX calls are being passed on to a far end terminator via SIP. I was going to scrap IAX entirely because it didn't seem to scale well (for non-trunking apps, at least), but many customers need it for various reasons. Daryl On Nov 30, 2007, at 8:52 AM, zoa wrote: IAX had some stability issues in the past, the recent releases have a lot of iax2 fixes and should no longer have those issues. Zoa ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set CALLERID(num) to a specific number only if ${CALLERID(num)} is not an NANP number
Im trying to figure out the base way to check the callerID being sent
to my Asterisk box and use it if it is a valid NANP number, but
replace it with a static NANP number if it is not. (Why? I have a
few carriers that require this, and a few international users - if it
happens to take one of the carriers that require it, I want it to set
a static number that is valid).
I'm playing with IF and REGEX in extensions.conf, but not getting
very far. Has anyone done this and/or know of a doc? I haven't had
any success searching.
At this point, I have a very broken setup of:
Set(CALLERID(num)=${IF(${REGEX(^(?:\([2-9]\d{2}\)\ ?|[2-9]\d{2}(?:
\-?|\ ?))[2-9]\d{2}[- ]?\d{4}$ ${CALLERID(num)})}?${CALLERID
(num)}:staticNumber)
I'm sure I'm pretty far off - and I've been through many permutations
of this so far. Any ideas?
Thanks,
Daryl
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Re: [asterisk-users] AGI answering the channel even though I neverasked it to
On Aug 13, 2007, at 4:37 PM, Martin Smith wrote: See http://www.asterisk.org/doxygen/1.4/ res__agi_8c.html#c631d48f46d51d4b057 b31807baa1f10 The AGI application will answer the channel if it isn't already answered. You probably need to do whatever you want to do in the dialplan, and keep using DeadAGI. Excellent information. That's what I spent an hour or so unsuccessfully looking for ;) Thank you very much. Now I just have to figure out how to do a database lookup without answering the channel, as that seems to indicate that the AGI is going to answer regardless of whether a play progress tones or not from the AGI. Daryl ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI answering the channel even though I never asked it to
I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a DIFFERENT progress tone for that situation. It appears that every since I've done that, Asterisk is answering the channel. I don't have an Answer command in my dialplan or AGI. Is this something that will automagically happen whether I want it to or not? If so, I'm going to have to do some ugly dial plan scripting to make this work. In case it matters, this is a PHP AGI. Thanks, Daryl ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Cianfarani Sent: Tuesday, June 21, 2005 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850 Does anyone know what the reason why Dell servers cause so many problems for the digium hardware? Better question any Dell models that don't have any these problems with the digium hardware? I've got a PowerEdge 1400SC (old, P4 1gHz, upgradable to dual proc) that's been a absolute tank. Got 2 TDM400P's in it and it supports a very small office with mixed SIP and POTS inbound/outbound. Running Debian, of course. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi IP Phones
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Wednesday, June 15, 2005 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] WiFi IP Phones Guys. I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. You are referring to (in the US anyway) certification as intrinsically safe. I don't know either way about phones listed as such, but with the right terminology you might have better liuck searching. voiceverified. | Daryl G. Jurbala -- | Chief Technology Officer | 215.862.1160 x235 (Office) It had to be you! | 215.862.9880 (FAX) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi IP Phones
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, June 16, 2005 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] WiFi IP Phones Ahmm Andrew, are you sure they are steel? It's been a long time since I did any work in this space but we used to install them in plastic not metal.plastic works better with the radio waves. IS does not necessarily mean steel. My Motorola alpha pager, and my Motorola XTS3000 radio are both plastic and IS listed. voiceverified. | Daryl G. Jurbala -- | Chief Technology Officer | 215.862.1160 x235 (Office) It had to be you! | 215.862.9880 (FAX) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Huddleston, Robert Sent: Tuesday, June 14, 2005 3:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Anyone paying over $450 for a T1 is being ripped off... If you are in VA,MD,DC,PA,DE,NJ you can get an integrated VoIP T1 for $300 - $400 and a flat internet t1 for about $400. The integrated VoIP T1 is great because it's handed off as an ethernet - no need for a csu/dsu Ummm...no. Maybe if you are in or very near a city you can, but not everywhere. You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400 and I'll give you the amount I save over the next quarter. NPA-NXX is 215-862. Good luck. voiceverified. | Daryl G. Jurbala -- | Chief Technology Officer | 215.862.1160 x235 (Office) It had to be you! | 215.862.9880 (FAX) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS rating for SOHO asterisk box
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Wednesday, June 01, 2005 5:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] UPS rating for SOHO asterisk box I have many sites that have a 35amp Charger with 2 x 400ah 900CCA deep cycle batteries (10 year warranty), and 1000VA inverters. The combination makes for perfect power and about 2.5 days run time with my network kit whish consists of several Dlink wifi access points, 1 xbox (hacked into a router/firewall) and a vsat system. Total cost for the power kit AUD$1400 all up, and not a single second of downtime in over a year. [...] Yepyou can (somewhat) build your own UPS with peoperly rated equipment. As a matter of fact, most telco installations don't have monolithic UPS's (like you'll see in most larger datacentersyou know..the big box that says Liebert on it), they use racks of batteries with separate charging circuits. Most of the equipment runs directly off of the battery voltage, but you will find places with some inverters as well. Of course, the room is properly designed (spaced, non-combustible racks, fire detection and supression systems, etc.) and, in most jurisdictions they also have to carry one or more operational permits (current Internation Fire Code requires permitting for stationar lead-acid battery systems exceeding 50 gallons liquid capacity). On the flipside, I have seen a ups flare when the transformer overheated and melted the varnish, nasty! I've seen completely unmodified (although not properly maintained) UPSes as large as 5000 Va completely melt down to the point where they destroyed their own chassis, damaged the rack they were sitting in, and activated the clean-agent supression system in the rooms they were in. This was actually a big problem with one of my customersthey hadn't been maintaining their UPSesthe replace battery lights had been lit for months (they had all been purchased at about the same time). Within a span of about 3 months, 4 of them melted down similarly. A quick call to APC revealed that the batteries in these units were rated for about 12 monts less than they had actually been in service, and a simple battery replacement would have prevented the problem (the chassis was rated for something like 3 sets of batteries...whatever the lifespan of the batteries was3 years I believe). So, don't do stupid things with high voltage, like modifying equipment that wasn't meant to be modified, using undersized equipment, failing to properly vent batteries, or storing your contraption on or near combustibles. It's just NOT worth the risk. Take it from someone who's pulled his share of bodies (of both the live and dead varities) out of buildings. I've seen way too many fires started by electrical system or device modifications similar to those described in previous posts. And most people who do things like this just never consider the life safety risk involved until its way too late. I'll get off my soap-box now and get back on topic. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS rating for SOHO asterisk box
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Tuesday, May 31, 2005 5:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box [...] Regarding this, I have done this hack yesterday: - Remove the battery from an existing UPS - Rewire the UPS onto biggest car lead acid battery (12v) you can find. Et voila! Bigger capacity. Put the batteries in parrallel and you do get monstruous UPS capacity... the only trouble with it is that re-charging the batteries may take some time. [...] Congratulationsyou've just given this part-time small town fire marshal and 14-year fire service veteran nightmares. Kidsdo NOT try this at home. The inverters in small UPSes are not designed to deal with runtimes that exceed the batteries in them. If you run this setup well past the time it was designed to run (by adding 3, 4, or more times that battery capacity it was ever designed to have) that chances of a catastrophic inverter failure (meaning flash, boom, fire) are very real and very likely. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Did nufone change allowed codecs?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Thursday, May 05, 2005 7:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Did nufone change allowed codecs? Hi, I've been using nufone DIDs for months with no problem. Now [...] No files will play on a call to asterisk because they aren't found in g729. Perhaps the desired codecs for DID have changed? I know you can specify them when ordering DID, but I see no way to change them once the DID are provisioned. Any help? Just a crazy idea herehave you contacted NuFone support yet? Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1800 DNIS and asterisk (HOW TO?)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, May 05, 2005 6:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 1800 DNIS and asterisk (HOW TO?) Hi I have applied for Qwest 1800 termination to a T1 ISDN PRI. They tell me that I will have to program a predefined DNIS number on my switch. According to them unless asterisk returns that DNIS number no call will get through. How do I program the DNIS, is it through zaptel.conf or some other way. Is it required??. As per qwest if the 8xx # is going to be routing to an ISDN TG, DNIS is required. [...] Huh? The last time I dealt with DNIS (admittedly, years ago) the provider sent the digits to ME via DTMF to tell ME what number was dialed to terminate on that line (you knowDialed Number Identification Service). Unless DNIS has turned into telcoBGP while I haven't been watching, what you're being asked to do doesn't seem quite right. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on VMWare ESX/blade servers
Has anyone had any experience (good or bad) running Asterisk under VMWare ESX server on a blade chassis? This application will (fairly obviously) not include Zap channelsactually, it will be SIP-only. Please feel free to contact me off-list and I'll summarize for the list later. Daryl G. Jurbala NGM Tec, Inc. Tel: 215-862-1160 ext. 235 Fax: 215-862-9880 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] small qos switch
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Sunday, March 27, 2005 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] small qos switch I heard a great solution at Linux World Boston. A rather talented young man mentioned using a IPV6 VPN on the IPV4 internet. IPV6 supports QOS by default. Just VPN straight back to the CO and have your POP there so you only need one firewall too. He may have been talented, just not in network engineering. While your IPv6 encapsulated VPN would have QOS, the underlying transport medium (IPv4) still would not (if it didn't have it before). Furthermore, if any Ipv4 hops in between would have prioritized your traffic higher based on its type, they now have no idea what is is, because it's encapsulated. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How NuFone.Net's customer service works.
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Linn Boyd Sent: Monday, March 14, 2005 6:05 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How NuFone.Net's customer service works. Hello All, I have been using asterisk for some time, and I would like for all to take a look at what NuFone does when they get [...] I hope that people that care about customer service avoid NuFone.net [...] Hmmm...I've had 2 problem with my NuFone service in the year or more I've used them. Each time I've treated them professionally when reporting the issue and received the same treatment in return. The issues were also resolved promptly. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which hardware for this solution?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Mandolfo Sent: Wednesday, March 09, 2005 8:59 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which hardware for this solution? Hello, we are a firm who wants to develop some VOIP solutions. [...] Straight to the point: what kind of hardware I need? I saw some PCI cards (like Digium Wildcard TE110P) but I am not sure what to buy. You need to but the appropriate cards to interface with the PBX you are trying to connect to. Without knowing what interfaces it has available, that's a difficult question to answer. If it's got an E1 or T1 interface, buy an appropriate port-density T1/E1 card (surprise) like a TE110P or TE410/405P. If it's analog, and appropritaely-configured TDM400P would be the way to go. Cards are cardsget what you need to make the interface happen. It's like asking what card you need to connect your computer to some undescribed network. If the network is ethernet, you need an ethernet card. If it's token ring, you need a token ring card, etc. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detect sound and continue, like BackgroundDetect() for voice
I'm looking for an application that can monitor a channel for voice input and then proceed on. The closest thing I've found is BackgroundDetect, which expects DTMF. Here's what I'm doing: -Call file generated which calls someone and connects them to an extension. -Extension plays stuff, etc. etc. etc (not important) With digital or VoIP termination, this works fine, because * knows when the line is answered. On analog POTS, it has no idea when the call is actually answered, only when its dialed, so the playback starts right after the line is dialed, not after the called party picks up. The Dialogic IVR SDK monitors call termination status this way, so I'm looking for something similar in *. Anyone have any ideas on this one? Or am I going about this the hard way and missing an obvious alternative? Thanks, Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Detect sound and continue, like BackgroundDetect() for voice
Yes, I'm replying to my own post. Roger Gulbranson suggested this: http://www.voip-info.org/tiki-index.php?page=NVBackgroundDetect As he's using it for FAX detect, and it has a talk option as well. If anyone is interested, I'll report back with my results. Thanks Roger! Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Detect sound and continue, like BackgroundDetect() for voice
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Thursday, March 03, 2005 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Detect sound and continue,like BackgroundDetect() for voice Yes, you missed an obvious option - search the mailing list. This has come up an number of times. I searched both the Wiki, and the list. But I obviously didn't come up with the right search terms, or overlooked relevant results, which is why I asked. How about this: the next time I have a question to ask, I'll call you first to ask for what search terms to use before posting. What's your mobile number? Thanks, Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:Terrible inbound call quality vs. outbound
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gene Willingham Sent: Tuesday, February 01, 2005 6:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE:Terrible inbound call quality vs. outbound I am experiencing the same problem, except I do not use Voicepulse outbound. I have 100 Mbps connection, so it should not be a bandwidth issue. Last Thursday they had a 4 hour outage on inbound calls. The call quality has deteriorated since. I am in the process of looking for another provider. [...] Not to just me too, butme too. I've contacted their support on numerous occasions, and have been given busywork to do (run ping plotter for 24 hours, send us the results, etc) and never receive a response that acknowledges a problem of any sort. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Tuesday, February 01, 2005 11:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit have a look at http://www.voip-info.org/wiki-Asterisk+zap+channels Here it tells you that you can specify a wait period. [...] Don't know if it will apply to those having issues with BRI/PRI, but in my case, a ww in front of the dial string has worked witout fail for the last few days. Thanks to all who helped, Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!
PLEASE CONFIGURE YOUR AUTORESPONSERS TO NOT SEND MESSAGES TO PEOPLE WHO POSTIN MAILING LISTS YOU SUBSCRIBE TO. This is an extremely rude thing to allow, and is becoming increasingly common, especially with users of the Asterisk-Users list. Daryl From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, February 04, 2005 6:41 PMTo: Daryl G. JurbalaSubject: AUTOREPLY RE: [Asterisk-Users] Zap channel occasionally misses di... Vielen Dank für Ihre Email! Ich bin vom 02.02.05 bis einschließlich 13.02.05 ausser Haus. Ihre Email wird bis dahin nicht bearbeitet oder weitergeleitet. Bei dringenden Fragen wenden Sie sich bitte an meinen Kollegen Herrn Rüdiger Hoog Email: [EMAIL PROTECTED] Telefon: 02331/473101-11Telefax: 02331/473101-19 Für weitere Fragen stehe ich Ihnen gerne zur Verfügung und verbleibe mit freundlichem Gruß, Stefan SpeckenheuerTechnische Leitung POS Service, Logistik Handels GmbHAuf dem Graskamp 2D-58099 Hagen Tel. +49 2331 473101-21FAX +49 2331 473101-39 mailto:[EMAIL PROTECTED]http://www.posservice.de Sitz der Gesellschaft: Walter-Rathenau-Ring 9-11, 59581 Warstein BeleckeHandelsregister Arnsberg: HRB 2958Ust.IdNr.: DE 198 933 818Geschäftsführer: Martin Menzel, Christian Woelke Diese E-Mail einschließlich aller Anhänge ist vertraulich.Wir bitten, eine fehlgeleitete Mail unverzüglich vollständigzu löschen und uns eine Nachricht zukommen zu lassen.Wir haben die Mail beim Ausgang auf Viren geprüft;wir raten jedoch, auf Grund der Gefahr auf denÜbertragungswegen, zu einer Eingangskontrolle.Eine Haftung für Virenfreiheit schließen wir aus. This e-mail and any attachments are confidential.If you are not the intended recipient of this e-mail,please immediately delete its contents and notify us.This e-mail was checked for virus contamination beforebeing sent; nevertheless, it is advisable to checkfor any contamination occuring during transmission.We cannot accept any liability for virus contamination. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk > Sent: Tuesday, February 01, 2005 11:26 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Zap channel occasionally misses > dialing thefirst digit > > have a look at http://www.voip-info.org/wiki-Asterisk+zap+channels > > Here it tells you that you can specify a wait period. [...] Don't know if it will apply to those having issues with BRI/PRI, but in my case, a ww in front of the dial string has worked witout fail for the last few days. Thanks to all who helped, Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap channel occasionally misses dialing the first digit
I THINK. When dialing 1+10 digits, I occasionally get a telco message You must first dial a 1. When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not) randomly. Does anyone know what this might be and/or an easy way to have the ZAP channel come off-hook, delay for 1/2 second or so, and then dial? Thanks, Daryl G. Jurbala NGM Tec, Inc. Tel: 215-862-1160 ext. 235 Fax: 215-862-9880 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, January 03, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards [...] For business use, I would suggest you first find a BRI card you can use here in the states. Hint, bug Kapejod into making that 4 port card US ready. Then move any business user over [...] That might work out where you do your deployments. In Verizon territory, you can get analog business lines with unlimited long distance and no metered minutes for about $37 a month. A BRI costs you about double that for the loop, with metered minutes and bring your own LD. Past the technology aspects, BRI just doesn't work here. And I'm going to guess that pricing structure is similar in other areas as well. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qs about FXO/FXS cards
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, January 01, 2005 9:12 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Qs about FXO/FXS cards [...] I'm running older, but solid hardware and not seeing any issues. I'm using a Compaq Proliant 1850R Gen1 dual PII 400 with 512MB ram, GB ethernet, and SATA Hardware RAID. Cheap, efficient, redundant. And for a Debian box, good enough. [...] I just have to add my $0.02 here. I've got a PIII-550 Proliant 800 that NEVER has any issues like this. It's running Debian woody, and has a TDM400P that never has any of these issues. It's also running 208v from a high quality UPS. As a telephone system should, it simply works. It is forgotten about, and used andused and used. No one has to do much of anything to it, and no one has to make excuses for it (sorry..it's VoIP). Anyone who wants to run junk hardware and beta code pretty much loses their right to complain about the results of doing so. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is H323 dying?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin List-Petersen Sent: Thursday, November 18, 2004 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is H323 dying? [...] That is correct. H.323 is something nobody real will deal with, but it's still supported because a lot of the old fashioned carriers do H.323. [...] Nobody real deals with it and it's supported by old fashioned carriers? Please, don't thak this as an insult, but you need to qualify that your background obviously doesn't include any carrier-class bulk VoIP termination whatsoever when you make broad statement like that. Millions and milions of minutes of voice and fax traffic each day are carried over h.323, for end users that don't even know they are using VoIP, and in most cases don't even know what VoIP is. Minutes handled by bold old and new companies. Now if you wanted to say that it's not in vogue for soft PBXen and key systems to support h.323, I'll buy that. But I'm going to guess that voice traffic over SIP is a mere fraction of voice traffic over h.323 on any given day. Daryl Jurbala ___ 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] LineJack + Asterisk HELP!
-Original Message- From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 16, 2003 11:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] LineJack + Asterisk HELP! To install driver for LineJack I need kernel source. I have debian, and I installed from apt-get install kernel-source.2.4.20 but while it make ./configure it still asks me for the kernel source. What can be wrong ? [...] Debian kernel sources are tared and compressed. You should see a kernel-source.2.4.20.tar.bz2 or similar in /usr/src. Un bzip2 it, untar it, and make a /usr/src/linux symbolic link to the directory it unpacks to. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog FXO Card
-Original Message- From: nathan [mailto:[EMAIL PROTECTED] Sent: Monday, September 15, 2003 10:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Analog FXO Card These cards are replicas of the X100P sold for use in an Asterisk ( www.asterisk.org) phone system. They are fully functional and work with the same wcfxo driver as the actual X100Ps. So is someone pirating digium hardware? Interesting that it has 2 ports on it, and a speaker. The picture looks a whole lot like a modem to me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to connect 2 TE410P
-Original Message- From: Kelvin Chua [mailto:[EMAIL PROTECTED] Sent: Monday, September 08, 2003 1:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] how to connect 2 TE410P neat! actually we are just in the process of planning for an asterisk based simulation lab for the university. do you have a cable pin-out descriptions for that purpose? thanks! A T1 cross-over cable is: 1-4 2-5 3-3 4-1 5-2 6-6 7-7 8-8 Daryl G. Jurbala BMPC Network Operations Tel: +1 215 825 8401 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Maximum number of X100P cards in the same * box
-Original Message- From: Dan [mailto:[EMAIL PROTECTED] Sent: Monday, September 08, 2003 6:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Maximum number of X100P cards in the same * box Hi, max we've got running happily without issues is 4 x X100P and 3 X100P + TDM40B in another install... we didn't get 4 X100P + TDM40B running, but that could have been an issue with our install at that time i have a feeling I could make it work :) I need to run JUST the X100P cards (as many as possible). There is no need for FXS cards. There is no E1 available, just pure PSTN analog linesI know that is a lot cheaper to get an E1, but it is not possible in those circumstances... Or there is any cheaper option to get 18 FXO It sounds like a channel bank and an E1 or T1 card in your * box would be a better way to go. (analog lines to the channel bank, which muxes them to an E1 or T1). That way you have have fewer cards in your * box, fewer interrrupts to poll, fewer things to go wrong. Channel banks are arguable much more reliable than PCs and PC hardware. Just a thought...might not be possible depending on your setup. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to connect 2 TE410P
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, September 07, 2003 8:15 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] how to connect 2 TE410P [...] do you break any standards if you forgo 3, 6, 7 8 ? i've never bothered patching those in, and it hasn't hurt so far, but maybe i'm missing something (which is oft the case) Since a T1 is only two pair form the telco, no. But properly made cables ought to be fully pinned, for strength if no other reason, and that's the standard way to do it. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Maximum number of X100P cards in the same * box
-Original Message- From: Dan [mailto:[EMAIL PROTECTED] Sent: Monday, September 08, 2003 11:20 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Maximum number of X100P cards in the same * box [...] The problem is that I want to have the full functionality of an X100P card. It is possible with a CB? Define full functionality. I'm not aware of ANY advantages (other than cost in low density installations) for using FXO cards. Maybe you have a functionality requirement that I'm not aware of, and is out of the ordinary. Daryl G. Jurbala BMPC Network Operations Tel: +1 215 825 8401 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940/7960 ethernet ports
-Original Message- From: Travis Johnson [mailto:[EMAIL PROTECTED] Sent: Monday, September 08, 2003 1:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7940/7960 ethernet ports [...] We are having a problem with Cisco 7940 and 7960 phones when the PC is plugged into the 2nd ethernet port on the phone. It will drop the PC's connection for about 30 seconds, then bring the connection back up for about 30 seconds. It does this continually regardless of how the port is configured on the phone. [...] Sounds like failing autonegotiation. Lock it down to a speed an duplex on the phone as well as on the device attached to the phone. Daryl G. Jurbala Introspect.net Consulting Tel: +1 215 825 8401 Fax: +1 508 526 8500 http://www.introspect.net PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VONAGE or IP Dialtone
-Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Saturday, September 06, 2003 12:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VONAGE or IP Dialtone Not necessarily preposterous; I would certainly allow that its optimality is arguable. [several very good point deleted] Thank you. Well stated, and you saved me the typing ;) Find me SIP termination with unlimited minutes at a reasonable flat rate to US destinations that works natively with * and I'll dump Vonage tomorrow (and deal with the rest). Seriouslyplease? Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NuFone.net Was:VONAGE or IP Dialtone
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, September 06, 2003 8:39 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone Thanks for the great feedback on these options. I am fairly new at this and not familiar with the IAX/IAX2 capabilities offered by Nufone. Could you expand on this and is Nufone inbound and outbound from the PSTN? Personal and recent NuFon.net experience: The are a great group of people, who resond to email very quickly. You tell them you want termination and a DID for an * box, and they'll get it set up and send you configuration snippets to get it working. They are still working on their billing system, so they just email you invoices, etc. at this point, but they tell me they are going to have a new system done soon where you can check your balance online, etc. At the moment, I believe they have DIDs in Michigan (and 800) only. I am told they are working on agreements for more locations. My vonage # forwarded to the Michigan DID seems to work just fine ;). Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbee Question
Modems are FXO devices (designed to connect to a telephone line from the phone company). You need an FXS device, like a Digium card or an Internet PhoneJack to attach your standard analog phone to. Other than than, yes, Asterisk will do exactly what you are asking. You just need the right hardware and you're all set. Daryl G. Jurbala Introspect.net Consulting Tel: +1 215 825 8401 Fax: +1 508 526 8500 http://www.introspect.net PGP Key: http://www.introspect.net/pgp -Original Message- From: Hemant Kumar [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 3:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbee Question I am new to this list, and I searched to find the answer to my question, but could not find it. Can I do the following using Asterisk ... Load Asterisk on a PC running linux. Logon to VoIP service like http://www.freeworldialup.com/ to using your ethernet. Asterisk, routes the call from the PC to a regular phone connected to through the modem. When I am receiving the VOIP call, I hear the ring on my phone connected to the modem. Has anybody done it? Can you please direct me to the instructions for the same. Thanks, Hemant ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pointer to upgrade 7960sip beyond v3.2.0?
You have to go to Settings- #9 Unlock Config in v4+ firmware. The unlock password no longer works from just anywhere. Daryl -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 9:58 PM To: Asterisk-users-list Subject: [Asterisk-Users] Pointer to upgrade 7960sip beyond v3.2.0? Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't unlock it to do any configs. Any thoughts? Rich [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - setup help
Considering www.asteriskpbx.org doesn't mention zapata either, care to enlighten us at to what it does? I've got zapter, libpri, and asterisk compiled and running. While I haven't had the change to play with ALL of the features, all seems to be working fine with my setup. Daryl -Original Message- From: Scott Stingel [mailto:[EMAIL PROTECTED] Sent: Sunday, August 31, 2003 11:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Newbie - setup help I was looking at that setup guide you mention, and it's well written but does seem slightly out of date. For example, it mentions that there are 3 directories in which you have to do a make clean; make install. There are actually 4: the zapata directory is also necessary. Don't know if that would be your problem? Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Sent: Sunday, August 31, 2003 3:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie - setup help Hi, I'm trying to setup Asterisk on a Linux (redhat 8) machine. There are no analogue phones to be used use is purely for internet traffic (SIPs). I've followed the setup guide from : http://www.automated.it/guidetoasterisk.htm But cannot get asterisk to run. If I type asterisk at the command prompt I get invalid instruction If I type ./asterisk start I get [FAILED] Can anybody recommend as site with setup instructions for a novice ? Thanks Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - setup help
Actually, that's incorrect. The code in zapata has long since been incorporated into other code, so zapata is no longer necessary. In that case you can cancel my last question. ;) Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installation Problem
* cd /usr/src/linux (you did unarchive the sources, and ln -s /usr/src/whatever Debian called it /usr/src/linux, right?) * make config, hit enter through the whole thing * make dep Go about compiling *. modversions.h is generated by make dep. Daryl -Original Message- From: Phillip Britt [mailto:[EMAIL PROTECTED] Sent: Saturday, August 30, 2003 11:09 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Installation Problem Thanks for the info. I am running Debian. Do you know if l would need to install my own Kernel on that distribution. I have checked and it looks like the kernel sources are installed. Any other suggestions? Thanks, Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, August 30, 2003 8:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Installation Problem Depending on your distribution, you will either need to install your own kernel, or install the kernel source. it should come as a package for your distro. On Sat, Aug 30, 2003 at 08:23:08PM +1000, Phillip Britt wrote: Hi, I am quite new to Asterisk and Linux in general. When l try to install the Zaptel component, l get the following error: asterisk:/usr/src/zaptel# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTANDALONE_ZAPATA -c zaptel.c In file included from zaptel.c:36: /usr/include/linux/module.h:21: linux/modversions.h: No such file or directory make: *** [zaptel.o] Error 1 asterisk:/usr/src/zaptel# Can anyone point me in the right direction. Cheers, Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: SPAMWARNING:216.207.245.21:RE: [Asterisk-Users] Re: LAN switches with PoE? PoE phones?
-Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Monday, August 18, 2003 6:03 PM To: [EMAIL PROTECTED] Subject: SPAMWARNING:216.207.245.21:RE: [Asterisk-Users] Re: LAN switches with PoE? PoE phones? [...] Who does network punchdowns on a 66 block. You do them on a patch panel and they usually have nice plastic guides that keep your fingers away from the terminals. [...] Otherwise known as a 110 block. Daryl G. Jurbala Introspect.net Consulting Tel: +1 215 825 8401 Fax: +1 508 526 8500 http://www.introspect.net PGP Key and Adobe Digital Signature: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
