Re: [Asterisk-Users] newbie question; can * screen calls?

--On Friday, January 09, 2004 10:11 PM -0600 Alan Andrews <[EMAIL PROTECTED]> wrote: On Fri, 2004-01-09 at 20:55, Ken Alker wrote: Does * have the capability to screen calls? IOW, if someone calls in from outside (ie. not a local extension), can * ask the calling party to state their name, rec

Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway

--- Steve Underwood <[EMAIL PROTECTED]> wrote: > WipeOut wrote: > > > Granted five 9's is never easy but in a cluster of 10+ servers the > > system should survive just about anything short of an act of God.. > > You do realise that is a real dumb statement, don't you? :-) > > A cluster of 10 m

[Asterisk-Users] picking a channel bank

I have never had to pick out a channel bank before but I'd like to use one with the Digium T-1 card to hook 8 analog CO lines to an * PBX. Is there a reference somewhere describing and comparing channel banks (old and new)? Can modern channel banks handle translating all the "new" analog signal

Re: [Asterisk-Users] file_inlcude .. why not?

Lion Templin wrote: TeleSIP wrote: Don't know if this has been addressed, but why isn't there a file_include style directive for extensions.conf? there is...search the archives or the wikiits something like #include filename.conf Oh yeah, it works, thanks .. Not entirely obvious, I guess .

Re: [Asterisk-Users] AbsoluteTimeout Users Messages

Andy Powell wrote: Nicolas, I'd appreciate a copy of this if possible... got a url where I can grab it? Thanks Andy *** REPLY SEPARATOR *** On 09/01/2004 at 10:43 Nicolas Gudino wrote: Andy Powell wrote: I'd be nice to be able to play a tone (or message) at AbsoluteTimeou

[Asterisk-Users] Music_on_hold adjust volume

Hi all, is there a posibility to change the volume of the music-on-hold ?? I tried with the different groups with default, and loud setup, but no changes. And the music is a little bit to loud ?? Are there any options, to deal with ?? Or do I have to recode my mp3 in any way ?? Thanks for any

Re: [Asterisk-Users] ChanIsAvail and SIP

B. J. Bomar wrote: Hello all. Has anyone had any success using ChanIsAvail with only SIP channels? Is there another, better way to check if an extension is busy without dialing it? Well, SIP devices live their own life and should really handle this signalling themselves. That's why ChanIsAvail

Re: [Asterisk-Users] AbsoluteTimeout Users Messages

Olle E. Johansson wrote: Andy Powell wrote: Nicolas, I'd appreciate a copy of this if possible... got a url where I can grab it? Thanks Andy *** REPLY SEPARATOR *** On 09/01/2004 at 10:43 Nicolas Gudino wrote: Andy Powell wrote: I'd be nice to be able to play a tone (or

Re: [Asterisk-Users] Chagres Technologies, Inc

Mail John Brown at Chagres. [EMAIL PROTECTED] He usually responds quickly and I get information about where my products are. Yes, I also have rest orders, but I have acceptable responses on why and when they are expected to arrive in this snowy winterland... /O ___

Re: [Asterisk-Users] Called Party Identification

Steven Critchfield wrote: On Fri, 2004-01-09 at 22:40, Brent Franks wrote: Does * support Called Party Identification? Say for example, I dial extension 2000, SIP sends back John Doe from the sip.conf file where extension 2000 is defined? Would this violate the SIP RFC? Maybe you didn't think

Re: [Asterisk-Users] Called Party Identification

Brent Franks wrote: No, but the Caller ID Information for a SIP extension is stored in sip.conf, so yes, I did think about that. As far as making sense, many meridian systems do this, and it is quite helpful. This could help with the implementation of gastman, and also end user phones. On the Ci

Re: [Asterisk-Users] disclaimed or not - that's the question /* New subject */

WipeOut wrote: And make sure to send in a disclaimer otherwise it will not even be looked at.. :) How do we know what is disclaimed or not disclaimed? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aste

RE: [Asterisk-Users] file_inlcude .. why not?

Lion Templin wrote: > Don't know if this has been addressed, but why isn't there a > file_include style directive for extensions.conf? > > I find that my extensions.conf grows a lot, and it would be a lot > nicer to have a tree of files rather than one big file to try and > navigate. Also, I've go

RE: [Asterisk-Users] USA dial plan

Title: Message >Yes, in most places in the USA local calls are totally free, no per min>charge.   This is not true in the US for business lines.  Residential lines have a "free" local calling area.  However, business lines from an incumbent local exchange carrier like SBC nearly always charg

Re: [Asterisk-Users] disclaimed or not - that's the question /* New subject */

Olle E. Johansson wrote: WipeOut wrote: And make sure to send in a disclaimer otherwise it will not even be looked at.. :) How do we know what is disclaimed or not disclaimed? /O Digium have all the Disclaimers and will not develop or include any code into the CVS without one.. Thats all I

[Asterisk-Users] Call transfer message

Hi all, A feature I think should be included in 1.0 version is playing a message to calling and called party while the call is being transferred. Something like this: Calling party (whose call is being transferred) "Please wait, your call is being transferred" Called party (who is transferring

Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway

Hi, I don't want to drag this into a long thread, but note the original says "the system should survive just about anything short of an act of God", and suddenly you are talking about a reliable server and a few switches. These are quite different things. I have yet to see a 5 x 9's server roo

[Asterisk-Users] Forums Need Help

Morning All,   I have created some virgin forums that I think may relinquish the mailing lists from major burdens.  Everything is .001 in version and I need help.   I need some advice as far as images and content.  I know the project is opensource but is content and graphics?  If not can you

[Asterisk-Users] RE: Client for P800/P900

Hi, is there a client which can be used on the SonyEricsson P800/P900...? I must agree, as a P800 owner, this would be sweet. However, it's uses are somewhat limited unless you also found some WLAN hardware for it ? (Doing VoIP over a 9.6 GSM dialup or even GPRS does not really make sense to me) N

Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway

Automated failover is a nice thought in this instance but in the Telco world it may not be necessary. Most industries will allow for weekend work as well as planned downtime (Yes, even in a three shift manufacturing facility) In my experience, fires and acts of God are far and few between but some

[Asterisk-Users] FYI: New SIP Flash Image 7940/7960 IP Phone

IP Phone Release 6.1 Bulletin Cisco IP Phone firmware images now contain a Cisco digital signature to provide authentication for improved security. Once firmware version 6.1 is loaded onto a phone, all subsequent firmware versions will be checked for authenticity. Firmware images prior to release

Re: [Asterisk-Users] Forums Need Help

> Morning All, > > I have created some virgin forums that I think may relinquish the mailing > lists from major burdens. Everything is .001 in version and I need help. > > I need some advice as far as images and content. I know the project is > opensource but is content and graphics? If not can

Re: [Asterisk-Users] Forums Need Help

Steve Totaro wrote: check it out at www.asteriskhelpdesk.com/forums I hope that you are aware there already are one or several forums, mostly ignored by the community. See http://asterisk.xvoip.com/ Xvoip also tried setting up a business list for Asterisk.

Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway

> I don't want to drag this into a long thread, but note the original says > "the system should survive just about anything short of an act of God", > and suddenly you are talking about a reliable server and a few switches. > These are quite different things. I have yet to see a 5 x 9's server

Re: [Asterisk-Users] (no subject)

T. Chan wrote: I recently came across DynEXTENdb, a way to be able to include thousands of Extensions (routes). In my application which is VOIP, we need to include more than 50,000 area codes due to the USA LATA routing, and there is simply no way to do that with extensions.conf. The way DynEX

RE: [Asterisk-Users] Mailing list growth

Hi! > [...] > > "higher-level implementation" list that deals specifically with > > channelbanks & T1 issues (=larger installations). VoIP will remain on > > asterisk-users. > [...] > > That doesn't quite sound right. Maybe it is from your perspective, but > are you telling me that the NOCs wi

[Asterisk-Users] E100P - Error 500

Hi, I am running * with E100P board. At least every our I got an Error 500 message and ISDN-PRI restarts: Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 12:54:27 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_erro

Re: [Asterisk-Users] USA dial plan

In some places, yes, but not all places. In Louisiana, for example business can get unlimited local calling (and most do). When I lived in Calif unlimited local calling was not available to businesses. Scott Stingel wrote: Just a little clarification on USA local calling: Local calls are gen

Re: [Asterisk-Users] AbsoluteTimeout Users Messages

> > Andy Powell wrote: > > > >> Nicolas, > >> > >> I'd appreciate a copy of this if possible... got a url where I can > >> grab it? > >> > >> Thanks You can grab a copy from the bugtracker: http://bugs.digium.com/bug_view_page.php?bug_id=773 I've already sent the disclaimer to Digium.. Bes

RE: [Asterisk-Users] Mailing list growth

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Philipp von Klitzing > Sent: Saturday, January 10, 2004 10:35 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Mailing list growth > > > - asterisk-users: VoIP and Asterisk in general (in

Re: [Asterisk-Users] Mailing list growth

everything is free or the cost of shipping if you think... dont worry, newbs will land at my forums but i still wanna know if i can cut and paste FAQs and the like. I plan on it so sue me, rofl. - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, Janu

RE: [Asterisk-Users] file_inlcude .. why not?

> I would agree with this as well. This way it should be much easier to > provide "virtual asterisk services"! > We all agree that * will be apache of VOIP! :) Well, apache has virtual > directives and include directives!!! > > Ta > SJ If you understand contexts and how to use them correctly, yo

RE: [Asterisk-Users] E100P - Error 500

I get lots of these in a very busy system, along with PRI frame errors/retransmissions. It is my understanding that this is due to an inadequate buffering mechanism in asterisk. Mark Spencer is aware of the problem, and has said he'll work on it soon. In small numbers, these can be safely ignore

Re: [Asterisk-Users] newbie question; can * screen calls?

Hi! > Does * have the capability to screen calls? IOW, if someone calls in from > outside (ie. not a local extension), can * ask the calling party to state > their name, record it, ring the recipient, play the caller's name for the > recipient, then give the recipient the choice of answering o

Re: [Asterisk-Users] Broken DNS makes Asterisk whacky!

Hi! > When DNS (or outside connection to the network, not sure which) is > broken and you have "register=>" lines in iax.conf, Asterisk gets whacky. > ... > Later, I get some other generally bizzare behavior including: > 1. I get "everyone is busy at this time" from devices that aren't. I mi

[Asterisk-Users] Codec problems (SIP)

I am trying to get my Voicetronix OpenLine4 card working in FXO mode in a PBX setup - so far I can only get it working as an IVR. I have managed to get my card to at least not crash now, and Asterisk does recognise it's existence... but I seem to be having codec problems. The same problems exist wh

Re: [Asterisk-Users] crontab

oHi! > Ladies and Gentlemen, can anyone please help and let me know what is > the way to start Asterisk automatically using a cronjob, thanks http://www.voip-info.org/wiki-Asterisk+administration Philipp ___ Asterisk-Users mailing list [EMAIL PROT

Re: [Asterisk-Users] Fwd: new cvs build failure

On Saturday 10 January 2004 12:36 am, Martin wrote: > I did a previous find / -name install and it couldn't find it, but I just > couldn't believe it. > The only thing I did recently was a kernel upgrade from 2.4.21-0.25mdk to > 2.4.21-0.27mdk but via rpmdrake. Did mandrake really remove it ??

Re: [Asterisk-Users] E100P - Error 500

Scott Stingel wrote: I get lots of these in a very busy system, along with PRI frame errors/retransmissions. It is my understanding that this is due to an inadequate buffering mechanism in asterisk. Mark Spencer is aware of the problem, and has said he'll work on it soon. In small numbers, the

[Asterisk-Users] Oops!

Didn't realise that replies are still tagged to specific threads in the mail headers. Oops! A few of my postings so far have been replies (to save me retyping the list address) - but aren't really replies (they are completely off topic). Hope this doesn't cause too many problems in the archive

Re: [Asterisk-Users] picking a channel bank

> Is there a reference somewhere describing and comparing channel banks (old > and new)? not comprehensive but a start http://voip-info.org/wiki-Asterisk+Hardware > Can modern channel banks handle translating all the "new" analog signaling > features into a T-1 format? For example, can it interpr

RE: [Asterisk-Users] E100P - Error 500

Ah, this *is* interesting! I have had similar occurrences, but always assumed they were related to extreme loading in my very busy IVR environment. In my situation, the channels come back to life in a few minutes (because asterisk restarts all PRI channels periodically). Is your situation reprod

Re: [Asterisk-Users] ChanIsAvail and SIP

Hi! > > Hello all. Has anyone had any success using ChanIsAvail with only SIP > > channels? Is there another, better way to check if an extension is busy > > without dialing it? > > Well, SIP devices live their own life and should really handle this signalling > themselves. That's why ChanIsA

Re: [Asterisk-Users] At last!!! :)

Jess: Try with: Dial(SIP/[EMAIL PROTECTED],20,t) Remove 'r' option from your dial command, maybe 'show application Dial' from CLI could help you more. Regards, Gus - Original Message - From: Jess Magnaye To: [EMAIL PROTECTED] Sent: Friday, January 09, 2004 7:55 PM Subject: [Asterisk-

RE: [Asterisk-Users] PSTN > SIP Gateways?

Avoid Audiocode. You can't get any support -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Graves Sent: Friday, January 09, 2004 2:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PSTN > SIP Gateways? Since my earlier inquiry about gateways w

Re: [Asterisk-Users] E100P - Error 500

Scott Stingel wrote: Ah, this *is* interesting! I have had similar occurrences, but always assumed they were related to extreme loading in my very busy IVR environment. In my situation, the channels come back to life in a few minutes (because asterisk restarts all PRI channels periodical

[Asterisk-Users] R2 Digital - Brazil

Hi all, I will start testing libr2 for brazilian R2. Any clue? Daniel ___ 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/li

Re: [Asterisk-Users] crontab

Philipp von Klitzing said: > oHi! > >> Ladies and Gentlemen, can anyone please help and let me know what is >> the way to start Asterisk automatically using a cronjob, thanks > > http://www.voip-info.org/wiki-Asterisk+administration > > Philipp > > Guess maybe I don't leave my system running long

[Asterisk-Users] Record all phone calls

I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks,   [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Mon

RE: [Asterisk-Users] Chagres Technologies, Inc

My experience has been one of unresponsiveness to my e-mails. I have ordered and received devices from other providers in the time I have been waiting for Chagres. As of now, based on my experiences and those of others that I have heard from I would highly recommend avoiding Chagres and Mr. Brown

Re: [Asterisk-Users] E1 - E100P connected to Cisco - problem - HELP

Stephen J. Wilcox wrote: You are trying to have both ends act as users, cisco can support emulating a network interface (isdn protocol-emulate in serial interface config) but in my experience i could get the circuit up but it would bounce and i couldnt get signalling to work.. to be fair

[Asterisk-Users] Bridging ethernet over hdlc

I'm attempting to bridge ethernet over hdlc between two * boxes. If anyone has any information they can offer concerning this, it would be greatly appreciated. Here's the configuration the companies IT guy wants to bridge. I have it working already without a bridge, but he wants the head box's e

Re: [Asterisk-Users] newbie question; can * screen calls?

Hi! > Does * have the capability to screen calls?  IOW, if someone calls in from > outside (ie. not a local extension), can * ask the calling party to state > their name, record it, ring the recipient, play the caller's name for the > recipient, then give the reci

Re: [Asterisk-Users] Record all phone calls

  - Original Message - From: Jimmy Riley To: '[EMAIL PROTECTED]' Sent: Saturday, January 10, 2004 10:01 AM Subject: [Asterisk-Users] Record all phone calls I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is wh

Re: [Asterisk-Users] E100P - Error 500

Daniel Bichara wrote: Hi, I am running * with E100P board. At least every our I got an Error 500 message and ISDN-PRI restarts: Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 12:54:27 WARNING[98311]: File chan_zap.c,

[Asterisk-Users] Mediatrix 1204

Someone have the MIB for MEdiatrix 1204 version 2.4.10.68? thanks -- Almada Tres SA de CV Mitel Networks Eng. Gonzalo Gasca Meza Service Engineer 52+(55)53730570 Mexico City, Mexico ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.dig

[Asterisk-Users] Free Software or not -- that's the question /* New subject */

(removed In-Reply-To header) On Sat, Jan 10, 2004 at 10:01:12AM +, WipeOut wrote: > >> > >>And make sure to send in a disclaimer otherwise it will not even be > >>looked at.. :) > >> > >How do we know what is disclaimed or not disclaimed? > >/O > > > Digium have all the Disclaimers and will

Re: [Asterisk-Users] Chagres Technologies, Inc

For the list, Mike received a partial order shipped 15-Dec, SN ending 4CD8. Mike received email replies on 3-Dec and 17-Dec advising him on his order. Mike ack'd those emails. This is the first time we have heard anything (phone calls or email) from Mike since 17-Dec. Our CDR and SMTP logs s

Re: [Asterisk-Users] Chagres Technologies, Inc

Hi List, Matt hasn't contacted us directly about this. I've responded to his previous statement that he hasn't recevied the last 20 units, and never heard back from him. Matt, again, if this is an issue please do contact us. Our CDR and SMTP logs show no such attempt. Our inventory records sho

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

[EMAIL PROTECTED] wrote: And why is this unnecessary cruft included in the source tree? So that Digium can leverage the Free Software community into developing proprietary software for them. Am I way off the mark? I think you're unfairly impugning Digium's motives. And I also think you're--ag

[Asterisk-Users] IAX v1 Changes

I plan on removing chan_iax from the normal build process, and making chan_iax2 register itself as both "IAX" and "IAX2". IAX1 if built will register itself as "IAX1". CVS asterisk has already been updated such that "IAX1" can be used to identify an IAX channel. The removal of chan_iax from the

Re: [Asterisk-Users] Chagres Technologies, Inc

Just refund the guy his money... - Original Message - From: " John Brown (CV)" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, January 10, 2004 2:46 PM Subject: Re: [Asterisk-Users] Chagres Technologies, Inc > For the list, > > Mike received a partial order shipped 15-Dec, SN

RE: [Asterisk-Users] Record all phone calls

Here is what I have now. Where should the line " exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]? Right now if I call sip to sip monitoring starts and the calls connect but I only get 44 byte files. If I call and iaxtel number monitoring starts but call never ge

Re: [Asterisk-Users] Chagres Technologies, Inc

I would feel sympathetic to Chagres Technologies but I have read many many posts to the same effect. If you are going to take someone's money then follow through on your service or product in a timely manner. If you cannot, close your business and stop taking people's money. - Original Mess

Re: [Asterisk-Users] Chagres Technologies, Inc

Sorry, but how can you ID his inbound packets? - Original Message - From: "admin" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, January 10, 2004 3:17 PM Subject: Re: [Asterisk-Users] Chagres Technologies, Inc > Just refund the guy his money... > - Original Message ---

[Asterisk-Users] ADSI Configs

Hi All If I want to get my ADSI Phones (successfully connected off a Rhino Channel Bank and TE410P) to connect to Asterisk to get their config downloaded, is there something specific needed in extensions.conf for them to dial to get this? Thanks :) __

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

I work for an interconnect that sells 3com and NEC. When I made this project my own and followed through to show my boss, he said, "this is going to ruin our industry" If that is the case then so be it. Same with mp3s and the music industry. Had they embraced the technology, everyone could be ma

RE: [Asterisk-Users] Chagres Technologies, Inc

just drop it! it is for them to iron out! and for the record, I received my order within a week of placing the order. -Original Message- From: admin [mailto:[EMAIL PROTECTED] Sent: Sat 1/10/2004 3:23 PM To: [EMAIL PROTECTED] Cc: S

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

w, You also have to consider that if Asterisk used any GPL code we would loose the ability to use/link to openh323, provide g729 of any sort. We would also Dialogic support. Now do you want to be the one to tell everyong that depends on h323, g729 or Dialogic cards they are just SOL? Aste

[Asterisk-Users] far end disconnect supervision

I'm starting to shop for my first channel bank and one of the features that eveyone seems to recommend is "far end disconnect supervision". What other terms do various manufactures use to describe this same feature ? Is "calling party disconnect" the same as "far end disconnect supervision" ? Th

Re: [Asterisk-Users] Chagres Technologies, Inc

Hi, You have read a small sample of posts. With over 300 customers there are maybe 25 to 30 in Nov that had issues. Thats less than 10 percent. And for Dec, our order lead times have gotten back on track. The average turn around time for an order in Jan 2004 is 2.1 business days. Compare th

Re: [Asterisk-Users] Chagres Technologies, Inc

its simple, i can lookup the MX for his zone, then look up the A RR for each MX, and then search the logs for IP's or I can even expand the search to look for CIDR prefixes. I can also lookup in my private RBL, any query my SMTP machine would have made to see if his IP(s) are spam sources or not.

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

admin said: > I work for an interconnect that sells 3com and NEC. When I made this > project my own and followed through to show my boss, he said, "this is > going > to ruin our industry" > > If that is the case then so be it. Same with mp3s and the music industry. > Had they embraced the technol

[Asterisk-Users] drop calls with T100P / PRI

Hi List, a number of our customers are reporting dropped calls. here is the config. 1 T100P T1 Card 1 Asterisk (Mid Nov build) T1 is signalled as a PRI(National) The card will only sync up if we clock, if we line side clock the card goes into yellow alarm and won't sync up. the only er

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

It's very hard to find a business model for working with Open Source Software in a for-profit software company. Mysql and Digium are success stories that work with a two-fold model that seems to work. Do not forget that there are companies out there that wants to buy the software with a more bus

Re: [Asterisk-Users] Chagres Technologies, Inc

common on john, stop the bs. we all know email can be sent from hundreds of different valid accounts that you can't trace that way (yahoo and msn as just two), and those of us that have been involved with security understand it rather well. > its simple, > i can lookup the

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

Brian West wrote: w, You also have to consider that if Asterisk used any GPL code we would loose the ability to use/link to openh323, provide g729 of any sort. We would also Dialogic support. Now do you want to be the one to tell everyong that depends on h323, g729 or Dialogic cards they a

[Asterisk-Users] My first E1 card is running :)

Just happy. hardware information: -- Some small factor IBM Celetron (coppermine) at 1100 (11*100FSB) 256 RAM 15GB Hard. 1 x Digium E100P - E1 Line from telco with 300 Dids 1 x TDM400P for local phones --- Few small machines (mainly brand PII at 233Mhz with TDM400P Cards. -

Re: [Asterisk-Users] Chagres Technologies, Inc

rick, i didn't say that they couldn't have sent email from another location. certainly yahoo and msn are harder to deal with. Yes, rick you can do some tracing the way I mentioned. lets see:dig routers.com mx routers.com.4H IN MX10 texas.routers.com. ;; AUTHORITY SECTION

Re: [Asterisk-Users] far end disconnect supervision

> I'm starting to shop for my first channel bank and one of the features > that eveyone seems to recommend is "far end disconnect supervision". > What other terms do various manufactures use to describe this same > feature ? > > Is "calling party disconnect" the same as "far end disconnect > super

RE: [Asterisk-Users] drop calls with T100P / PRI

Maybe you could look in /var/log/asterisk/messages and see if there are any errors that correspond to the times of dropped calls? If so, what kinds of errors do you see there? As far as the problems you report receiving emails from your customers, maybe your provider is spam-filtering your mail, a

Re: [Asterisk-Users] Cisco Gear

Hi, Would you mind giving me an idea of the price level for the 7970, 7960, 7940 and 7920 ? Qty 5 or more. Shipping to France. On Fri, Jan 09, 2004 at 06:00:29PM +, Adthrawn wrote: > Hi, > > I know it's not really the place, but if anybody in the UK (or US) is > interested, I'm clearing ou

Re: [Asterisk-Users] Cisco Gear

Whoops... echo set ignore_list_reply_to = yes >>.muttrc Sorry. I believe that the Reply-To setting on this list must have been discussed here a few times here, so I won't start :) On Fri, Jan 09, 2004 at 06:00:29PM +, Adthrawn wrote: > Hi, > > I know it's not really the place, but if anybod

Re: [Asterisk-Users] Chagres Technologies, Inc

John, Take your discussion off list... It is way off topic. I think you do yourself more harm than good by responding to these issues on list. If you want to build confidence in your company then ask your satisfied customers to reccommend you and give their testimonials regarding your speedy servic

[Asterisk-Users] Asterisk + BudgeTone (behind NAT)

I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS about a week ago. My problem is that I'm only getting half-duplex communication --

Re: [Asterisk-Users] far end disconnect supervision

Rich Adamson wrote: > > > I'm starting to shop for my first channel bank and one of the features > > that eveyone seems to recommend is "far end disconnect supervision". > > What other terms do various manufactures use to describe this same > > feature ? > > > > Is "calling party disconnect" the

Re: [Asterisk-Users] far end disconnect supervision

> If some channel banks don't support this, how on earth do they know when > the telco side of the call has hung up ? They don't. They rely on either a timeout or the called party hanging up to disconnect the call. ___ Asterisk-Users mailing list [EMAI

[Asterisk-Users] Record calls where to put line?

Here is what I have now. Where should the line " exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]? Right now if I call sip to sip monitoring starts and the calls connect but I only get 44 byte files. If I call and iaxtel number monitoring starts but call never ge

Re: [Asterisk-Users] far end disconnect supervision

> > > I'm starting to shop for my first channel bank and one of the features > > > that eveyone seems to recommend is "far end disconnect supervision". > > > What other terms do various manufactures use to describe this same > > > feature ? > > > > > > Is "calling party disconnect" the same as "far

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

On Sat, Jan 10, 2004 at 03:03:23PM -0500, Brian Capouch wrote: > > I think you're unfairly impugning Digium's motives. And I also think > you're--again--salting your post with enough innuendo that a reasonable > person might suspect you of flame-baiting. Baiting, perhaps, but not flames. If th

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

Hi, thats very probably a NAT problem. Your NAT box is probaly blocking the incoming UDP voice stream. If asteriks supports a RTP Proxy you can try that. best regards, Arnd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/m

Re: [Asterisk-Users] Oops!

On Sat, 2004-01-10 at 11:22, Terence Parker wrote: > Didn't realise that replies are still tagged to specific threads in the > mail headers. Oops! > > A few of my postings so far have been replies (to save me retyping the > list address) - but aren't really replies (they are completely off > to

Re: [Asterisk-Users] far end disconnect supervision

Rich Adamson wrote: > > > > > I'm starting to shop for my first channel bank and one of the features > > > > that eveyone seems to recommend is "far end disconnect supervision". > > > > What other terms do various manufactures use to describe this same > > > > feature ? > > > > > > > > Is "calli

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

I'm going to keep this short and to the point. Nobody is twisting your arm to use Asterisk... we didn't find you.. you found us. NEXT!!! bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To U

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

On Sat, 2004-01-10 at 18:47, [EMAIL PROTECTED] wrote: > I have always been suspicious of centralized control and dictatorship, > benevolent or otherwise. After thinking for some time about the > licensing structure of code for Asterisk, I am not sure that > their motives are so innocuous and altr

Re: [Asterisk-Users] drop calls with T100P / PRI

On Sat, 2004-01-10 at 15:19, John Brown (CV) wrote: > busydetect=yes > callprogress=yes > musiconhold=default > signalling=pri_cpe > group=1 > channel=> 1-4 Well seems you haven't been on the list, or maybe you haven't been paying attention since we have been covering that problem for a while late

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

i also had the same problem temporarily i solved my problem with both outside NAT. u can also do it if both inside NAT. * outside NAT and Budgetone behind NAT simply doesn't seem to work. if u ever solve this problem please let me know too. thanks cm - Original Message - From: "Owen

[Asterisk-Users] default music source for SIP channel

The wiki says this about the MusicOnHold command: "Plays hold music specified by class. If omitted, the default music source for the channel will be used." http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold How do I set the default music on hold class for the SIP channel ?

[Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB. Thanks ___ Asterisk-Users mailing

RE: [Asterisk-Users] default music source for SIP channel

> The wiki says this about the MusicOnHold command: > > "Plays hold music specified by class. If omitted, the default music > source for the channel will be used." > http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold > > How do I set the default music on hold class for the

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