Re: [asterisk-users] Congestion/Forbidden issue with new carrier
Kev,
Change the HOST from "sip.nsw.iinet.net.au" to "203.215.3.1"
Also, see:
http://forums.whirlpool.net.au/index.cfm?a=wiki&tag=iiNetPhone_asterisk
Sincerely,
Trevor Hammonds
-Original Message-
From: Kev S
Sent: Thursday, January 10, 2008 9:28 PM
> > > S N I P < < <
Trunk Info
[trunk_1]
disallow =
allow = all
callerid = 028012
contact =
context = DID_trunk_1
dialformat = ${EXTEN:1}
fromdomain = iinetphone.iinet.net.au
fromuser = 028012
group =
hasexten = no
hasiax = no
hassip = yes
host = sip.nsw.iinet.net.au
insecure = very
port = 5060
provider =
registeriax = no
registersip = yes
secret =
trunkname = Custom - iinet
trunkstyle = customvoip
username = 028012
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Re: [asterisk-users] Change Default Voicemail Message
Daniel, You could have Alison record a prompt "Welcome to (nursing home)" and re-record the prompt "The person at extension..." to be "The person in room...". Then have your dialplan play the "Welcome To..." message before sending the call to voice mail. Then callers will hear "Welcome to (Nursing Home). The person in room 5 is unavailable. Please leave your message..." and if the resident has a recorded personal greeting or name, it would replace the "The person..." portion with either the resident's recorded name or greeting. Sincerely, Trevor Hammonds From: Daniel Cole Sent: Sunday, January 06, 2008 6:15 PM Hello List, I have a client (a nursing home) that we are looking at installing a trixbox for. One of the features that they would really like is a customized, standard voicemail recording for each of the residents rooms. We are looking for something along the lines of a voicemail recording like this: "Welcome to (nursing home). You have reached room 5. Please leave a message after the tone". What would be the easiest way to get this to work. I have had a look at a few options, but I cant seem to find what I am after. Any help would be much appreciated. Thank You, Daniel Cole ___ --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] New feature: calling all bug marshals
Philip Prindeville wrote on Tuesday, 04 December 2007 at 11:58 PM: >Steve Edwards wrote: >> On Tue, 4 Dec 2007, Philip Prindeville wrote: >> >> >>> I wanted to write a "popcorn" app for myself, both to learn how to script in >>> >> >> Just out of curiosity, what does this have to do with popcorn? >> >> Thanks in advance, >> >> Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST >> Newline Fax: +1-760-731-3000 >> > >You used to be able to dial "popcorn" (767-2676) in any area code (at >least prior to 1982) and get the current time. > >-Philip Actually, this was specific to Northern California (767 prefix). In Southern California, the Time Announcement service has always been in the 853 prefix. The "official" numbers were 767-1212 and 853-1212, respectively -- though the entire prefix in all area codes of the respective halves of the state were reserved for, and rang to, the Time Announcement service. As of 19th September 2007, AT&T discontinued the service due to the unavailability of parts for the 1960s-era Audichron equipment, and declining use of the service. That being said, I would love to have this ability in Asterisk. Perhaps someone has even preserved Jane Barbe's original recordings in a way that they can be recorded for Asterisk. That would be a kick. Sincerely, Trevor Hammonds ___ --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] Patent issues, what features we can't use?
From: Zeeshan Zakaria Sent: Thursday, August 16, 2007 4:20 PM >Can you write it clearly what exactly you mean here. Who is J2, and do they hold the patent for fax-to-email? Google is your friend http://www.google.com/search?q=j2+patents Sincerely, Trevor Hammonds ___ --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@lists.digium.com
Bill, I am not aware of any commercial Asterisk-compatible cards that support North American BRIs right out of the box. The best I have been able to come up with was a card sold on eBay, where the seller promises to supply a patch that needs to be applied to Asterisk (based on BRIstuff) so that it will support North American BRIs. The driver allows only one SPID per BRI, so multiple DID/MSNs are not supported. Fortunately, PRIs are relatively cheap in California. As such, I have not yet made a concerted effort to find a card that does all that I need over BRI -- though I am really interested in having this capability. I wish the Digium BRI card had the drivers for North American ISDN. Such a shame that they went to the effort of getting FCC approval, but didn't bother to do the work to actually make it work in the US. As for ordering BRIs from AT&T: In California, there is an ISDN group that handles PRI/BRI orders. They are available at 1-800-4PB-ISDN. I believe the correct number for Michigan is 1-800-552-8647. If that is incorrect, or they do not know about BRIs, call the California number and ask if they have the corresponding number for Michigan. Hope this was of some assistance. Sincerely, Trevor Hammonds -Original Message- From: [EMAIL PROTECTED] Sent: Friday, August 10, 2007 11:27 AM Hello everyone, I'm hoping someone can help me with this. I have a business customer in the U.S. (Michigan, AT&T Territory). I need to get 4 trunks into an asterisk Box. My intention is to use an Eicon Diva Server card with 2 BRI Circuits. The reason for this is that the business needs DID's on the trunks (20 of them). A full or fractional PRI is overboard for them, as they will never need more than four channels. I also don't really want to go with any kind of analog trunk for other reasons (Disconnect supervision, potential echo problems, etc..) I've called AT&T About a dozen times now, and no-one can tell me who to call in order to get a BRI. Is anyone familiar with how to order a BRI from AT&T (Or a CLEC that covers Michigan)? I need 2 BRI Circuits (4 B Channels) with 20 DIDs accross the 4 circuits. Also, can anyone confirm that an Eicon Diva Server BRI Card will in fact support North American BRI? Are there any less expensive cards that will handle North American BRI? Any help highly appreciated Bill ___ --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] Teliax Quality of Service
From: SIP Sent: Saturday, August 04, 2007 2:57 PM >Stephen Bosch wrote: >> Douglas Garstang wrote: >> >>> I confused by this. Don't ITSP's have redundancy? Don't they have >>> multiple edge systems for accepting incoming calls? Don't their multiple >>> edge systems have multiple interfaces, connected to multiple subnets, >>> via multiple switches? And, don't they have multiple upstream providers? >>> About the only thing that could go wrong that affects all service like >>> this would be a badly pushed out software update, affecting all systems? >> >> >> Don't be confused. The answer to most of your questions is no. >> >> Barriers to entry are too small for ITSPs, and there are lots of >> basement operations masquerading as big carriers. >> >> -Stephen- >> >> > > There are also lots of big carriers masquerading as big carriers. ;) > > > If the ONLY people who could get into the business were the ones who > could, before offering any services to customers, afford to build out > multiple edge systems for accepting incoming calls, each with multiple > interfaces connected to multiple subnets via multiple switches using > multiple upstream providers, you would have ONE single choice for an ITSP. > > And AT&T doesn't have that amount of redundancy in their network. > Working in the carrier networking business, I can assure you that we've > NEVER run across a SINGLE carrier network (not from the largest to the > smallest) that has redundancy in ALL aspects (or even MOST aspects) of > its network. This is why there are uptime policies that allow a > percentage of outages to occur. Triple 9 uptime (Exceedingly rare, but a > purported goal -- 99.999%) still allows 15 full hours of downtime a > year. And that rarely includes the occasional lost packet or latency. Your math is incorrect. FIVE nines (99.999) allows only 5.26 MINUTES of annual downtime. Triple nine (99.9%) allows for 8.76 hours of annual downtime. Keep in mind that most SLAs do not include "planned" maintenance in their guaranteed uptime. > Face it. If you want service that never goes down, you're either able to > pay the hundreds of millions to provide your own networks and build out > your own redundancy, or you're stuck in the same boat with the rest of > us -- be it that you choose a gigantic carrier or a mom 'n' pop ITSP. > > N. h Trevor Hammonds ___ --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] Queue announcing hold sequence instead of hold time
From: Drew Gibson
>
> Hi,
>
> We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian
> Sarge) and the behaviour of our Call Centre queues has changed
> slightly.
> Before the upgrade, when a caller was waiting in the queue, the
> estimated hold time was announced as expected ("estimated hold time is
> less than 2 minutes ...").
> Now the caller gets an announcement of their sequence in the queue
> ("Your call is now first in line ...").
> I believe that the only changes I have made to queues.conf and
> agents.conf is the addition of the "context=" statement and editing the
> list of agents.
>
> Has anyone else seen this? What am I missing?
>
> regards,
>
> Drew
Drew,
This has been normal behaviour for as long as I can remember. The caller
hears the estimated time until they are next in line, then they hear the
'next in line' announcement.
Sincerely,
Trevor Hammonds
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[asterisk-users] New Linksys SPA Daylight Saving Time Rule for US/Canada
To work with the latest change to the US/Canadian DST, I made a new Daylight Saving Time Rule for my Linksys SPA-9XX phones. start=3/7/7/02:00:00;end=11/1/7/02:00:00;save=1 As I could see no way to tell the phones to begin DST on the second Sunday in March, I assumed that the second Sunday would always be at least on or after the 7th of the month. Let me know if you see any obvious flaws to my logic, or the rule itself. Thanks. Sincerely, Trevor Hammonds ___ --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: Setting Caller-ID / Point Codes
Matt, A Letter of Agency is almost always signed by the end subscriber and given to the ILEC/CLEC. Its purpose is to allow someone other than the subscriber (e.g. an "Enhanced Service Provider" or consultant) to make changes to, or get information about, the customer's account (e.g. your account with the CLEC). I am not sure how you are set up. Are you a subscriber of the CLEC, or does your ITSP get the DIDs from the CLEC? I would be interested in knowing more about the company you are using for the CNAM stuff. If you feel you should not disclose the information publicly, please e-mail me off list. Sincerely, Trevor Hammonds From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, February 21, 2007 4:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discussion Subject: [asterisk-users] Re: Setting Caller-ID / Point Codes Bump. Nothing heard. On 2/19/07, Matt <[EMAIL PROTECTED]> wrote: Greetings folks, I'm currently dealing with a company to let me set Caller-ID-Name on outbound calls. So far pretty happy with their services. The basic service works like this: * CLEC sets Point Code to point to this company * CLEC has to sign LOA saying they give me permission to set the Caller-ID-Name through this company. * I go into web interface and set name. However, the CLEC is currently asking questions about the LOA, and I am concerned they may not sign it. What do other people here know about this procedure. Have any of you signed up with a company to allow you to set the Caller-ID-Name? If so, was an LOA required? Did your CLEC sign it? Who do you all work with? ___ --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] Disconnection supervision: what about PBX
From: Stephen Bosch > Hi, Trevor: > > Trevor G. Hammonds wrote: > >> Stephen Bosch wrote: > >>> Are BRI circuits what phone companies call "digital" lines for use > >>> with digital sets, such as with digital Centrex? > >>> > >>> I'm not aware that Telus even offers BRI. > >>> > >>> Sorry -- BRI is ISDN, not digital Centrex. > >>> > >>> I'm still not aware that Telus even offers ISDN anymore :) > >> ...and by that I mean ISDN BRI ;) > >> > >> -Stephen- > > > > Throughout most of the United States, "Digital Centrex" or > "CentrexIS" is > > ISDN as part of a Centrex group. If the circuit is meant for a > single > > device, it would be a BRI. If the circuit is "Hi-Cap" or meant to be > hooked > > up to a PBX or the like, it would be a PRI. > > > > I am not that familiar with Telus, but what Bell is calling "Digital > Voice" > > service is merely VoIP over one of their DSL connections. While I > know that > > both companies offer Centrex over PRI, I am unsure if either company > > supports BRI widely anymore. I know BRI service is available, and > most of > > their switches are capable of offering BRI circuits. For example, > digital > > secretarial enhanced key telephone sets are ISDN phones that work via > a BRI. > > > > In my experience, most telcos in the US and Canada will not tell you > about > > BRI unless you specifically ask. And if you do, they shuffle you off > to > > another department where they may or may not know how to properly > provision > > the circuit. Somehow, all the LECs in North American look at BRI as > a > > data-only service and never really saw the advantages of offering it > to > > voice-only customers. As such, now that 128k (or 144k) is too slow > of a > > data connection for most, BRI has just been passed by. Such a > shame... > > I can still find information pages on BRI on the Telus website (buried, > but there); as you point out, though, they refer to data connections > only. > > I am going to give it a try and see what I come up with. > > There's every possibility they'll offer it but at a ridiculous price, > just to discourage adoption enough to let them phase it out. I'll bet > that this stuff will disappear when the switching equipment is > upgraded. > > -Stephen- Stephen, I often find that the telcos discourage voice BRI adoption by making it hard for you to obtain the correct information or correct department to order the circuits -- not necessarily by making it overly expensive. I can guarantee that most telcos have no immediate interest in discontinuing BRI (Switched 56, perhaps). However, I cannot tell you how many times I have heard a telco employee say, "BRI is for data only." If you can get past the standard business office or residential order center and into the ISDN or complex services group, they will usually be able to help you. In fact, if you have an existing relationship with the telco, the same place where you would order a complex Centrex group or pretty much any T1 would either be able to help you with the BRI order or at least be able to get you to the right place. Take a look at Telus' General Tariff item 485, which covers BRI service: http://about.telus.com/publicpolicy/tariffs/docs2/CRTC180_1/General_2/item48 5.pdf After a brief glance, it looks like Telus charges $91.75 to $107.80 per month (depending on the "Rate Band" of your exchange) for a 2B+D on a one-year contract. On a five-year contract, that drops to $79.85 to $99.80 per month. Without a five-year commitment, this is quite a bit more than I have seen in Southern California (around US$60/month with the voice feature package). However, California has seemed to be one of the least expensive places for ISDN services. Good luck! Sincerely, Trevor Hammonds ___ --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] Problem with busydetect and cell phones
> Ryan McDaniel wrote: > >> I have a very strange problem I'm hoping someone has encountered > > already. > >> I've scoured the internet for an answer to this one. My phone > company > >> provides no disconnect supervision. Hence I'm forced to use the > > busydetect > >> feature. I have a TDM400 with two FXO ports. If I call from an > > internal > >> extension to a landline and then hangup the landline Asterisk > detects > > the > >> busy signal correctly and clears the line. If I call from an > internal > >> extension to a cell phone and then hangup the cell phone Asterisk > will > >> never > >> detect the busy signal though it is clearly there. Asterisk will > > happily > >> sit there listening to the busy signal. I suspect that the busy > signal > >> styles are slightly different though it is undetectable to me. How > can > > I > >> fix this??? It causes severe issues when a call is forwarded to a > cell > >> phone via the Zap interfaces as once you hangup the cell phone > Asterisk > >> never releases the channel. > >> > > > > The landlines are with AT&T. The cell phones I'm testing with are > > Cingular (AT&T subsidiary). There must be a subtle difference in the > > busy signals. How can I make it catch busy signals from both > carriers? > > Have you tried calling AT&T and asking for call disconnect supervision? > > I realise that this can be a thankless and tedious endeavour, but it IS > worth trying. There are almost no commercial switches that don't > support > this; it's a matter of activating it for the specific circuit in > software. Particularly if you have a business line -- you can demand > it. > All PBXs need it if they use analog lines (and plenty still do) so I'm > sure this is not an alien concept to AT&T. It's just a matter of > finding > the right Earthling there who can help you. > > This might be one of those times where a "beer with the technician" > will > get you some joy, if calling Repair doesn't give you any joy. > > -Stephen- > > > Unfortunately I tried that. Apparently my lines are on one of the last > really ancient junction boxes in Southern California. When using > busydetect is it looking for any on / off repetitive sound to identify > the busy signal, or for a specific length sound as defined in the > indications.conf region? I'd really like to avoid using callprogress > if > possible. Is there a way to tweak it so it will accept a wider variety > of busy patterns? > > - Ryan Ryan, Even 1AESS switches offer disconnect supervision -- and I am not aware of any of those still in primary service in Southern California. By early 2000, Pacific Bell (then SBC, now AT&T) replaced all the analogue 1As with DMS-100s. If you care to contact me off list, I may be able to help get you in touch with the right department to assist you. Sincerely, Trevor Hammonds ___ --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] Disconnection supervision: what about PBX
> Stephen Bosch wrote: >> Are BRI circuits what phone companies call "digital" lines for use >> with digital sets, such as with digital Centrex? >> >> I'm not aware that Telus even offers BRI. >> >> Sorry -- BRI is ISDN, not digital Centrex. >> >> I'm still not aware that Telus even offers ISDN anymore :) > > ...and by that I mean ISDN BRI ;) > > -Stephen- Throughout most of the United States, "Digital Centrex" or "CentrexIS" is ISDN as part of a Centrex group. If the circuit is meant for a single device, it would be a BRI. If the circuit is "Hi-Cap" or meant to be hooked up to a PBX or the like, it would be a PRI. I am not that familiar with Telus, but what Bell is calling "Digital Voice" service is merely VoIP over one of their DSL connections. While I know that both companies offer Centrex over PRI, I am unsure if either company supports BRI widely anymore. I know BRI service is available, and most of their switches are capable of offering BRI circuits. For example, digital secretarial enhanced key telephone sets are ISDN phones that work via a BRI. In my experience, most telcos in the US and Canada will not tell you about BRI unless you specifically ask. And if you do, they shuffle you off to another department where they may or may not know how to properly provision the circuit. Somehow, all the LECs in North American look at BRI as a data-only service and never really saw the advantages of offering it to voice-only customers. As such, now that 128k (or 144k) is too slow of a data connection for most, BRI has just been passed by. Such a shame... Trevor Hammonds ___ --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] Disconnection supervision: what about PBX
> From: Yuan LIU > Sent: Tuesday, February 06, 2007 8:11 PM > > After reading through several recent threads, I started to wonder why > the > Cisco document (and other VoIP documents) appears to present this issue > as > VoIP gateway specific. Don't (plain old) PBX' face the same issue if > they > use analogue interfaces? If there are analogue PBX' at all, how do > they > solve the problem? Yuan, Well engineered analogue PBXs typically do not use standard loop start subscriber lines. When digital trunks are not an option, they use analogue PBX and/or DID trunks. At the very least, ground start circuits are preferred to avoid "glare". The best call quality for analogue is achieved by using four-wire E&M trunks that provide answer and disconnect supervision. There are two-wire trunks (which are probably more common), as well as different signalling methods. These trunks require special interface hardware, and I am unaware of any that work with Asterisk. As the cards are typically very expensive, it is usually better to go with digital if you require that functionality. It would be nice to see a BRI interface for Asterisk that works in North America, as BRI circuits are often comparable in price to analogue lines. Sincerely, Trevor Hammonds ___ --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] Assterisk prompts
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Obelix wrote: > Does Asterisk have voice prompts for the following. > > 1. The number you dialled is not available. Please try again later. > > 2. The number you dialled is not recognised Take a look at the following URLs for a good list of the sounds available: http://www.voip-info.org/wiki-Asterisk+sound+files http://www.voip-info.org/wiki/view/Asterisk+sound+files+additional -BEGIN PGP SIGNATURE- Version: PGP Desktop 9.0.5 (Build 5050) iQEVAwUBRF3UajXfYB1SJBTyAQhhMQf+MHMROt2Ag6BNWB+0T173Nw39wAW5WOEq NkBVF7SYWfSsnkUFNbzDtFdwaJbdvwpgjoPbBJpZWuvw4RPm0DigSUSLYVKqwHlq 0QCUSGWqZMQy6bSVo2qTzhDJeGCqGjyT3XWsD7vPgwvuOXqzaFgg3x3kKeqRSR6n Xt4pFoz0D8uBR2AqV7+VAUTyWL7zvxtP7jF99Cd6XPUZs/XynYvKBexXHEjVVnzp 908Nm+wKV+9B22oY0NbW3OV/fynm7ae589uNV7JmDbBMQL+toauHFd7FMLVTd+0W w2jInIsBEIC6GN5rMMBEFTH/pvjCEF1kEGmxMxIjP5ctqkz4a0pZig== =oA6N -END PGP SIGNATURE- ___ --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] MOH from RCA jack?
Rich Adamson wrote on Friday, 17 February 2006 5:42 AM: > Have a potential client that wants to replace their old key system > with *, but they want to integrate a commercial message service (they > pay a monthly fee to have special MOH messages generated) into their > system. The messages are essentially delivered to this customer via > older generation audio equipment that interfaces to their key system > via a standard audio RCA jack. (We're reseaching other alternative > deliver mechanisms such as mp3's, etc, from the supplier, but have to > assume for now that we need to inject MOH audio into asterisk via > this RCA jack.) Why not just make an MP3 each time the client receives the updated media? You could have their existing MOH source device connected to the input of a PC in their office (or yours) like mentioned in previous messages. When the new media arrives, make a single MP3 file, and then upload it to the Asterisk server. Sincerely, Trevor Hammonds ___ --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 and NAT - best practices?
Leo Ann Boon <> wrote on Sunday, 22 January 2006 4:32 PM: > Trevor G. Hammonds wrote: > >> While I have not used siproxd, I have read a bit about it. From my >> understanding of the docs, the local SIP agents register to siproxd, >> but siproxd does not register to Asterisk. So the calls will >> traverse >> the NAT properly, but features like MWI will not work in this >> scenario. Also, this would be pure SIP URL dialling (e.g. >> [EMAIL PROTECTED]) as opposed to traditional telephone dialling >> (e.g. 1-213-555-8080). >> >> Please correct me if I am wrong, because I would really like to be >> (in this case). :-) >> >> > The docs are a little confusing. Look in the FAQ section: What types > of operation does siproxd support? > Here's the text. > >> 1) Siproxd as outbound proxy: >> - Configure your local client to register with some 3rd party >>service like Sipphone, FWD, Sipgate or any other. >> - Configure your local client to use siproxd as OUTBOUND PROXY >> >> Note: In this case, the local client does NOT register with >> siproxd but only with the external SIP restration service. The >> only condition is that siproxd needs to stay in the path of >> communication, therefore the local client must be configured as >> to use an OUTBOUND PROXY. >> > That's all you need to do. All your clients will still register to > Asterisk through siproxd, siproxd will take care of rewritting the > SIP headers to differentiate requests for each client. > > Leo Thank you, Leo! This is exactly what I need. I am going to play around with that really soon. Trevor ___ --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 and NAT - best practices?
Leo Ann Boon wrote on Saturday, 21 January 2006 6:21 PM: > Trevor G. Hammonds wrote: > >> How about when you have four or five SIP devices at a single >> location? Do you manually assign each phone a separate port and add >> firewall/router rules? I am looking for an inexpensive device or >> method that will allow this happen automatically. Rather than going >> that route, my current solution is to put an Asterisk server at the >> client's location to handle the SIP clients and do an outbound >> trunked IAX connection back to the main server. >> >> > Use an outbound proxy either a stanadlone appliance like ix-66 or you > can build one using Siproxd running on your Linux gateway. > http://siproxd.sourceforge.net/ > > There's a WIP port of siproxd to OpenWRT so you can run it on a > Linksys WRT54G. While I have not used siproxd, I have read a bit about it. From my understanding of the docs, the local SIP agents register to siproxd, but siproxd does not register to Asterisk. So the calls will traverse the NAT properly, but features like MWI will not work in this scenario. Also, this would be pure SIP URL dialling (e.g. [EMAIL PROTECTED]) as opposed to traditional telephone dialling (e.g. 1-213-555-8080). Please correct me if I am wrong, because I would really like to be (in this case). :-) Sincerely, Trevor Hammonds ___ --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] SPA-941 auto-answer capability
Mike Myers wrote on Friday, 20 January 2006 1:56 PM: > Hi. I am thinking about building an asterisk system for a small business > and want to be able to page through the phones. It seems like to do this > asterisk needs auto-answer support in the phone. I know the SPA-841's > support this, as do Cisco phones, but I have been unable to determine if > the new Cisco/Linksys SPA-941's do. > > Does anyone here have experience with trying to use auto-answer on the > SPA-941, or a pointer to some documentation that talks about this? > > Thanks, > Mike Mike, If you are running version 1.2 or SVN trunk up to revision 8059, you can make the SPA-941 auto-answer on the speaker phone by adding a dialplan entry like this before the Dial command: exten => 1234,1,SIPAddHeader(Call-Info:\;answer-after=0) If you have Polycom phones, you may also do this, but it requires a little more effort. Check the wiki (www.voip-info.org) for more information. Sincerely, Trevor Hammonds ___ --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 and NAT - best practices?
How about when you have four or five SIP devices at a single location? Do you manually assign each phone a separate port and add firewall/router rules? I am looking for an inexpensive device or method that will allow this happen automatically. Rather than going that route, my current solution is to put an Asterisk server at the client's location to handle the SIP clients and do an outbound trunked IAX connection back to the main server. Sincerely, Trevor Hammonds Mark Phillips wrote on Saturday, 21 January 2006 12:36 PM: > Most often the simple addition of nat=yes in the relevant sip.conf > stanza is all that's required to make a remote SIP phone work from > behind a firewall. > > for example > > [2201] > user=blah > secret=blah > auth=blah > allow=blah > host=dynamic > nat=yes > > I've been running 4 remote SIP phones across the internet from my > families houses all over the world in this manner. The only issues I > get are those of bandwidth availability or rather occasional lack of > it. > > Hosted PBX's are no different. The hosting service should be > providing a similar mechanism (although it might not be Asterisk > based). > > Mark, G7LTT/KC2ENI > Randolph, NJ > http://www.g7ltt.com > > > Michaël Gaudette wrote: >> Thanks Moises. I was kind of hoping that, at least if I hosted my >> Asterisk server somewhere where there was no NAT for the * box that >> the SIP phones wouldn't create any issues. >> >> How do you people with Hosted PBX handle the deployment of SIP phones >> behind NAT firewalls? Is it just elbow grease and configuring every >> single phone for the customer, or is there a way? >> >> Mike >> >> >> >> you can redirect the ports of the router as well. Or you can >> configure your SIP phone to use a STUN server. Please read in >> voip-info.org about SIP NAT, there are good suggestions. >> >> regards >> >> On 1/20/06, Michakl Gaudette <[EMAIL PROTECTED]> wrote: >> >>> Hello, >>> >>> I'm a bit new to SIP, and I've set up a SIP line with Asterisk and >>> my wholesale provider. That worked, fine. I ahd to open up the >>> ports on my router, forward them to the correct box, again fine. >>> >>> Now, if I get one of my customers to connect his SIP phone to my >>> Asterisk box, and HE'S behind a NAT firewall, does he have to go >>> through the same process, or is it just the Asterisk box that needs >>> to translate the SIP >> >> and >> >>> RTP port? >>> >>> In other words: if my SIP phone is behind a Linksys router, do I >>> need to configure the Router for any reason? >>> >>> Mike ___ --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] tuning an x100p in Australia for echocancellation
James Harper wrote on Saturday, 14 January 2006 3:17 PM: >> Are the AU telephone standards the same as US standards (eg, 600 ohm >> impedence)? > > This is a question I've been trying to answer too. I had a look at > the standard phone that Telstra would provide to customers about 5 > years ago, and it has an impedence switch on the bottom to toggle > between 'NORM' and '600', which suggests that 600 ohms isn't the > normal impedence. > > On an au configuration example for the pap2 I have seen on the web, > the impedence is set to '220+820||120nF', which suggests that our > standard here isn't 600. Indeed, the AU standard is just what you quoted. It is complex impedance. A 220-ohm resistive load connected in series to an 820-ohm resistive load which is connected in parallel with a 120nF cap. >>> Does anyone know of an addon device which can do impedence matching >>> on the line, or of a modification to the > card I don't know if this will help you, but you may do a Google search for the ETAL P3324 and P3356. I believe they may handle the impedance matching you need. Here is a link to a PDF spec sheets: http://www.ibselectronics.com/pdf/pa/etal/line_P3324.pdf http://www.ibselectronics.com/pdf/pa/etal/line_P3356.pdf -- Trevor Hammonds ___ --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] CALLERIDNUM::3 do not working on 1.2.1
You are using incorrect syntax. Notice where the close bracket is placed,
using your examples:
${CALLERIDNUM:3} erase first 3 digits
${CALLERIDNUM::3} returns first 3 digits
${CALLERIDNUM:3:3} should erase first 3 digits and return next 3 digits
Pisac wrote on Saturday, 14 January 2006 5:10 AM:
> No,
> ${CALLERIDNUM}:3 erase first 3 digits
> ${CALLERIDNUM}::3 returns first 3 digits
> ${CALLERIDNUM}:3:3 should erase first 3 digits and return next 3
> digits
>
> So,
> if
> ${CALLERIDNUM}=0123456789
> Then
> ${CALLERIDNUM}:3 returns 3456789
> ${CALLERDINUM}::3 returns 012
> ${CALLERIDNUM}:3:3 returns 345
>
> But this do not work anymore in 1.2.1, and if I do not found solution
> for this I will downgrade to 1.0.9
>
>
>
>
> - Original Message -
> From: "Dinesh Nair" <[EMAIL PROTECTED]>
>>
>> i believe the syntax is ${CALLERIDNUM:3} and not as you're using it
>> with double colons. also, the present accepted method is to use the
>> CALLERID() function rather than the variable which may be deprecated
>> in future releases.
>>
>> --
>> Regards, /\_/\ "All dogs go to heaven."
>> [EMAIL PROTECTED](0 0)http://www.alphaque.com/
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RE: [Asterisk-Users] SoCal Users Group Meeting Schedule
Kerry Garrison wrote on Monday, 9 January 2006 6:41 PM: > The SoCal Asterisk Users Group will be meeting at the Heritage Park > Public Library on the corner of Walnut and Yale in Irvine on the 3rd > Thursday every month. The following dates are already secured: > > Thurs Jan 19 > Thurs Feb 16 > Thurs Mar 17 What time? Any particular place within the library? Sincerely, Trevor Hammonds ___ --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] Sipura SPA-2002 Double Ring
Rich Adamson wrote on Sunday, 13 November 2005 7:36 PM: >> I recently implemented a Sipura SPA-2002 with one of my Asterisk >> installations. On internal calls, the SPA generates ringtone as >> expected. However, when I dial out via my IAX-based service >> provider, I hear >> both the telco-generated ringtone as well as the SPA-generated >> ringtone. Sometimes, the SPA continues to generate the ringtone even >> after the call has been answered. > > I don't have a spa-2002, but do use a spa3k. I doubt very much the > sipura device is actually providing ringback tone, and I don't recall > any parameters that would enable/disable such an item. (The Admin > manual does not mention it either.) > > You might check your extensions.conf entry for dialing your provider > to see if you have an "r" in that line. If so, remove it. The SPA-2002 is definitely generating the additional ringback. I verified this by temporarily changing the frequency of the ringback in the SPA's "Regional" settings. I also verified that I am not using the "r" option in the Dial command. If I were, however, only the Asterisk-generated ringback would be heard, and then only until the call supervised (i.e. I would not be hearing two distinct ring signals, and the ringback would not occasionally persist for the duration of a call while still hearing the called party). This problem is present only with the SPA-2002, and none of the other SIP devices connected to this Asterisk server. I have also tried making outbound calls via different service providers, all with the same results. Thanks again. Sincerely, Trevor Hammonds ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura SPA-2002 Double Ring
I recently implemented a Sipura SPA-2002 with one of my Asterisk installations. On internal calls, the SPA generates ringtone as expected. However, when I dial out via my IAX-based service provider, I hear both the telco-generated ringtone as well as the SPA-generated ringtone. Sometimes, the SPA continues to generate the ringtone even after the call has been answered. I saw this item mentioned on the list some time ago. However, I did not see a resolution. I am interested to know if anyone has come up with a "fix" for this situation. Any ideas or suggestions are appreciated. Thank you for your assistance. Sincerely, Trevor Hammonds ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
Joel Newkirk wrote on Friday, 30 September 2005 7:20 AM: > Looking into setting up a couple asterisk servers at a country club, > with VOIP phones in each of 100 short-term residential rental units. > Approx 100 extensions, approx 24 outside lines. > > Since everything is geographically at one location, reaching 911 > correctly shouldn't present a problem. However, the club wishes to > ensure that 911 authorities are able to identify the precise rental > unit placing the call. Mr. Newkirk, This and similar situations present a very serious issue for emergency responders. When you dial 911, your call is routed to the appropriate PSAP (Public Safety Answering Point) based on your ANI (Automatic Number Identification) or ELIN (Emergency Line Identification Number -- usually just another term for ANI). As your call arrives, the PSAP does a query of their ALI (Automatic Location Information) database to get your location information. Please note that the PSAP does NOT use Caller ID for this purpose. End users are not able to block their ANI (under normal circumstances), even though they may block their Caller ID. Either the ILEC or a company like Intrado will maintain the ALI database in your area. If you are getting your PRI and DIDs from your local ILEC, they would be responsible for getting the correct information entered into the ALI database. Typically, the information entered is only the physical address where the primary service is installed. In most circumstances, this information is enough to get police/fire/EMS to you in an emergency. However, I suspect the entire country club shares a single street address. If so, when someone dials 911, the PSAP will get only the main address of the country club. In this and similar situations, such as calling from within a multi-floor office building, a campus environment, etc., the main street address is simply not enough information to get emergency responders to you in a timely manner. Consider this not-so-unusual hypothetical scenario. A guest of the Pennsauken Country Club is having a heart attack in his bungalow. He dials 911. The dispatcher's screen at the PSAP shows the main information for the club "(856) 662-4961 - 3800 Haddonfield Rd - Pennsauken Country Club - Pennsauken, NJ". The guest explains that he is experiencing severe chest pain, then either passes out before he can tell the dispatcher his exact location at the country club, or is confused or unaware of his exact location. The dispatcher would roll fire, EMS, and/or police to the main address. However, when they arrive, the emergency responders would have to knock on all 100+ doors to even attempt to determine who was having the emergency. Now you probably have a dead guest. Not good for business. First off, you should be using a PRI to connect your Asterisk server to the PSTN. You should also have a block of DIDs, with each guest room assigned its own, unique DID. This way you can differentiate among the individual rooms when people are making outbound calls, and guests may receive incoming calls in their room without going through an operator. Asterisk is capable of setting ANI in addition to Caller ID, on a per-call basis. This would ensure that the correct data is sent to the phone company when someone dials 911. As to getting the data to the PSAP to indicate where within the country club each DID is assigned, you have a couple of solutions. You can implement PS/ALI (Private Switch/Automatic Location Identification), or you can work with your telecom provider to have them enter the extended data into the ALI database for each DID individually. PS/ALI is the "best" solution, from a technical standpoint -- but it is usually quite expensive. PS/ALI allows you to provide the E-911 system with current, specific tenant location information to expedite emergency response times to the site of the emergency -- not just to the building or general site location. So when your guest having a heard attack in room 119 dials 911, the PSAP gets something more along the line of "(856) 324-4119 - 3800 Haddonfield Rd - Building 5 Room 119 - Pennsauken Country Club - Pennsauken, NJ". PS/ALI is geared toward larger telecom users such as colleges, office buildings, large office campuses, etc., with a somewhat mobile population. It is utilized best when most of your extensions or DIDs are assigned to a person, as opposed to a location. This way, when the person moves from one office to another, your staff can push the change to the ALI database within minutes of the move, rather than phoning in a service order to the LEC, and waiting days for the change to be pushed to ALI. In your situation, I am assuming an extension or DID would most likely stay at a fixed location for quite some time (e.g. extension 4119 is always going to be guest room 119). So PS/ALI may be overkill in your situation. In that case, I would go the second route mentioned above. Work wit
[Asterisk-Users] Unable to transfer external calls to MeetMeconference (re-post)
This message was just bounced back to me. I am not sure if it made
it to the list originally or not, as I received no responses.
Since this message was written, I have installed Zap hardware into
this server. The Zap channels can be transferred to the Meetme
conference. The IAX2 calls still cannot.
Any suggestions will be greatly appreciated.
Sincerely,
Trevor Hammonds
Trevor G. Hammonds wrote:
> I have a peculiar situation, and am hoping someone on the list can
> offer assistance. I am running CVS HEAD, and am using ITSPs for
> DIDs. The server has no Zap hardware, but is configured to use
> ztdummy. All incoming calls are via IAX2.
>
> Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I
> am also able to transfer calls among my SIP devices, voice mail, IVR,
> etc. All of my SIP devices are able to call into a MeetMe
> conference without issue. However, when I attempt to transfer an
> inbound IAX call from one of my SIP devices to a MeetMe conference,
> the call is dropped. If I complete the transfer while the "You are
> currently the only person in
> this conference"
> prompt is playing, the call will successfully make it into the MeetMe
> conference, and remains without trouble. That is the ONLY
> circumstance in which I have been able to transfer an external user
> into the conference. Also, If I point a DID to the conference in
> extensions.conf, the call will ring right into the conference
> without trouble.
>
> As an aside, I created a few MOH queues and some corresponding
> extensions, so users may hear the music. When I try to transfer an
> external call to any of these MOH extensions, the external caller
> either hears silence, or the call is dropped. Either way, they never
> hear the MOH. I do not know if this is related, but I thought I
> would mention it.
>
> I have included CLI output below. Any assistance will be greatly
> appreciated.
>
> Sincerely,
> Trevor Hammonds
>
>
>
> Console output
>
> -- Accepting UNAUTHENTICATED call from x.x.x.x:
>> requested format = ulaw,
>> requested prefs = (ulaw),
>> actual format = ulaw,
>> host prefs = (),
>> priority = caller
> -- Executing Goto("IAX2/[EMAIL PROTECTED]", "default|4500|1") in new
> stack
> -- Goto (default,4500,1)
> -- Executing SetMusicOnHold("IAX2/[EMAIL PROTECTED]", "ultra-lounge") in
> new stack
> -- Executing Set("IAX2/[EMAIL PROTECTED]", "Mailbox=4500") in new stack
> -- Executing Dial("IAX2/[EMAIL PROTECTED]", "SIP/4500|20|t") in new stack
> -- Called 4500
> -- SIP/4500-b9aa is ringing
> -- SIP/4500-b9aa answered IAX2/[EMAIL PROTECTED]
> -- Started music on hold, class 'ultra-lounge', on IAX2/[EMAIL PROTECTED]
> -- Executing SetMusicOnHold("SIP/4500-98b6", "ultra-lounge") in
> new stack
> -- Executing MeetMe("SIP/4500-98b6", "8600|Ms") in new stack
> == Parsing '/etc/asterisk/meetme.conf': Found
> -- Created MeetMe conference 1023 for conference '8600'
> -- Playing 'conf-onlyperson' (language 'en')
> -- Started music on hold, class 'ultra-lounge', on SIP/4500-98b6
> -- Stopped music on hold on SIP/4500-98b6
> -- Stopped music on hold on IAX2/[EMAIL PROTECTED] Aug 18 22:14:55
> WARNING[24383]: app_meetme.c:841 conf_run: Error getting conference
> -- Hungup 'Zap/pseudo-2091567275'
> == Spawn extension (from-sip, 8600, 2) exited non-zero on
> 'IAX2/[EMAIL PROTECTED]' -- Hungup 'IAX2/[EMAIL PROTECTED]'
> == Spawn extension (default, 4500, 3) exited non-zero on
> 'SIP/4500-98b6'
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[Asterisk-Users] Passing variables across an IAX2 call
I have seen talk of adding the capability to pass variables on an IAX2 call. I would like to know if this is possible, yet. Thanks! Sincerely, Trevor Hammonds ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Balancing traffic between two routes
Sahil Gupta wrote on Tuesday, 23 August 2005 10:17 AM:
> We are currently running our own equipment to break calls out in a
> location I need to balance the calls out between two sites so that
> one site doesn't keep getting hit again and again.
>
> So currently we have something like this:
> exten => _1.,1,Dial(IAX2/pop1/${EXTEN})
> exten => _1.,2,Dial(IAX2/pop2/${EXTEN})
>
> But the above.. would hammer pop1 any tips ?
Try something like this:
exten => _1.,1,Random(50:3)
exten => _1.,2,Dial(IAX2/pop1/${EXTEN})
exten => _1.,3,Dial(IAX2/pop2/${EXTEN})
Or
exten => _1.,1,Random(50:4)
exten => _1.,2,Dial(IAX2/pop1/${EXTEN})
exten => _1.,3,Goto(whatever happens next)
exten => _1.,4,Dial(IAX2/pop2/${EXTEN})
exten => _1.,5,Goto(whatever happens next)
Hope this helps!
Sincerely,
Trevor Hammonds
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RE: [Asterisk-Users] Installing to a prefix.
Kevin Walsh wrote on Friday, 19 August 2005 6:58 PM: > Andrew Kohlsmith [EMAIL PROTECTED] wrote: >> On Friday 19 August 2005 21:27, Kevin Walsh wrote: >>> I'll send the modified Makefiles to anyone who needs them. >>> >> May I humbly request they be attached to a feature request on Mantis? >> > I've been less than humbly requested not to do that sort of thing any > longer, as I haven't signed a "disclaimer". Sorry about that. > > The Asterisk change is trivial; Just set the INSTALL_PREFIX variable > in the Makefile and then modify asterisk.conf and possibly > musiconhold.conf. The Zaptel Makefile changes are a bit more > involved. The diff file is 148 lines long. I've never had cause to > look at libpri. How about submitting a disclaimer to Digium for the modified makefiles? Sincerely, Trevor Hammonds ___ 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] FXO not picking up; baffled
Karl S. Katzke wrote on Friday, 19 August 2005 6:30 PM: > Of course, after spending a full 12 hours bashing my head against the > problem, I found my low voltage contractor had reversed the tip and > the ring. *bashes head against the wiring cabinet*... I feel your pain! :-) --Trevor Hammonds ___ 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] Unable to transfer external calls to MeetMeconference
Trevor G. Hammonds wrote on Thursday, 18 August 2005 10:43 PM: > I have a peculiar situation, and am hoping someone on the list can > offer assistance. I am running CVS HEAD, and am using ITSPs for > DIDs. The server has no Zap hardware, but is configured to use > ztdummy. All incoming calls are via IAX2. > > Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I > am also able to transfer calls among my SIP devices, voice mail, IVR, > etc. All of my SIP devices are able to call into a MeetMe conference > without issue. > However, when I attempt to transfer an inbound call from one of my > SIP devices to a MeetMe conference, the call is dropped. If I > complete the transfer while the "You are currently the only person in > this conference" > prompt is playing, the call will successfully make it into the MeetMe > conference, and remains without trouble. That is the ONLY > circumstance in which I have been able to transfer an external user > into the conference. > Also, If I point a DID to the conference in extensions.conf, the call > will ring right into the conference without trouble. > > As an aside, I created a few MOH queues and some corresponding > extensions, so users may hear the music. When I try to transfer an > external call to any of these MOH extensions, the external caller > either hears silence, or the call is dropped. Either way, they never > hear the MOH. I do not know if this is related, but I thought I > would mention it. An update... I added an analogue FXO card to this Asterisk server, and calls coming in on the Zap channels are able to be transferred to the conference. However, they do not hear MOH at all (in the conference or otherwise). I have to be missing something. Hopefully it is obvious to one of you, because it is obviously escaping me. ;-) Sincerely, Trevor Hammonds ___ 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] 1-800 number
Christoph Eicke wrote on Friday, 19 August 2005 12:15 AM: >> Just call a milliwatt..? > you have a number? > I'm also willing to pay my regular fees to my provider for those 3-4 > minutes of testing. You can reach the milliwatt test lines throughout most of the former Pacific Bell region of SBC on NPA-NXX-0002 (e.g. 714-999-0002, 213-473-0002, 909-390-0002, etc.). If you want some numbers that keep the line open for a long period of time, and you actually have to listen to it, here are some suggestions: 202-762-1401 USNO Master Clock, Washington, D.C. (UTC/ET) 202-762-1069 USNO Master Clock, Washington, D.C. (UTC/ET) 719-567-6742 USNO Master Clock, Colorado Springs, CO (MT) 303-499-7111 NIST Atomic Clock (WWV), Fort Collins, CO (UTC) 808-335-4363 NIST Atomic Clock (WWVH), Kauai, HI (UTC) 800-525-7623 American Express Foreign Exchange Rates Hope this helps! Sincerely, Trevor Hammonds ___ 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] Unable to transfer external calls to MeetMe conference
I have a peculiar situation, and am hoping someone on the list can offer
assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server
has no Zap hardware, but is configured to use ztdummy. All incoming calls
are via IAX2.
Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also
able to transfer calls among my SIP devices, voice mail, IVR, etc. All of
my SIP devices are able to call into a MeetMe conference without issue.
However, when I attempt to transfer an inbound call from one of my SIP
devices to a MeetMe conference, the call is dropped. If I complete the
transfer while the "You are currently the only person in this conference"
prompt is playing, the call will successfully make it into the MeetMe
conference, and remains without trouble. That is the ONLY circumstance in
which I have been able to transfer an external user into the conference.
Also, If I point a DID to the conference in extensions.conf, the call will
ring right into the conference without trouble.
As an aside, I created a few MOH queues and some corresponding extensions,
so users may hear the music. When I try to transfer an external call to any
of these MOH extensions, the external caller either hears silence, or the
call is dropped. Either way, they never hear the MOH. I do not know if
this is related, but I thought I would mention it.
I have included CLI output below. Any assistance will be greatly
appreciated.
Sincerely,
Trevor Hammonds
Console output
-- Accepting UNAUTHENTICATED call from x.x.x.x:
> requested format = ulaw,
> requested prefs = (ulaw),
> actual format = ulaw,
> host prefs = (),
> priority = caller
-- Executing Goto("IAX2/[EMAIL PROTECTED]", "default|4500|1") in new stack
-- Goto (default,4500,1)
-- Executing SetMusicOnHold("IAX2/[EMAIL PROTECTED]", "ultra-lounge") in new
stack
-- Executing Set("IAX2/[EMAIL PROTECTED]", "Mailbox=4500") in new stack
-- Executing Dial("IAX2/[EMAIL PROTECTED]", "SIP/4500|20|t") in new stack
-- Called 4500
-- SIP/4500-b9aa is ringing
-- SIP/4500-b9aa answered IAX2/[EMAIL PROTECTED]
-- Started music on hold, class 'ultra-lounge', on IAX2/[EMAIL PROTECTED]
-- Executing SetMusicOnHold("SIP/4500-98b6", "ultra-lounge") in new
stack
-- Executing MeetMe("SIP/4500-98b6", "8600|Ms") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600'
-- Playing 'conf-onlyperson' (language 'en')
-- Started music on hold, class 'ultra-lounge', on SIP/4500-98b6
-- Stopped music on hold on SIP/4500-98b6
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]
Aug 18 22:14:55 WARNING[24383]: app_meetme.c:841 conf_run: Error getting
conference
-- Hungup 'Zap/pseudo-2091567275'
== Spawn extension (from-sip, 8600, 2) exited non-zero on 'IAX2/[EMAIL
PROTECTED]'
-- Hungup 'IAX2/[EMAIL PROTECTED]'
== Spawn extension (default, 4500, 3) exited non-zero on
'SIP/4500-98b6'
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RE: [Asterisk-Users] Password for Conf Room
>From http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe Authenticated conference room exten => 18,1,Answer exten => 18,2,Wait(1) exten => 18,3,Authenticate(5678) exten => 18,4,MeetMe(18|p) exten => 18,5,Playback(vm-goodbye) exten => 18,6,Hangup Hope this helps. -- Trevor Hammonds From: Sharadindu Mohanty Sent: Thursday, 18 August 2005 5:26 AM Hi, Can i set Password to Conf rooms?Please Let me know the procedure. Thanks Sharadindu Mohanty ___ 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] IAX2 one way audio
Duane Cox wrote on Friday, 29 April 2005 10:17 AM: > Do you get 2-way audio that sometimes drops off to 1-way audio then > picks back up as 2-way? (Thats what I see) Not sure if my problem is > a lost packet issue as I am sending IAX off net. My experience has been that there is no two-way audio, except that I am able to send a SINGLE, brief touch tone digit. Let me know if I can provide anything to assist in getting this bug fixed. Sincerely, Trevor Hammonds ___ 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] problem with MUSICONHOLD
Gianluca Colucci wrote on Wednesday, 16 March 2005 9:54 AM: > So, it seems to work: when I call the 98 extension I can hear the > music. > > The problem is when I try to make some test with three other > extensions: I call one of these from my SIP client, the called > answers me and I try to transfer the call to a third extension, the > called listen no music. Make sure you are using a recent version of Asterisk. I suggest downloading from CVS and recompiling to ensure you are up to date. This was a problem in both CVS HEAD and Stable versions of asterisk a couple of weeks ago. Sincerely, Trevor Hammonds ___ 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.
Linn Boyd wrote on Monday, 14 March 2005 3:05 PM: > Finally I got someone on the phone reguarding this and they said > "don't bother us until after the 18th as we won't do anything, it > says 10 days to provision a vanity" I told them, if the site said > that, I wouldn't have accepted the terms. Seven to 10 days to provision an 8XX number where a change of RESP ORG is involved is certainly not unreasonable. In fact, that is a fairly good turnaround time, compared to MANY other ITSPs out there. I have had great luck with NuFone for quite some time, save one technical problem which was promptly resolved. While some of NuFone's employee's emails (to this list, at least) may be terse, they have never been unreasonable, in my opinion. They run a good service. Sincerely, Trevor Hammonds ___ 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] ASTCC and NuFone billing is different!!
Ronald Wiplinger wrote on Friday, 11 March 2005 8:14 AM: > I have ASTCC installed, and compare it with NuFone, however, I find > that the billing of NuFone is always a few secondes more (6 to 24 > seconds) > > Does anybody has an explanation / solution for it? Is the difference on the inbound or outbound legs of the call? Sincerely, Trevor Hammonds ___ 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] Music on hold on timing sources
> From: Marty Mastera > Sent: Wednesday, 2 March 2005 1:01 PM > > I have read that music on hold requires a timing source (which I > never had to worry about previously since the server had zaptel hardware > in it)...now I'm configuring a server in a colo which has no zaptel > hardware. You can use ztdummy as your timing source. http://www.voip-info.org/wiki-Asterisk+timer+ztdummy > If I use the dialplan to run MusicOnHold(), I do get the music > upon dialling that extension, but if I try to use the hold button > on either a 7960 or X-Lite I get nothing. Is this the expected > behavior? I figured that if a timing source was needed that > MusicOnHold() should not work, but it does This is a result of a bug (3701) which has been corrected in CVS. Updating your installation with the latest CVS code from either stable or head will get MOH working properly again. Sincerely, Trevor Hammonds ___ 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] Limit the call & recording when pressing *1
Joseph wrote on Saturday, 26 February 2005 8:38 PM:
> I'm testing two options from dial command and can not make them to
> work.
< < < S N I P > > >
> exten => 21,1,Dial(${phone1},20,r,w)
> exten => 21,1,Dial(${phone1},20,r,L(5[:4][:1]))
Try:
exten => 21,1,Dial(${phone1},20,rw)
Options should be combined.
and
exten => 21,1,Dial(${phone1},20,rwL(30:24:6))
This limits the call to five minutes, warning every 60 seconds when four
minutes are remaining. Keep in mind that time is specified in milliseconds.
(FYI: There are 60,000 milliseconds in a minute.)
Sincerely,
Trevor Hammonds
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RE: [Asterisk-Users] Re: FRS and GMRS via *
Rich Adamson wrote on Friday, 25 February 2005 4:18 AM: >> GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 >> CFR 95.141). There has been a lot of talk from lobbyists to clarify >> this rule, but as it stands you could conceivably connect a *private* >> network to GMRS or MURS radios (you can't make any plugins or >> modifications to an FRS radio that isn't type accepted with the >> radio, so connecting a phone line or * box would be out). The >> language is vague, see the history at http://www.provide.net/~prsg/ > > Would plugging into the headphone jack with a phone-patch-type device > be considered a modification for radios with vox capability? You may not legally connect any device to an FRS radio which has not been FCC-certified as part of that radio. This means that you cannot legally connect anything to the headset jack other than the headset (or other device) the manufacturer certified for that particular model of FRS radio. The FCC has not approved, and is not likely to approve, any phone patch or interconnection device for FRS radios. A first-offense violation of this rule will result in a fine of $10,000. GMRS and MURS radios have different rules. I would look to using them for this type of interconnection. Sincerely, Trevor Hammonds ___ 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] Problem with Avaya 4602 / SIP response 481
Trevor G. Hammonds wrote on Monday, 21 February 2005 2:54 PM: > I suspect this may be related to the MWI indicator and the "mailbox=" > statement in my sip.conf file, as the last time I was using this > phone with *, it was not set up to use voicemail. I have confirmed that this issue is directly related to the voice mail MWI. After commenting out the "mailbox=" statement in my sip.conf, the problem has not presented itself in over 10 hours. If anyone has any ideas or suggestions, I would appreciate hearing them. I really would like to use the MWI light. Sincerely, Trevor Hammonds ___ 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] Problem with Avaya 4602 / SIP response 481
I have an Avaya 4602 IP phone that was previously working with Asterisk. It was being used elsewhere for several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98 After this message, the phone no longer works properly -- if at all. When attempting to dial, the phone seems to ignore my dial plan, accepting no more than five digits (I get the tone burst indicating that it is done accepting digits), and it does not transmit anything to Asterisk. Attempting to call the phone's extension results in a busy condition (SIP response 486). Soft resetting the phone restores it to working order for several minutes. I have searched Google and the Wiki for answers to this problem, and see that Brian Elton has experienced this exact situation. However, I saw no resolution posted to the list, and do not have Brian's contact information. I suspect this may be related to the MWI indicator and the "mailbox=" statement in my sip.conf file, as the last time I was using this phone with *, it was not set up to use voicemail. Any ideas or assistance will be greatly appreciated. Sincerely, Trevor Hammonds SIP Debug output: Sip read: SIP/2.0 481 Call Does Not Exist Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY From: "asterisk" ;tag=as5860bf17 To: ;tag=cad443b1cd74b1e Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK69c2e97b Content-Length: 0 Contact: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 9 headers, 0 lines -- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98 Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' ___ 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] Brian Elton / Avaya 4602
I would like to get in contact with Brian Elton. He posted information to this list regarding problems with an Avaya 4602, late last year. I am now experiencing a similar issue, and would like to know if/how it was resolved. Thank you. Sincerely, Trevor Hammonds ___ 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] Extra sounds (Weather)
Liaan vd Merwe wrote on Wednesday, 16 February 2005 2:53 AM: > This is the example script (extracted from that link) you will need > to find a weather page for your region an then change the urls and > grep statements chow L Once again, this is NOT the script mentioned at Eric Wieling's former site, http://www.fnords.org/~eric/asterisk/, referenced it the message in the archives at http://lists.digium.com/pipermail/asterisk-users/2003-November/025983.html. Sincerely, Trevor Hammonds ___ 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] Extra sounds (Weather)
Liaan vd Merwe wrote on Tuesday, 15 February 2005 1:37 PM: > http://edgett.bc.ca/simonsays/archives/000228.html Thank you, but this is not the script. Sincerely, Trevor Hammonds ___ 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] Extra sounds (Weather)
C F wrote on Tuesday, 15 February 2005 10:23 AM: > http://lists.digium.com/pipermail/asterisk-users/2003-November/025983.html Unfortunately, the site referenced in that message is no longer active. It mentions that the code was donated to the Asterisk Documentation Project, but I see no reference to the code on that site. Any ideas? Sincerely, Trevor Hammonds ___ 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] Speech Recognition
dean collins wrote on Saturday, 12 February 2005 9:16 PM: > Check out www.angel.com For that matter, check out Tellme. 1-800-555-TELL Speaker-independent automatic speech recognition, when implemented properly, is VERY good right now. However, good ASR is usually fairly expensive. Do not confuse desktop speech recognition applications like Dragon Dictate and Via Voice with telco-grade ASR engines like Nuance, SpeechPearl/Speechworks, Loquendo, etc. Tellme has a developer platform that you can use to experiment with VoiceXML, TTS, and ASR. You create the "voice applications" on their website, and can access them via the PSTN or SIP. Check out: http://studio.tellme.com/ I, for one, would love to have the ability to use ASR engines with Asterisk. I think a good start would be a Sphinx/Asterisk integration project. Sincerely, Trevor Hammonds ___ 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] ASTCC error on free calls
Karl H. Putz wrote on Saturday, 5 February 2005 11:45 AM:
> The problem is a "division by zero" issue in the astcc.agi @ line 538
> (I have made a few mods so my line #'s may not be exactly the same).
> The line reads
>
> $maxmins = int(($credit - $adjconn) / $adjcost);
>
> you may want to change the script to something like:
>
> if ($adjcost < 1) {
> $maxmins =call>; } else { $maxmins = int(($credit - $adjconn) / $adjcost); }
That did the trick! Thank you, Karl!
> I set $maxmins = $credit on my system and that will give 1 minute per
> penny of account balance as the max duration of the call but will not
> charge any duration related fees against the account. Any connection
> fees would still be assessed at call termination, though.
Good idea. I did this as well.
Sincerely,
Trevor Hammonds
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[Asterisk-Users] ASTCC error on free calls
I set up certain routes in my ASTCC application to be free of charge. When a user attempts to dial one of these numbers, the announcement plays the prompts "This call will cost", "nothing", and then terminates the script, dropping the call, leaving the card locked in the database as being in use. Any ideas? Sincerely, Trevor Hammonds ___ 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] astcc digit timeout
Karl H. Putz wrote on Thursday, 3 February 2005 12:14 PM:
> modify the agi to add the specific inter-digit timeout in
> milliseconds you would like after the prompt filename in the get_data
> calls.
>
> i.e. use: $cardno = $AGI->get_data("astcc-accountnum",5000);
>
> if you want a 5 second allowable delay between digits.
Karl,
This is a great suggestion! I think it should be added to the ASTCC
distribution.
Sincerely,
Trevor Hammonds
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RE: [Asterisk-Users] Good 800 Number provider
Ed Greenberg wrote on Thursday, 3 February 2005 11:46 AM: > Are there any recommendations for high quality providers that will > assign a Toll Free number and deliver it over VOIP, while still > allowing port-out if the service doesn't work out? I have had great luck with TXLink. http://txlink.net/voipsolutions.php Sincerely, Trevor Hammonds ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic Boards
James Ellis wrote on Tuesday, 1 February 2005 4:34 PM: > Alright I was able to download the DNA 5.1 drivers from Intel for the > D/41 card that I have and get the card working. Now my question is > what or where do I configure its use in Asterisk. > > Thanks. > > Jim As I understand it, you need to purchase the drivers from Digium. -- Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial and Macro Do not seem to be working
Randy Johnson <> wrote on Thursday, 27 January 2005 9:02 PM:
> Hello,
>
> Here is the dial command:
>
> exten =>
> 790,2,Dial(SIP/[EMAIL PROTECTED]|60|M(screen^${CALLERIDNUM}))
>
>
> Here is the macro
>
> [macro-screen]
> exten => s,1,Wait(0.2)
> exten => s,2,say number ${ARG1}
Perhaps you should try:
exten => s,2,SayDigits(${ARG1})
> exten => s,3,Read(ACCEPT|screen-accept|1)
> exten => s,4,GotoIf($[${ACCEPT} = 1 ] ?7:6)
> exten => s,5,SetVar(MACRO_RESULT=CONTINUE)
> exten => s,6,System(/bin/rm ${ARG1})
>
>
> When I do the above it dials the extension, user picks up but it does
> not say the number. the number is actually getting passed correctly
> because it is in the logs... but it does not say the number.
>
> It is almost as if the macro is not getting ran???
>
> Any thoughts?
>
> Thanks!
>
> Randy
Good luck!
Sincerely,
Trevor Hammonds
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RE: [Asterisk-Users] Asterisk Install on Kernel 2.6.x
> is it possible to build asterisk/zaptel on a linux 2.6.x kernel? Yes. ___ 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] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)
First off, ensure you have the following packages installed (using YAST or otherwise): CVSupkernel-sourceskernel-symsncursesncurses-developensslopenssl-develbisondoxygen Next, ensure you symlink /usr/src/linux to the current linux source (e.g. /usr/src/linux-2.6.5-7.104). Finally, do the following: #cd /usr/src/linux #make cloneconfig && make dep This worked for me. As usual, however, YMMV. Let me know if this works for you. Sincerely, Trevor Hammonds From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johannes van HulstSent: Monday, August 16, 2004 9:30 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk) Have anybody experience with the following error on a linux system. The system is for the rest running perfect without problems. The system was full installed with Suse 9.1 and updated. Following uname is my kernel 2.6.5-7.104-default Greatings Han Suse 9.1 professional AMD Atlhon XP 2200 Asus A7V600-X bios 1005 1Gb memory 400Mhz Geforce MX 4000 64MB 40 GB Harddisk US040814:/usr/src # ln -s /lib/modules/2.6.5-7.104-default/build linux-2.6US040814:/usr/src # cd zaptelUS040814:/usr/src/zaptel # make cleanrm -f torisatool makefw tor2fw.hrm -f zttoolrm -f *.o ztcfg tzdriver sethdlc sethdlc-newrm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lorm -f *.ko *.mod.c .*o.cmdrm -f gendigits tones.hrm -f libtonezone*rm -f tor2eerm -f coreUS040814:/usr/src/zaptel # make linux26cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.ccc -o gendigits gendigits.o -lm./gendigitscc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA makefw.c -o makefw./makefw tormenta2.rbt tor2fw > tor2fw.hLoaded 69900 bytes from filecc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o ztcfg.o ztcfg.ccc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -o zonedata.lo zonedata.ccc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -o tonezone.lo tonezone.car rcs libtonezone.a zonedata.lo tonezone.locc -o ztcfg ztcfg.o -lm -L. libtonezone.acc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o torisatool.o torisatool.ccc -o torisatool torisatool.occ -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o ztmonitor.o ztmonitor.ccc -o ztmonitor ztmonitor.occ -c ztspeed.ccc -o ztspeed ztspeed.omake -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modulesmake[1]: Entering directory `/usr/src/linux-2.6.5-7.104-obj/i386/default'make -C ../../../linux-2.6.5-7.104 O=../linux-2.6.5-7.104-obj/i386/default modules WARNING: Symbol version dump /usr/src/linux-2.6.5-7.104-obj/i386/default/Module.symvers is missing, modules will have CONFIG_MODVERSIONS disabled. CC [M] /usr/src/zaptel/zaptel.o CC [M] /usr/src/zaptel/tor2.o CC [M] /usr/src/zaptel/torisa.o/usr/src/zaptel/torisa.c:1139: warning: `set_tor_base' defined but not used CC [M] /usr/src/zaptel/wcusb.o CC [M] /usr/src/zaptel/wcfxo.o CC [M] /usr/src/zaptel/wcfxs.o/usr/src/zaptel/wcfxs.c: In function `wcfxs_interrupt':/usr/src/zaptel/wcfxs.c:813: internal compiler error: Segmentation faultPlease submit a full bug report,with preprocessed source if appropriate.See for instructions.The bug is not reproduceable, so it is likely a hardware or OS problemmake[4]: *** [/usr/src/zaptel/wcfxs.o] Error 1make[3]: *** [_module_/usr/src/zaptel] Error 2make[2]: *** [modules] Error 2make[1]: *** [modules] Error 2make[1]: Leaving directory `/usr/src/linux-2.6.5-7.104-obj/i386/default'make: *** [linux26] Error 2 US040814:/usr/src/zaptel #
[Asterisk-Users] Difference between Tormenta/Zapata and Digium Hardware
Title: Message I am interested in possibly building a few Tormenta 2 Rev B cards for myself. Before I get much further, though, I would like to determine the difference between the T2B card and the Digium Quad-span T1 cards. I get the general impression that they are similar, but the Digium hardware is improved. Is this the case? Are the drivers for the T2B valid anymore? Thanks for any advice or assistance. Sincerely, Trevor Hammonds
